A new display stream compression standard under development in VESA

NASA Astrophysics Data System (ADS)

Jacobson, Natan; Thirumalai, Vijayaraghavan; Joshi, Rajan; Goel, James

2017-09-01

The Advanced Display Stream Compression (ADSC) codec project is in development in response to a call for technologies from the Video Electronics Standards Association (VESA). This codec targets visually lossless compression of display streams at a high compression rate (typically 6 bits/pixel) for mobile/VR/HDR applications. Functionality of the ADSC codec is described in this paper, and subjective trials results are provided using the ISO 29170-2 testing protocol.

Performance comparison of AV1, HEVC, and JVET video codecs on 360 (spherical) video

NASA Astrophysics Data System (ADS)

Topiwala, Pankaj; Dai, Wei; Krishnan, Madhu; Abbas, Adeel; Doshi, Sandeep; Newman, David

2017-09-01

This paper compares the coding efficiency performance on 360 videos, of three software codecs: (a) AV1 video codec from the Alliance for Open Media (AOM); (b) the HEVC Reference Software HM; and (c) the JVET JEM Reference SW. Note that 360 video is especially challenging content, in that one codes full res globally, but typically looks locally (in a viewport), which magnifies errors. These are tested in two different projection formats ERP and RSP, to check consistency. Performance is tabulated for 1-pass encoding on two fronts: (1) objective performance based on end-to-end (E2E) metrics such as SPSNR-NN, and WS-PSNR, currently developed in the JVET committee; and (2) informal subjective assessment of static viewports. Constant quality encoding is performed with all the three codecs for an unbiased comparison of the core coding tools. Our general conclusion is that under constant quality coding, AV1 underperforms HEVC, which underperforms JVET. We also test with rate control, where AV1 currently underperforms the open source X265 HEVC codec. Objective and visual evidence is provided.

Design considerations for computationally constrained two-way real-time video communication

NASA Astrophysics Data System (ADS)

Bivolarski, Lazar M.; Saunders, Steven E.; Ralston, John D.

2009-08-01

Today's video codecs have evolved primarily to meet the requirements of the motion picture and broadcast industries, where high-complexity studio encoding can be utilized to create highly-compressed master copies that are then broadcast one-way for playback using less-expensive, lower-complexity consumer devices for decoding and playback. Related standards activities have largely ignored the computational complexity and bandwidth constraints of wireless or Internet based real-time video communications using devices such as cell phones or webcams. Telecommunications industry efforts to develop and standardize video codecs for applications such as video telephony and video conferencing have not yielded image size, quality, and frame-rate performance that match today's consumer expectations and market requirements for Internet and mobile video services. This paper reviews the constraints and the corresponding video codec requirements imposed by real-time, 2-way mobile video applications. Several promising elements of a new mobile video codec architecture are identified, and more comprehensive computational complexity metrics and video quality metrics are proposed in order to support the design, testing, and standardization of these new mobile video codecs.

Energy minimization of mobile video devices with a hardware H.264/AVC encoder based on energy-rate-distortion optimization

NASA Astrophysics Data System (ADS)

Kang, Donghun; Lee, Jungeon; Jung, Jongpil; Lee, Chul-Hee; Kyung, Chong-Min

2014-09-01

In mobile video systems powered by battery, reducing the encoder's compression energy consumption is critical to prolong its lifetime. Previous Energy-rate-distortion (E-R-D) optimization methods based on a software codec is not suitable for practical mobile camera systems because the energy consumption is too large and encoding rate is too low. In this paper, we propose an E-R-D model for the hardware codec based on the gate-level simulation framework to measure the switching activity and the energy consumption. From the proposed E-R-D model, an energy minimizing algorithm for mobile video camera sensor have been developed with the GOP (Group of Pictures) size and QP(Quantization Parameter) as run-time control variables. Our experimental results show that the proposed algorithm provides up to 31.76% of energy consumption saving while satisfying the rate and distortion constraints.

Java Image I/O for VICAR, PDS, and ISIS

NASA Technical Reports Server (NTRS)

Deen, Robert G.; Levoe, Steven R.

2011-01-01

This library, written in Java, supports input and output of images and metadata (labels) in the VICAR, PDS image, and ISIS-2 and ISIS-3 file formats. Three levels of access exist. The first level comprises the low-level, direct access to the file. This allows an application to read and write specific image tiles, lines, or pixels and to manipulate the label data directly. This layer is analogous to the C-language "VICAR Run-Time Library" (RTL), which is the image I/O library for the (C/C++/Fortran) VICAR image processing system from JPL MIPL (Multimission Image Processing Lab). This low-level library can also be used to read and write labeled, uncompressed images stored in formats similar to VICAR, such as ISIS-2 and -3, and a subset of PDS (image format). The second level of access involves two codecs based on Java Advanced Imaging (JAI) to provide access to VICAR and PDS images in a file-format-independent manner. JAI is supplied by Sun Microsystems as an extension to desktop Java, and has a number of codecs for formats such as GIF, TIFF, JPEG, etc. Although Sun has deprecated the codec mechanism (replaced by IIO), it is still used in many places. The VICAR and PDS codecs allow any program written using the JAI codec spec to use VICAR or PDS images automatically, with no specific knowledge of the VICAR or PDS formats. Support for metadata (labels) is included, but is format-dependent. The PDS codec, when processing PDS images with an embedded VIAR label ("dual-labeled images," such as used for MER), presents the VICAR label in a new way that is compatible with the VICAR codec. The third level of access involves VICAR, PDS, and ISIS Image I/O plugins. The Java core includes an "Image I/O" (IIO) package that is similar in concept to the JAI codec, but is newer and more capable. Applications written to the IIO specification can use any image format for which a plug-in exists, with no specific knowledge of the format itself.

Analysis of the possibility of using G.729 codec for steganographic transmission

NASA Astrophysics Data System (ADS)

Piotrowski, Zbigniew; Ciołek, Michał; Dołowski, Jerzy; Wojtuń, Jarosław

2017-04-01

Network steganography is dedicated in particular for those communication services for which there are no bridges or nodes carrying out unintentional attacks on steganographic sequence. In order to set up a hidden communication channel the method of data encoding and decoding was implemented using code books of codec G.729. G.729 codec includes, in its construction, linear prediction vocoder CS-ACELP (Conjugate Structure Algebraic Code Excited Linear Prediction), and by modifying the binary content of the codebook, it is easy to change a binary output stream. The article describes the results of research on the selection of these bits of the codebook codec G.729 which the negation of the least have influence to the loss of quality and fidelity of the output signal. The study was performed with the use of subjective and objective listening tests.

Evaluation of voice codecs for the Australian mobile satellite system

NASA Technical Reports Server (NTRS)

Bundrock, Tony; Wilkinson, Mal

1990-01-01

The evaluation procedure to choose a low bit rate voice coding algorithm is described for the Australian land mobile satellite system. The procedure is designed to assess both the inherent quality of the codec under 'normal' conditions and its robustness under 'severe' conditions. For the assessment, normal conditions were chosen to be random bit error rate with added background acoustic noise and the severe condition is designed to represent burst error conditions when mobile satellite channel suffers from signal fading due to roadside vegetation. The assessment is divided into two phases. First, a reduced set of conditions is used to determine a short list of candidate codecs for more extensive testing in the second phase. The first phase conditions include quality and robustness and codecs are ranked with a 60:40 weighting on the two. Second, the short listed codecs are assessed over a range of input voice levels, BERs, background noise conditions, and burst error distributions. Assessment is by subjective rating on a five level opinion scale and all results are then used to derive a weighted Mean Opinion Score using appropriate weights for each of the test conditions.

A Dynamic Image Quality Evaluation of Videofluoroscopy Images: Considerations for Telepractice Applications.

PubMed

Burns, Clare L; Keir, Benjamin; Ward, Elizabeth C; Hill, Anne J; Farrell, Anna; Phillips, Nick; Porter, Linda

2015-08-01

High-quality fluoroscopy images are required for accurate interpretation of videofluoroscopic swallow studies (VFSS) by speech pathologists and radiologists. Consequently, integral to developing any system to conduct VFSS remotely via telepractice is ensuring that the quality of the VFSS images transferred via the telepractice system is optimized. This study evaluates the extent of change observed in image quality when videofluoroscopic images are transmitted from a digital fluoroscopy system to (a) current clinical equipment (KayPentax Digital Swallowing Workstation, and b) four different telepractice system configurations. The telepractice system configurations consisted of either a local C20 or C60 Cisco TelePresence System (codec unit) connected to the digital fluoroscopy system and linked to a second remote C20 or C60 Cisco TelePresence System via a network running at speeds of either 2, 4 or 6 megabits per second (Mbit/s). Image quality was tested using the NEMA XR 21 Phantom, and results demonstrated some loss in spatial resolution, low contrast detectability and temporal resolution for all transferred images when compared to the fluoroscopy source. When using higher capacity codec units and/or the highest bandwidths to support data transmission, image quality transmitted through the telepractice system was found to be comparable if not better than the current clinical system. This study confirms that telepractice systems can be designed to support fluoroscopy image transfer and highlights important considerations when developing telepractice systems for VFSS analysis to ensure high-quality radiological image reproduction.

An overview of new video coding tools under consideration for VP10: the successor to VP9

NASA Astrophysics Data System (ADS)

Mukherjee, Debargha; Su, Hui; Bankoski, James; Converse, Alex; Han, Jingning; Liu, Zoe; Xu, Yaowu

2015-09-01

Google started an opensource project, entitled the WebM Project, in 2010 to develop royaltyfree video codecs for the web. The present generation codec developed in the WebM project called VP9 was finalized in mid2013 and is currently being served extensively by YouTube, resulting in billions of views per day. Even though adoption of VP9 outside Google is still in its infancy, the WebM project has already embarked on an ambitious project to develop a next edition codec VP10 that achieves at least a generational bitrate reduction over the current generation codec VP9. Although the project is still in early stages, a set of new experimental coding tools have already been added to baseline VP9 to achieve modest coding gains over a large enough test set. This paper provides a technical overview of these coding tools.

Secure videoconferencing equipment switching system and method

DOEpatents

Hansen, Michael E [Livermore, CA

2009-01-13

A switching system and method are provided to facilitate use of videoconference facilities over a plurality of security levels. The system includes a switch coupled to a plurality of codecs and communication networks. Audio/Visual peripheral components are connected to the switch. The switch couples control and data signals between the Audio/Visual peripheral components and one but nor both of the plurality of codecs. The switch additionally couples communication networks of the appropriate security level to each of the codecs. In this manner, a videoconferencing facility is provided for use on both secure and non-secure networks.

On transform coding tools under development for VP10

NASA Astrophysics Data System (ADS)

Parker, Sarah; Chen, Yue; Han, Jingning; Liu, Zoe; Mukherjee, Debargha; Su, Hui; Wang, Yongzhe; Bankoski, Jim; Li, Shunyao

2016-09-01

Google started the WebM Project in 2010 to develop open source, royaltyfree video codecs designed specifically for media on the Web. The second generation codec released by the WebM project, VP9, is currently served by YouTube, and enjoys billions of views per day. Realizing the need for even greater compression efficiency to cope with the growing demand for video on the web, the WebM team embarked on an ambitious project to develop a next edition codec, VP10, that achieves at least a generational improvement in coding efficiency over VP9. Starting from VP9, a set of new experimental coding tools have already been added to VP10 to achieve decent coding gains. Subsequently, Google joined a consortium of major tech companies called the Alliance for Open Media to jointly develop a new codec AV1. As a result, the VP10 effort is largely expected to merge with AV1. In this paper, we focus primarily on new tools in VP10 that improve coding of the prediction residue using transform coding techniques. Specifically, we describe tools that increase the flexibility of available transforms, allowing the codec to handle a more diverse range or residue structures. Results are presented on a standard test set.

Towards a next generation open-source video codec

NASA Astrophysics Data System (ADS)

Bankoski, Jim; Bultje, Ronald S.; Grange, Adrian; Gu, Qunshan; Han, Jingning; Koleszar, John; Mukherjee, Debargha; Wilkins, Paul; Xu, Yaowu

2013-02-01

Google has recently been developing a next generation opensource video codec called VP9, as part of the experimental branch of the libvpx repository included in the WebM project (http://www.webmproject.org/). Starting from the VP8 video codec released by Google in 2010 as the baseline, a number of enhancements and new tools have been added to improve the coding efficiency. This paper provides a technical overview of the current status of this project along with comparisons and other stateoftheart video codecs H. 264/AVC and HEVC. The new tools that have been added so far include: larger prediction block sizes up to 64x64, various forms of compound INTER prediction, more modes for INTRA prediction, ⅛pel motion vectors and 8tap switchable subpel interpolation filters, improved motion reference generation and motion vector coding, improved entropy coding and framelevel entropy adaptation for various symbols, improved loop filtering, incorporation of Asymmetric Discrete Sine Transforms and larger 16x16 and 32x32 DCTs, frame level segmentation to group similar areas together, etc. Other tools and various bitstream features are being actively worked on as well. The VP9 bitstream is expected to be finalized by earlyto mid2013. Results show VP9 to be quite competitive in performance with mainstream stateoftheart codecs.

Novel inter and intra prediction tools under consideration for the emerging AV1 video codec

NASA Astrophysics Data System (ADS)

Joshi, Urvang; Mukherjee, Debargha; Han, Jingning; Chen, Yue; Parker, Sarah; Su, Hui; Chiang, Angie; Xu, Yaowu; Liu, Zoe; Wang, Yunqing; Bankoski, Jim; Wang, Chen; Keyder, Emil

2017-09-01

Google started the WebM Project in 2010 to develop open source, royalty- free video codecs designed specifically for media on the Web. The second generation codec released by the WebM project, VP9, is currently served by YouTube, and enjoys billions of views per day. Realizing the need for even greater compression efficiency to cope with the growing demand for video on the web, the WebM team embarked on an ambitious project to develop a next edition codec AV1, in a consortium of major tech companies called the Alliance for Open Media, that achieves at least a generational improvement in coding efficiency over VP9. In this paper, we focus primarily on new tools in AV1 that improve the prediction of pixel blocks before transforms, quantization and entropy coding are invoked. Specifically, we describe tools and coding modes that improve intra, inter and combined inter-intra prediction. Results are presented on standard test sets.

Feasibility of video codec algorithms for software-only playback

NASA Astrophysics Data System (ADS)

Rodriguez, Arturo A.; Morse, Ken

1994-05-01

Software-only video codecs can provide good playback performance in desktop computers with a 486 or 68040 CPU running at 33 MHz without special hardware assistance. Typically, playback of compressed video can be categorized into three tasks: the actual decoding of the video stream, color conversion, and the transfer of decoded video data from system RAM to video RAM. By current standards, good playback performance is the decoding and display of video streams of 320 by 240 (or larger) compressed frames at 15 (or greater) frames-per- second. Software-only video codecs have evolved by modifying and tailoring existing compression methodologies to suit video playback in desktop computers. In this paper we examine the characteristics used to evaluate software-only video codec algorithms, namely: image fidelity (i.e., image quality), bandwidth (i.e., compression) ease-of-decoding (i.e., playback performance), memory consumption, compression to decompression asymmetry, scalability, and delay. We discuss the tradeoffs among these variables and the compromises that can be made to achieve low numerical complexity for software-only playback. Frame- differencing approaches are described since software-only video codecs typically employ them to enhance playback performance. To complement other papers that appear in this session of the Proceedings, we review methods derived from binary pattern image coding since these methods are amenable for software-only playback. In particular, we introduce a novel approach called pixel distribution image coding.

Software-codec-based full motion video conferencing on the PC using visual pattern image sequence coding

NASA Astrophysics Data System (ADS)

Barnett, Barry S.; Bovik, Alan C.

1995-04-01

This paper presents a real time full motion video conferencing system based on the Visual Pattern Image Sequence Coding (VPISC) software codec. The prototype system hardware is comprised of two personal computers, two camcorders, two frame grabbers, and an ethernet connection. The prototype system software has a simple structure. It runs under the Disk Operating System, and includes a user interface, a video I/O interface, an event driven network interface, and a free running or frame synchronous video codec that also acts as the controller for the video and network interfaces. Two video coders have been tested in this system. Simple implementations of Visual Pattern Image Coding and VPISC have both proven to support full motion video conferencing with good visual quality. Future work will concentrate on expanding this prototype to support the motion compensated version of VPISC, as well as encompassing point-to-point modem I/O and multiple network protocols. The application will be ported to multiple hardware platforms and operating systems. The motivation for developing this prototype system is to demonstrate the practicality of software based real time video codecs. Furthermore, software video codecs are not only cheaper, but are more flexible system solutions because they enable different computer platforms to exchange encoded video information without requiring on-board protocol compatible video codex hardware. Software based solutions enable true low cost video conferencing that fits the `open systems' model of interoperability that is so important for building portable hardware and software applications.

A hybrid video codec based on extended block sizes, recursive integer transforms, improved interpolation, and flexible motion representation

NASA Astrophysics Data System (ADS)

Karczewicz, Marta; Chen, Peisong; Joshi, Rajan; Wang, Xianglin; Chien, Wei-Jung; Panchal, Rahul; Coban, Muhammed; Chong, In Suk; Reznik, Yuriy A.

2011-01-01

This paper describes video coding technology proposal submitted by Qualcomm Inc. in response to a joint call for proposal (CfP) issued by ITU-T SG16 Q.6 (VCEG) and ISO/IEC JTC1/SC29/WG11 (MPEG) in January 2010. Proposed video codec follows a hybrid coding approach based on temporal prediction, followed by transform, quantization, and entropy coding of the residual. Some of its key features are extended block sizes (up to 64x64), recursive integer transforms, single pass switched interpolation filters with offsets (single pass SIFO), mode dependent directional transform (MDDT) for intra-coding, luma and chroma high precision filtering, geometry motion partitioning, adaptive motion vector resolution. It also incorporates internal bit-depth increase (IBDI), and modified quadtree based adaptive loop filtering (QALF). Simulation results are presented for a variety of bit rates, resolutions and coding configurations to demonstrate the high compression efficiency achieved by the proposed video codec at moderate level of encoding and decoding complexity. For random access hierarchical B configuration (HierB), the proposed video codec achieves an average BD-rate reduction of 30.88c/o compared to the H.264/AVC alpha anchor. For low delay hierarchical P (HierP) configuration, the proposed video codec achieves an average BD-rate reduction of 32.96c/o and 48.57c/o, compared to the H.264/AVC beta and gamma anchors, respectively.

On the definition of adapted audio/video profiles for high-quality video calling services over LTE/4G

NASA Astrophysics Data System (ADS)

Ndiaye, Maty; Quinquis, Catherine; Larabi, Mohamed Chaker; Le Lay, Gwenael; Saadane, Hakim; Perrine, Clency

2014-01-01

During the last decade, the important advances and widespread availability of mobile technology (operating systems, GPUs, terminal resolution and so on) have encouraged a fast development of voice and video services like video-calling. While multimedia services have largely grown on mobile devices, the generated increase of data consumption is leading to the saturation of mobile networks. In order to provide data with high bit-rates and maintain performance as close as possible to traditional networks, the 3GPP (The 3rd Generation Partnership Project) worked on a high performance standard for mobile called Long Term Evolution (LTE). In this paper, we aim at expressing recommendations related to audio and video media profiles (selection of audio and video codecs, bit-rates, frame-rates, audio and video formats) for a typical video-calling services held over LTE/4G mobile networks. These profiles are defined according to targeted devices (smartphones, tablets), so as to ensure the best possible quality of experience (QoE). Obtained results indicate that for a CIF format (352 x 288 pixels) which is usually used for smartphones, the VP8 codec provides a better image quality than the H.264 codec for low bitrates (from 128 to 384 kbps). However sequences with high motion, H.264 in slow mode is preferred. Regarding audio, better results are globally achieved using wideband codecs offering good quality except for opus codec (at 12.2 kbps).

Pre-processing SAR image stream to facilitate compression for transport on bandwidth-limited-link

DOEpatents

Rush, Bobby G.; Riley, Robert

2015-09-29

Pre-processing is applied to a raw VideoSAR (or similar near-video rate) product to transform the image frame sequence into a product that resembles more closely the type of product for which conventional video codecs are designed, while sufficiently maintaining utility and visual quality of the product delivered by the codec.

ISO-IEC MPEG-2 software video codec

NASA Astrophysics Data System (ADS)

Eckart, Stefan; Fogg, Chad E.

1995-04-01

Part 5 of the International Standard ISO/IEC 13818 `Generic Coding of Moving Pictures and Associated Audio' (MPEG-2) is a Technical Report, a sample software implementation of the procedures in parts 1, 2 and 3 of the standard (systems, video, and audio). This paper focuses on the video software, which gives an example of a fully compliant implementation of the standard and of a good video quality encoder, and serves as a tool for compliance testing. The implementation and some of the development aspects of the codec are described. The encoder is based on Test Model 5 (TM5), one of the best, published, non-proprietary coding models, which was used during MPEG-2 collaborative stage to evaluate proposed algorithms and to verify the syntax. The most important part of the Test Model is controlling the quantization parameter based on the image content and bit rate constraints under both signal-to-noise and psycho-optical aspects. The decoder has been successfully tested for compliance with the MPEG-2 standard, using the ISO/IEC MPEG verification and compliance bitstream test suites as stimuli.

Backwards compatible high dynamic range video compression

NASA Astrophysics Data System (ADS)

Dolzhenko, Vladimir; Chesnokov, Vyacheslav; Edirisinghe, Eran A.

2014-02-01

This paper presents a two layer CODEC architecture for high dynamic range video compression. The base layer contains the tone mapped video stream encoded with 8 bits per component which can be decoded using conventional equipment. The base layer content is optimized for rendering on low dynamic range displays. The enhancement layer contains the image difference, in perceptually uniform color space, between the result of inverse tone mapped base layer content and the original video stream. Prediction of the high dynamic range content reduces the redundancy in the transmitted data while still preserves highlights and out-of-gamut colors. Perceptually uniform colorspace enables using standard ratedistortion optimization algorithms. We present techniques for efficient implementation and encoding of non-uniform tone mapping operators with low overhead in terms of bitstream size and number of operations. The transform representation is based on human vision system model and suitable for global and local tone mapping operators. The compression techniques include predicting the transform parameters from previously decoded frames and from already decoded data for current frame. Different video compression techniques are compared: backwards compatible and non-backwards compatible using AVC and HEVC codecs.

Fast and predictable video compression in software design and implementation of an H.261 codec

NASA Astrophysics Data System (ADS)

Geske, Dagmar; Hess, Robert

1998-09-01

The use of software codecs for video compression becomes commonplace in several videoconferencing applications. In order to reduce conflicts with other applications used at the same time, mechanisms for resource reservation on endsystems need to determine an upper bound for computing time used by the codec. This leads to the demand for predictable execution times of compression/decompression. Since compression schemes as H.261 inherently depend on the motion contained in the video, an adaptive admission control is required. This paper presents a data driven approach based on dynamical reduction of the number of processed macroblocks in peak situations. Besides the absolute speed is a point of interest. The question, whether and how software compression of high quality video is feasible on today's desktop computers, is examined.

Implementing MANETS in Android based environment using Wi-Fi direct

NASA Astrophysics Data System (ADS)

Waqas, Muhammad; Babar, Mohammad Inayatullah Khan; Zafar, Mohammad Haseeb

2015-05-01

Packet loss occurs in real-time voice transmission over wireless broadcast Ad-hoc network which creates disruptions in sound. Basic objective of this research is to design a wireless Ad-hoc network based on two Android devices by using the Wireless Fidelity (WIFI) Direct Application Programming Interface (API) and apply the Network Codec, Reed Solomon Code. The network codec is used to encode the data of a music wav file and recover the lost packets if any, packets are dropped using a loss module at the transmitter device to analyze the performance with the objective of retrieving the original file at the receiver device using the network codec. This resulted in faster transmission of the files despite dropped packets. In the end both files had the original formatted music files with complete performance analysis based on the transmission delay.

Java Library for Input and Output of Image Data and Metadata

NASA Technical Reports Server (NTRS)

Deen, Robert; Levoe, Steven

2003-01-01

A Java-language library supports input and output (I/O) of image data and metadata (label data) in the format of the Video Image Communication and Retrieval (VICAR) image-processing software and in several similar formats, including a subset of the Planetary Data System (PDS) image file format. The library does the following: It provides low-level, direct access layer, enabling an application subprogram to read and write specific image files, lines, or pixels, and manipulate metadata directly. Two coding/decoding subprograms ("codecs" for short) based on the Java Advanced Imaging (JAI) software provide access to VICAR and PDS images in a file-format-independent manner. The VICAR and PDS codecs enable any program that conforms to the specification of the JAI codec to use VICAR or PDS images automatically, without specific knowledge of the VICAR or PDS format. The library also includes Image I/O plugin subprograms for VICAR and PDS formats. Application programs that conform to the Image I/O specification of Java version 1.4 can utilize any image format for which such a plug-in subprogram exists, without specific knowledge of the format itself. Like the aforementioned codecs, the VICAR and PDS Image I/O plug-in subprograms support reading and writing of metadata.

Digital Signal Processing For Low Bit Rate TV Image Codecs

NASA Astrophysics Data System (ADS)

Rao, K. R.

1987-06-01

In view of the 56 KBPS digital switched network services and the ISDN, low bit rate codecs for providing real time full motion color video are under various stages of development. Some companies have already brought the codecs into the market. They are being used by industry and some Federal Agencies for video teleconferencing. In general, these codecs have various features such as multiplexing audio and data, high resolution graphics, encryption, error detection and correction, self diagnostics, freezeframe, split video, text overlay etc. To transmit the original color video on a 56 KBPS network requires bit rate reduction of the order of 1400:1. Such a large scale bandwidth compression can be realized only by implementing a number of sophisticated,digital signal processing techniques. This paper provides an overview of such techniques and outlines the newer concepts that are being investigated. Before resorting to the data compression techniques, various preprocessing operations such as noise filtering, composite-component transformation and horizontal and vertical blanking interval removal are to be implemented. Invariably spatio-temporal subsampling is achieved by appropriate filtering. Transform and/or prediction coupled with motion estimation and strengthened by adaptive features are some of the tools in the arsenal of the data reduction methods. Other essential blocks in the system are quantizer, bit allocation, buffer, multiplexer, channel coding etc.

Test and Evaluation of Teleconferencing Video Codecs Transmitting at 1.5 Mbps.

DTIC Science & Technology

1985-08-01

video teleconferencing codecs on the market as of November 1984 to facilitate the choice of an appropriate frame format and data compression algorithm...Engineer, computer company, male 5. Chapter Officer, national civic organization, female Group Y: 6. Marketing Representative, communication systems...both mon:tors to C4ve t e evi uators an idea what kind of cictures they will have to ; ucge . Special suggestions were given regardinc the sequences witn

Present state of HDTV coding in Japan and future prospect

NASA Astrophysics Data System (ADS)

Murakami, Hitomi

The development status of HDTV digital codecs in Japan is evaluated; several bit rate-reduction codecs have been developed for 1125 lines/60-field HDTV, and performance trials have been conducted through satellite and optical fiber links. Prospective development efforts will attempt to achieve more efficient coding schemes able to reduce the bit rate to as little as 45 Mbps, as well as to apply coding schemes to automated teller machine networks.

Testing and Performance Analysis of the Multichannel Error Correction Code Decoder

NASA Technical Reports Server (NTRS)

Soni, Nitin J.

1996-01-01

This report provides the test results and performance analysis of the multichannel error correction code decoder (MED) system for a regenerative satellite with asynchronous, frequency-division multiple access (FDMA) uplink channels. It discusses the system performance relative to various critical parameters: the coding length, data pattern, unique word value, unique word threshold, and adjacent-channel interference. Testing was performed under laboratory conditions and used a computer control interface with specifically developed control software to vary these parameters. Needed technologies - the high-speed Bose Chaudhuri-Hocquenghem (BCH) codec from Harris Corporation and the TRW multichannel demultiplexer/demodulator (MCDD) - were fully integrated into the mesh very small aperture terminal (VSAT) onboard processing architecture and were demonstrated.

System on a chip with MPEG-4 capability

NASA Astrophysics Data System (ADS)

Yassa, Fathy; Schonfeld, Dan

2002-12-01

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

Flexible high-speed CODEC

NASA Technical Reports Server (NTRS)

Segallis, Greg P.; Wernlund, Jim V.; Corry, Glen

1993-01-01

This report is prepared by Harris Government Communication Systems Division for NASA Lewis Research Center under contract NAS3-25087. It is written in accordance with SOW section 4.0 (d) as detailed in section 2.6. The purpose of this document is to provide a summary of the program, performance results and analysis, and a technical assessment. The purpose of this program was to develop a flexible, high-speed CODEC that provides substantial coding gain while maintaining bandwidth efficiency for use in both continuous and bursted data environments for a variety of applications.

Speech processing using conditional observable maximum likelihood continuity mapping

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

Hogden, John; Nix, David

A computer implemented method enables the recognition of speech and speech characteristics. Parameters are initialized of first probability density functions that map between the symbols in the vocabulary of one or more sequences of speech codes that represent speech sounds and a continuity map. Parameters are also initialized of second probability density functions that map between the elements in the vocabulary of one or more desired sequences of speech transcription symbols and the continuity map. The parameters of the probability density functions are then trained to maximize the probabilities of the desired sequences of speech-transcription symbols. A new sequence ofmore » speech codes is then input to the continuity map having the trained first and second probability function parameters. A smooth path is identified on the continuity map that has the maximum probability for the new sequence of speech codes. The probability of each speech transcription symbol for each input speech code can then be output.« less

Experimental research and comparison of LDPC and RS channel coding in ultraviolet communication systems.

PubMed

Wu, Menglong; Han, Dahai; Zhang, Xiang; Zhang, Feng; Zhang, Min; Yue, Guangxin

2014-03-10

We have implemented a modified Low-Density Parity-Check (LDPC) codec algorithm in ultraviolet (UV) communication system. Simulations are conducted with measured parameters to evaluate the LDPC-based UV system performance. Moreover, LDPC (960, 480) and RS (18, 10) are implemented and experimented via a non-line-of-sight (NLOS) UV test bed. The experimental results are in agreement with the simulation and suggest that based on the given power and 10(-3)bit error rate (BER), in comparison with an uncoded system, average communication distance increases 32% with RS code, while 78% with LDPC code.

Performance comparison of leading image codecs: H.264/AVC Intra, JPEG2000, and Microsoft HD Photo

NASA Astrophysics Data System (ADS)

Tran, Trac D.; Liu, Lijie; Topiwala, Pankaj

2007-09-01

This paper provides a detailed rate-distortion performance comparison between JPEG2000, Microsoft HD Photo, and H.264/AVC High Profile 4:4:4 I-frame coding for high-resolution still images and high-definition (HD) 1080p video sequences. This work is an extension to our previous comparative study published in previous SPIE conferences [1, 2]. Here we further optimize all three codecs for compression performance. Coding simulations are performed on a set of large-format color images captured from mainstream digital cameras and 1080p HD video sequences commonly used for H.264/AVC standardization work. Overall, our experimental results show that all three codecs offer very similar coding performances at the high-quality, high-resolution setting. Differences tend to be data-dependent: JPEG2000 with the wavelet technology tends to be the best performer with smooth spatial data; H.264/AVC High-Profile with advanced spatial prediction modes tends to cope best with more complex visual content; Microsoft HD Photo tends to be the most consistent across the board. For the still-image data sets, JPEG2000 offers the best R-D performance gains (around 0.2 to 1 dB in peak signal-to-noise ratio) over H.264/AVC High-Profile intra coding and Microsoft HD Photo. For the 1080p video data set, all three codecs offer very similar coding performance. As in [1, 2], neither do we consider scalability nor complexity in this study (JPEG2000 is operating in non-scalable, but optimal performance mode).

Efficient random access high resolution region-of-interest (ROI) image retrieval using backward coding of wavelet trees (BCWT)

NASA Astrophysics Data System (ADS)

Corona, Enrique; Nutter, Brian; Mitra, Sunanda; Guo, Jiangling; Karp, Tanja

2008-03-01

Efficient retrieval of high quality Regions-Of-Interest (ROI) from high resolution medical images is essential for reliable interpretation and accurate diagnosis. Random access to high quality ROI from codestreams is becoming an essential feature in many still image compression applications, particularly in viewing diseased areas from large medical images. This feature is easier to implement in block based codecs because of the inherent spatial independency of the code blocks. This independency implies that the decoding order of the blocks is unimportant as long as the position for each is properly identified. In contrast, wavelet-tree based codecs naturally use some interdependency that exploits the decaying spectrum model of the wavelet coefficients. Thus one must keep track of the decoding order from level to level with such codecs. We have developed an innovative multi-rate image subband coding scheme using "Backward Coding of Wavelet Trees (BCWT)" which is fast, memory efficient, and resolution scalable. It offers far less complexity than many other existing codecs including both, wavelet-tree, and block based algorithms. The ROI feature in BCWT is implemented through a transcoder stage that generates a new BCWT codestream containing only the information associated with the user-defined ROI. This paper presents an efficient technique that locates a particular ROI within the BCWT coded domain, and decodes it back to the spatial domain. This technique allows better access and proper identification of pathologies in high resolution images since only a small fraction of the codestream is required to be transmitted and analyzed.

Digital CODEC for real-time processing of broadcast quality video signals at 1.8 bits/pixel

NASA Technical Reports Server (NTRS)

Shalkhauser, Mary JO; Whyte, Wayne A., Jr.

1989-01-01

Advances in very large-scale integration and recent work in the field of bandwidth efficient digital modulation techniques have combined to make digital video processing technically feasible and potentially cost competitive for broadcast quality television transmission. A hardware implementation was developed for a DPCM-based digital television bandwidth compression algorithm which processes standard NTSC composite color television signals and produces broadcast quality video in real time at an average of 1.8 bits/pixel. The data compression algorithm and the hardware implementation of the CODEC are described, and performance results are provided.

Digital CODEC for real-time processing of broadcast quality video signals at 1.8 bits/pixel

NASA Technical Reports Server (NTRS)

Shalkhauser, Mary JO; Whyte, Wayne A.

1991-01-01

Advances in very large scale integration and recent work in the field of bandwidth efficient digital modulation techniques have combined to make digital video processing technically feasible an potentially cost competitive for broadcast quality television transmission. A hardware implementation was developed for DPCM (differential pulse code midulation)-based digital television bandwidth compression algorithm which processes standard NTSC composite color television signals and produces broadcast quality video in real time at an average of 1.8 bits/pixel. The data compression algorithm and the hardware implementation of the codec are described, and performance results are provided.

Loss tolerant speech decoder for telecommunications

NASA Technical Reports Server (NTRS)

Prieto, Jr., Jaime L. (Inventor)

1999-01-01

A method and device for extrapolating past signal-history data for insertion into missing data segments in order to conceal digital speech frame errors. The extrapolation method uses past-signal history that is stored in a buffer. The method is implemented with a device that utilizes a finite-impulse response (FIR) multi-layer feed-forward artificial neural network that is trained by back-propagation for one-step extrapolation of speech compression algorithm (SCA) parameters. Once a speech connection has been established, the speech compression algorithm device begins sending encoded speech frames. As the speech frames are received, they are decoded and converted back into speech signal voltages. During the normal decoding process, pre-processing of the required SCA parameters will occur and the results stored in the past-history buffer. If a speech frame is detected to be lost or in error, then extrapolation modules are executed and replacement SCA parameters are generated and sent as the parameters required by the SCA. In this way, the information transfer to the SCA is transparent, and the SCA processing continues as usual. The listener will not normally notice that a speech frame has been lost because of the smooth transition between the last-received, lost, and next-received speech frames.

Analysis of glottal source parameters in Parkinsonian speech.

PubMed

Hanratty, Jane; Deegan, Catherine; Walsh, Mary; Kirkpatrick, Barry

2016-08-01

Diagnosis and monitoring of Parkinson's disease has a number of challenges as there is no definitive biomarker despite the broad range of symptoms. Research is ongoing to produce objective measures that can either diagnose Parkinson's or act as an objective decision support tool. Recent research on speech based measures have demonstrated promising results. This study aims to investigate the characteristics of the glottal source signal in Parkinsonian speech. An experiment is conducted in which a selection of glottal parameters are tested for their ability to discriminate between healthy and Parkinsonian speech. Results for each glottal parameter are presented for a database of 50 healthy speakers and a database of 16 speakers with Parkinsonian speech symptoms. Receiver operating characteristic (ROC) curves were employed to analyse the results and the area under the ROC curve (AUC) values were used to quantify the performance of each glottal parameter. The results indicate that glottal parameters can be used to discriminate between healthy and Parkinsonian speech, although results varied for each parameter tested. For the task of separating healthy and Parkinsonian speech, 2 out of the 7 glottal parameters tested produced AUC values of over 0.9.

Wavelet-based scalable L-infinity-oriented compression.

PubMed

Alecu, Alin; Munteanu, Adrian; Cornelis, Jan P H; Schelkens, Peter

2006-09-01

Among the different classes of coding techniques proposed in literature, predictive schemes have proven their outstanding performance in near-lossless compression. However, these schemes are incapable of providing embedded L(infinity)-oriented compression, or, at most, provide a very limited number of potential L(infinity) bit-stream truncation points. We propose a new multidimensional wavelet-based L(infinity)-constrained scalable coding framework that generates a fully embedded L(infinity)-oriented bit stream and that retains the coding performance and all the scalability options of state-of-the-art L2-oriented wavelet codecs. Moreover, our codec instantiation of the proposed framework clearly outperforms JPEG2000 in L(infinity) coding sense.

Toward objective image quality metrics: the AIC Eval Program of the JPEG

NASA Astrophysics Data System (ADS)

Richter, Thomas; Larabi, Chaker

2008-08-01

Objective quality assessment of lossy image compression codecs is an important part of the recent call of the JPEG for Advanced Image Coding. The target of the AIC ad-hoc group is twofold: First, to receive state-of-the-art still image codecs and to propose suitable technology for standardization; and second, to study objective image quality metrics to evaluate the performance of such codes. Even tthough the performance of an objective metric is defined by how well it predicts the outcome of a subjective assessment, one can also study the usefulness of a metric in a non-traditional way indirectly, namely by measuring the subjective quality improvement of a codec that has been optimized for a specific objective metric. This approach shall be demonstrated here on the recently proposed HDPhoto format14 introduced by Microsoft and a SSIM-tuned17 version of it by one of the authors. We compare these two implementations with JPEG1 in two variations and a visual and PSNR optimal JPEG200013 implementation. To this end, we use subjective and objective tests based on the multiscale SSIM and a new DCT based metric.

High-performance software-only H.261 video compression on PC

NASA Astrophysics Data System (ADS)

Kasperovich, Leonid

1996-03-01

This paper describes an implementation of a software H.261 codec for PC, that takes an advantage of the fast computational algorithms for DCT-based video compression, which have been presented by the author at the February's 1995 SPIE/IS&T meeting. The motivation for developing the H.261 prototype system is to demonstrate a feasibility of real time software- only videoconferencing solution to operate across a wide range of network bandwidth, frame rate, and resolution of the input video. As the bandwidths of current network technology will be increased, the higher frame rate and resolution of video to be transmitted is allowed, that requires, in turn, a software codec to be able to compress pictures of CIF (352 X 288) resolution at up to 30 frame/sec. Running on Pentium 133 MHz PC the codec presented is capable to compress video in CIF format at 21 - 23 frame/sec. This result is comparable to the known hardware-based H.261 solutions, but it doesn't require any specific hardware. The methods to achieve high performance, the program optimization technique for Pentium microprocessor along with the performance profile, showing the actual contribution of the different encoding/decoding stages to the overall computational process, are presented.

An Analysis of The Parameters Used In Speech ABR Assessment Protocols.

PubMed

Sanfins, Milaine D; Hatzopoulos, Stavros; Donadon, Caroline; Diniz, Thais A; Borges, Leticia R; Skarzynski, Piotr H; Colella-Santos, Maria Francisca

2018-04-01

The aim of this study was to assess the parameters of choice, such as duration, intensity, rate, polarity, number of sweeps, window length, stimulated ear, fundamental frequency, first formant, and second formant, from previously published speech ABR studies. To identify candidate articles, five databases were assessed using the following keyword descriptors: speech ABR, ABR-speech, speech auditory brainstem response, auditory evoked potential to speech, speech-evoked brainstem response, and complex sounds. The search identified 1288 articles published between 2005 and 2015. After filtering the total number of papers according to the inclusion and exclusion criteria, 21 studies were selected. Analyzing the protocol details used in 21 studies suggested that there is no consensus to date on a speech-ABR protocol and that the parameters of analysis used are quite variable between studies. This inhibits the wider generalization and extrapolation of data across languages and studies.

A magnetic resonance imaging study on the articulatory and acoustic speech parameters of Malay vowels

PubMed Central

2014-01-01

The phonetic properties of six Malay vowels are investigated using magnetic resonance imaging (MRI) to visualize the vocal tract in order to obtain dynamic articulatory parameters during speech production. To resolve image blurring due to the tongue movement during the scanning process, a method based on active contour extraction is used to track tongue contours. The proposed method efficiently tracks tongue contours despite the partial blurring of MRI images. Consequently, the articulatory parameters that are effectively measured as tongue movement is observed, and the specific shape of the tongue and its position for all six uttered Malay vowels are determined. Speech rehabilitation procedure demands some kind of visual perceivable prototype of speech articulation. To investigate the validity of the measured articulatory parameters based on acoustic theory of speech production, an acoustic analysis based on the uttered vowels by subjects has been performed. As the acoustic speech and articulatory parameters of uttered speech were examined, a correlation between formant frequencies and articulatory parameters was observed. The experiments reported a positive correlation between the constriction location of the tongue body and the first formant frequency, as well as a negative correlation between the constriction location of the tongue tip and the second formant frequency. The results demonstrate that the proposed method is an effective tool for the dynamic study of speech production. PMID:25060583

A magnetic resonance imaging study on the articulatory and acoustic speech parameters of Malay vowels.

PubMed

Zourmand, Alireza; Mirhassani, Seyed Mostafa; Ting, Hua-Nong; Bux, Shaik Ismail; Ng, Kwan Hoong; Bilgen, Mehmet; Jalaludin, Mohd Amin

2014-07-25

The phonetic properties of six Malay vowels are investigated using magnetic resonance imaging (MRI) to visualize the vocal tract in order to obtain dynamic articulatory parameters during speech production. To resolve image blurring due to the tongue movement during the scanning process, a method based on active contour extraction is used to track tongue contours. The proposed method efficiently tracks tongue contours despite the partial blurring of MRI images. Consequently, the articulatory parameters that are effectively measured as tongue movement is observed, and the specific shape of the tongue and its position for all six uttered Malay vowels are determined.Speech rehabilitation procedure demands some kind of visual perceivable prototype of speech articulation. To investigate the validity of the measured articulatory parameters based on acoustic theory of speech production, an acoustic analysis based on the uttered vowels by subjects has been performed. As the acoustic speech and articulatory parameters of uttered speech were examined, a correlation between formant frequencies and articulatory parameters was observed. The experiments reported a positive correlation between the constriction location of the tongue body and the first formant frequency, as well as a negative correlation between the constriction location of the tongue tip and the second formant frequency. The results demonstrate that the proposed method is an effective tool for the dynamic study of speech production.

Flexible High Speed Codec (FHSC)

NASA Technical Reports Server (NTRS)

Segallis, G. P.; Wernlund, J. V.

1991-01-01

The ongoing NASA/Harris Flexible High Speed Codec (FHSC) program is described. The program objectives are to design and build an encoder decoder that allows operation in either burst or continuous modes at data rates of up to 300 megabits per second. The decoder handles both hard and soft decision decoding and can switch between modes on a burst by burst basis. Bandspreading is low since the code rate is greater than or equal to 7/8. The encoder and a hard decision decoder fit on a single application specific integrated circuit (ASIC) chip. A soft decision applique is implemented using 300 K emitter coupled logic (ECL) which can be easily translated to an ECL gate array.

JPEG 2000-based compression of fringe patterns for digital holographic microscopy

NASA Astrophysics Data System (ADS)

Blinder, David; Bruylants, Tim; Ottevaere, Heidi; Munteanu, Adrian; Schelkens, Peter

2014-12-01

With the advent of modern computing and imaging technologies, digital holography is becoming widespread in various scientific disciplines such as microscopy, interferometry, surface shape measurements, vibration analysis, data encoding, and certification. Therefore, designing an efficient data representation technology is of particular importance. Off-axis holograms have very different signal properties with respect to regular imagery, because they represent a recorded interference pattern with its energy biased toward the high-frequency bands. This causes traditional images' coders, which assume an underlying 1/f2 power spectral density distribution, to perform suboptimally for this type of imagery. We propose a JPEG 2000-based codec framework that provides a generic architecture suitable for the compression of many types of off-axis holograms. This framework has a JPEG 2000 codec at its core, extended with (1) fully arbitrary wavelet decomposition styles and (2) directional wavelet transforms. Using this codec, we report significant improvements in coding performance for off-axis holography relative to the conventional JPEG 2000 standard, with Bjøntegaard delta-peak signal-to-noise ratio improvements ranging from 1.3 to 11.6 dB for lossy compression in the 0.125 to 2.00 bpp range and bit-rate reductions of up to 1.6 bpp for lossless compression.

NASA's mobile satellite communications program; ground and space segment technologies

NASA Technical Reports Server (NTRS)

Naderi, F.; Weber, W. J.; Knouse, G. H.

1984-01-01

This paper describes the Mobile Satellite Communications Program of the United States National Aeronautics and Space Administration (NASA). The program's objectives are to facilitate the deployment of the first generation commercial mobile satellite by the private sector, and to technologically enable future generations by developing advanced and high risk ground and space segment technologies. These technologies are aimed at mitigating severe shortages of spectrum, orbital slot, and spacecraft EIRP which are expected to plague the high capacity mobile satellite systems of the future. After a brief introduction of the concept of mobile satellite systems and their expected evolution, this paper outlines the critical ground and space segment technologies. Next, the Mobile Satellite Experiment (MSAT-X) is described. MSAT-X is the framework through which NASA will develop advanced ground segment technologies. An approach is outlined for the development of conformal vehicle antennas, spectrum and power-efficient speech codecs, and modulation techniques for use in the non-linear faded channels and efficient multiple access schemes. Finally, the paper concludes with a description of the current and planned NASA activities aimed at developing complex large multibeam spacecraft antennas needed for future generation mobile satellite systems.

An articulatorily constrained, maximum entropy approach to speech recognition and speech coding

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

Hogden, J.

Hidden Markov models (HMM`s) are among the most popular tools for performing computer speech recognition. One of the primary reasons that HMM`s typically outperform other speech recognition techniques is that the parameters used for recognition are determined by the data, not by preconceived notions of what the parameters should be. This makes HMM`s better able to deal with intra- and inter-speaker variability despite the limited knowledge of how speech signals vary and despite the often limited ability to correctly formulate rules describing variability and invariance in speech. In fact, it is often the case that when HMM parameter values aremore » constrained using the limited knowledge of speech, recognition performance decreases. However, the structure of an HMM has little in common with the mechanisms underlying speech production. Here, the author argues that by using probabilistic models that more accurately embody the process of speech production, he can create models that have all the advantages of HMM`s, but that should more accurately capture the statistical properties of real speech samples--presumably leading to more accurate speech recognition. The model he will discuss uses the fact that speech articulators move smoothly and continuously. Before discussing how to use articulatory constraints, he will give a brief description of HMM`s. This will allow him to highlight the similarities and differences between HMM`s and the proposed technique.« less

Multichannel Speech Enhancement Based on Generalized Gamma Prior Distribution with Its Online Adaptive Estimation

NASA Astrophysics Data System (ADS)

Dat, Tran Huy; Takeda, Kazuya; Itakura, Fumitada

We present a multichannel speech enhancement method based on MAP speech spectral magnitude estimation using a generalized gamma model of speech prior distribution, where the model parameters are adapted from actual noisy speech in a frame-by-frame manner. The utilization of a more general prior distribution with its online adaptive estimation is shown to be effective for speech spectral estimation in noisy environments. Furthermore, the multi-channel information in terms of cross-channel statistics are shown to be useful to better adapt the prior distribution parameters to the actual observation, resulting in better performance of speech enhancement algorithm. We tested the proposed algorithm in an in-car speech database and obtained significant improvements of the speech recognition performance, particularly under non-stationary noise conditions such as music, air-conditioner and open window.

Monaural room acoustic parameters from music and speech.

PubMed

Kendrick, Paul; Cox, Trevor J; Li, Francis F; Zhang, Yonggang; Chambers, Jonathon A

2008-07-01

This paper compares two methods for extracting room acoustic parameters from reverberated speech and music. An approach which uses statistical machine learning, previously developed for speech, is extended to work with music. For speech, reverberation time estimations are within a perceptual difference limen of the true value. For music, virtually all early decay time estimations are within a difference limen of the true value. The estimation accuracy is not good enough in other cases due to differences between the simulated data set used to develop the empirical model and real rooms. The second method carries out a maximum likelihood estimation on decay phases at the end of notes or speech utterances. This paper extends the method to estimate parameters relating to the balance of early and late energies in the impulse response. For reverberation time and speech, the method provides estimations which are within the perceptual difference limen of the true value. For other parameters such as clarity, the estimations are not sufficiently accurate due to the natural reverberance of the excitation signals. Speech is a better test signal than music because of the greater periods of silence in the signal, although music is needed for low frequency measurement.

Flexible high speed codec

NASA Technical Reports Server (NTRS)

Boyd, R. W.; Hartman, W. F.

1992-01-01

The project's objective is to develop an advanced high speed coding technology that provides substantial coding gains with limited bandwidth expansion for several common modulation types. The resulting technique is applicable to several continuous and burst communication environments. Decoding provides a significant gain with hard decisions alone and can utilize soft decision information when available from the demodulator to increase the coding gain. The hard decision codec will be implemented using a single application specific integrated circuit (ASIC) chip. It will be capable of coding and decoding as well as some formatting and synchronization functions at data rates up to 300 megabits per second (Mb/s). Code rate is a function of the block length and can vary from 7/8 to 15/16. Length of coded bursts can be any multiple of 32 that is greater than or equal to 256 bits. Coding may be switched in or out on a burst by burst basis with no change in the throughput delay. Reliability information in the form of 3-bit (8-level) soft decisions, can be exploited using applique circuitry around the hard decision codec. This applique circuitry will be discrete logic in the present contract. However, ease of transition to LSI is one of the design guidelines. Discussed here is the selected coding technique. Its application to some communication systems is described. Performance with 4, 8, and 16-ary Phase Shift Keying (PSK) modulation is also presented.

Advanced modulation technology development for earth station demodulator applications. Coded modulation system development

NASA Technical Reports Server (NTRS)

Miller, Susan P.; Kappes, J. Mark; Layer, David H.; Johnson, Peter N.

1990-01-01

A jointly optimized coded modulation system is described which was designed, built, and tested by COMSAT Laboratories for NASA LeRC which provides a bandwidth efficiency of 2 bits/s/Hz at an information rate of 160 Mbit/s. A high speed rate 8/9 encoder with a Viterbi decoder and an Octal PSK modem are used to achieve this. The BER performance is approximately 1 dB from the theoretically calculated value for this system at a BER of 5 E-7 under nominal conditions. The system operates in burst mode for downlink applications and tests have demonstrated very little degradation in performance with frequency and level offset. Unique word miss rate measurements were conducted which demonstrate reliable acquisition at low values of Eb/No. Codec self tests have verified the performance of this subsystem in a stand alone mode. The codec is capable of operation at a 200 Mbit/s information rate as demonstrated using a codec test set which introduces noise digitally. The measured performance is within 0.2 dB of the computer simulated predictions. A gate array implementation of the most time critical element of the high speed Viterbi decoder was completed. This gate array add-compare-select chip significantly reduces the power consumption and improves the manufacturability of the decoder. This chip has general application in the implementation of high speed Viterbi decoders.

Voice Quality Modelling for Expressive Speech Synthesis

PubMed Central

Socoró, Joan Claudi

2014-01-01

This paper presents the perceptual experiments that were carried out in order to validate the methodology of transforming expressive speech styles using voice quality (VoQ) parameters modelling, along with the well-known prosody (F 0, duration, and energy), from a neutral style into a number of expressive ones. The main goal was to validate the usefulness of VoQ in the enhancement of expressive synthetic speech in terms of speech quality and style identification. A harmonic plus noise model (HNM) was used to modify VoQ and prosodic parameters that were extracted from an expressive speech corpus. Perception test results indicated the improvement of obtained expressive speech styles using VoQ modelling along with prosodic characteristics. PMID:24587738

Psychovisual masks and intelligent streaming RTP techniques for the MPEG-4 standard

NASA Astrophysics Data System (ADS)

Mecocci, Alessandro; Falconi, Francesco

2003-06-01

In today multimedia audio-video communication systems, data compression plays a fundamental role by reducing the bandwidth waste and the costs of the infrastructures and equipments. Among the different compression standards, the MPEG-4 is becoming more and more accepted and widespread. Even if one of the fundamental aspects of this standard is the possibility of separately coding video objects (i.e. to separate moving objects from the background and adapt the coding strategy to the video content), currently implemented codecs work only at the full-frame level. In this way, many advantages of the flexible MPEG-4 syntax are missed. This lack is due both to the difficulties in properly segmenting moving objects in real scenes (featuring an arbitrary motion of the objects and of the acquisition sensor), and to the current use of these codecs, that are mainly oriented towards the market of DVD backups (a full-frame approach is enough for these applications). In this paper we propose a codec for MPEG-4 real-time object streaming, that codes separately the moving objects and the scene background. The proposed codec is capable of adapting its strategy during the transmission, by analysing the video currently transmitted and setting the coder parameters and modalities accordingly. For example, the background can be transmitted as a whole or by dividing it into "slightly-detailed" and "highly detailed" zones that are coded in different ways to reduce the bit-rate while preserving the perceived quality. The coder can automatically switch in real-time, from one modality to the other during the transmission, depending on the current video content. Psychovisual masks and other video-content based measurements have been used as inputs for a Self Learning Intelligent Controller (SLIC) that changes the parameters and the transmission modalities. The current implementation is based on the ISO 14496 standard code that allows Video Objects (VO) transmission (other Open Source Codes like: DivX, Xvid, and Cisco"s Mpeg-4IP, have been analyzed but, as for today, they do not support VO). The original code has been deeply modified to integrate the SLIC and to adapt it for real-time streaming. A personal RTP (Real Time Protocol) has been defined and a Client-Server application has been developed. The viewer can decode and demultiplex the stream in real-time, while adapting to the changing modalities adopted by the Server according to the current video content. The proposed codec works as follows: the image background is separated by means of a segmentation module and it is transmitted by means of a wavelet compression scheme similar to that used in the JPEG2000. The VO are coded separately and multiplexed with the background stream. At the receiver the stream is demultiplexed to obtain the background and the VO that are subsequently pasted together. The final quality depends on many factors, in particular: the quantization parameters, the Group Of Video Object (GOV) length, the GOV structure (i.e. the number of I-P-B VOP), the search area for motion compensation. These factors are strongly related to the following measurement parameters (that have been defined during the development): the Objects Apparent Size (OAS) in the scene, the Video Object Incidence factor (VOI), the temporal correlation (measured through the Normalized Mean SAD, NMSAD). The SLIC module analyzes the currently transmitted video and selects the most appropriate settings by choosing from a predefined set of transmission modalities. For example, in the case of a highly temporal correlated sequence, the number of B-VOP is increased to improve the compression ratio. The strategy for the selection of the number of B-VOP turns out to be very different from those reported in the literature for B-frames (adopted for MPEG-1 and MPEG-2), due to the different behaviour of the temporal correlation when limited only to moving objects. The SLIC module also decides how to transmit the background. In our implementation we adopted the Visual Brain theory i.e. the study of what the "psychic eye" can get from a scene. According to this theory, a Psychomask Image Analysis (PIA) module has been developed to extract the visually homogeneous regions of the background. The PIA module produces two complementary masks one for the visually low variance zones and one for the higly variable zones; these zones are compressed with different strategies and encoded into two multiplexed streams. From practical experiments it turned out that the separate coding is advantageous only if the low variance zones exceed 50% of the whole background area (due to the overhead given by the need of transmitting the zone masks). The SLIC module takes care of deciding the appropriate transmission modality by analyzing the results produced by the PIA module. The main features of this codec are: low bitrate, good image quality and coding speed. The current implementation runs in real-time on standard PC platforms, the major limitation being the fixed position of the acquisition sensor. This limitation is due to the difficulties in separating moving objects from the background when the acquisition sensor moves. Our current real-time segmentation module does not produce suitable results if the acquisition sensor moves (only slight oscillatory movements are tolerated). In any case, the system is particularly suitable for tele surveillance applications at low bit-rates, where the camera is usually fixed or alternates among some predetermined positions (our segmentation module is capable of accurately separate moving objects from the static background when the acquisition sensor stops, even if different scenes are seen as a result of the sensor displacements). Moreover, the proposed architecture is general, in the sense that when real-time, robust segmentation systems (capable of separating objects in real-time from the background while the sensor itself is moving) will be available, they can be easily integrated while leaving the rest of the system unchanged. Experimental results related to real sequences for traffic monitoring and for people tracking and afety control are reported and deeply discussed in the paper. The whole system has been implemented in standard ANSI C code and currently runs on standard PCs under Microsoft Windows operating system (Windows 2000 pro and Windows XP).

Design of UAV high resolution image transmission system

NASA Astrophysics Data System (ADS)

Gao, Qiang; Ji, Ming; Pang, Lan; Jiang, Wen-tao; Fan, Pengcheng; Zhang, Xingcheng

2017-02-01

In order to solve the problem of the bandwidth limitation of the image transmission system on UAV, a scheme with image compression technology for mini UAV is proposed, based on the requirements of High-definition image transmission system of UAV. The video codec standard H.264 coding module and key technology was analyzed and studied for UAV area video communication. Based on the research of high-resolution image encoding and decoding technique and wireless transmit method, The high-resolution image transmission system was designed on architecture of Android and video codec chip; the constructed system was confirmed by experimentation in laboratory, the bit-rate could be controlled easily, QoS is stable, the low latency could meets most applied requirement not only for military use but also for industrial applications.

Effect of classroom acoustics on the speech intelligibility of students.

PubMed

Rabelo, Alessandra Terra Vasconcelos; Santos, Juliana Nunes; Oliveira, Rafaella Cristina; Magalhães, Max de Castro

2014-01-01

To analyze the acoustic parameters of classrooms and the relationship among equivalent sound pressure level (Leq), reverberation time (T₃₀), the Speech Transmission Index (STI), and the performance of students in speech intelligibility testing. A cross-sectional descriptive study, which analyzed the acoustic performance of 18 classrooms in 9 public schools in Belo Horizonte, Minas Gerais, Brazil, was conducted. The following acoustic parameters were measured: Leq, T₃₀, and the STI. In the schools evaluated, a speech intelligibility test was performed on 273 students, 45.4% of whom were boys, with an average age of 9.4 years. The results of the speech intelligibility test were compared to the values of the acoustic parameters with the help of Student's t-test. The Leq, T₃₀, and STI tests were conducted in empty and furnished classrooms. Children showed better results in speech intelligibility tests conducted in classrooms with less noise, a lower T₃₀, and greater STI values. The majority of classrooms did not meet the recommended regulatory standards for good acoustic performance. Acoustic parameters have a direct effect on the speech intelligibility of students. Noise contributes to a decrease in their understanding of information presented orally, which can lead to negative consequences in their education and their social integration as future professionals.

Assessment of voice and speech symptoms in early Parkinson's disease by the Robertson dysarthria profile.

PubMed

Defazio, Giovanni; Guerrieri, Marta; Liuzzi, Daniele; Gigante, Angelo Fabio; di Nicola, Vincenzo

2016-03-01

Changes in voice and speech are thought to involve 75-90% of people with PD, but the impact of PD progression on voice/speech parameters is not well defined. In this study, we assessed voice/speech symptoms in 48 parkinsonian patients staging <3 on the modified Hoehn and Yahr scale and 37 healthy subjects using the Robertson dysarthria profile (a clinical-perceptual method exploring all components potentially involved in speech difficulties), the Voice handicap index (a validated measure of the impact of voice symptoms on quality of life) and the speech evaluation parameter contained in the Unified Parkinson's Disease Rating Scale part III (UPDRS-III). Accuracy and metric properties of the Robertson dysarthria profile were also measured. On Robertson dysarthria profile, all parkinsonian patients yielded lower scores than healthy control subjects. Differently, the Voice Handicap Index and the speech evaluation parameter contained in the UPDRS-III could detect speech/voice disturbances in 10 and 75% of PD patients, respectively. Validation procedure in Parkinson's disease patients showed that the Robertson dysarthria profile has acceptable reliability, satisfactory internal consistency and scaling assumptions, lack of floor and ceiling effects, and partial correlations with UPDRS-III and Voice Handicap Index. We concluded that speech/voice disturbances are widely identified by the Robertson dysarthria profile in early parkinsonian patients, even when the disturbances do not carry a significant level of disability. Robertson dysarthria profile may be a valuable tool to detect speech/voice disturbances in Parkinson's disease.

V2S: Voice to Sign Language Translation System for Malaysian Deaf People

NASA Astrophysics Data System (ADS)

Mean Foong, Oi; Low, Tang Jung; La, Wai Wan

The process of learning and understand the sign language may be cumbersome to some, and therefore, this paper proposes a solution to this problem by providing a voice (English Language) to sign language translation system using Speech and Image processing technique. Speech processing which includes Speech Recognition is the study of recognizing the words being spoken, regardless of whom the speaker is. This project uses template-based recognition as the main approach in which the V2S system first needs to be trained with speech pattern based on some generic spectral parameter set. These spectral parameter set will then be stored as template in a database. The system will perform the recognition process through matching the parameter set of the input speech with the stored templates to finally display the sign language in video format. Empirical results show that the system has 80.3% recognition rate.

Perceptual consequences of changes in vocoded speech parameters in various reverberation conditions.

PubMed

Drgas, Szymon; Blaszak, Magdalena A

2009-08-01

To study the perceptual consequences of changes in parameters of vocoded speech in various reverberation conditions. The 3 controlled variables were number of vocoder bands, instantaneous frequency change rate, and reverberation conditions. The effects were quantified in terms of (a) nonsense words' recognition scores for young normal-hearing listeners, (b) ease of listening based on the time of response (response delay), and (c) the subjective measure of difficulty (10-degree scale). It has been shown that the fine structure of a signal is a relevant cue in speech perception in reverberation conditions. The results obtained for different number of bands, frequency-modulation cutoff frequencies, and reverberation conditions have shown that all these parameters are important for speech perception in reverberation. Only slow variations in the instantaneous frequency (<50 Hz) seem to play a critical role in speech intelligibility in anechoic conditions. In reverberant enclosures, however, fast fluctuations of instantaneous frequency are also significant.

Evaluation of selected speech parameters after prosthesis supply in patients with maxillary or mandibular defects.

PubMed

Müller, Rainer; Höhlein, Andreas; Wolf, Annette; Markwardt, Jutta; Schulz, Matthias C; Range, Ursula; Reitemeier, Bernd

2013-01-01

Ablative surgery of oropharyngeal tumors frequently leads to defects in the speech organs, resulting in impairment of speech up to the point of unintelligibility. The aim of the present study was the assessment of selected parameters of speech with and without resection prostheses. The speech sounds of 22 patients suffering from maxillary and mandibular defects were recorded using a digital audio tape (DAT) recorder with and without resection prostheses. Evaluation of the resonance and the production of the sounds /s/, /sch/, and /ch/ was performed by 2 experienced speech therapists. Additionally, the patients completed a non-standardized questionnaire containing a linguistic self-assessment. After prosthesis supply, the number of patients with rhinophonia aperta decreased from 7 to 2 while the number of patients with intelligible speech increased from 2 to 20. Correct production of the sounds /s/, /sch/, and /ch/ increased from 2 to 13 patients. A significant improvement of the evaluated parameters could be observed only in patients with maxillary defects. The linguistic self-assessment showed a higher satisfaction in patients with maxillary defects. In patients with maxillary defects due to ablative tumor surgery, an increase in speech performance and intelligibility is possible by supplying resection prostheses. © 2013 S. Karger GmbH, Freiburg.

Recognition of speech in noise after application of time-frequency masks: Dependence on frequency and threshold parameters

PubMed Central

Sinex, Donal G.

2013-01-01

Binary time-frequency (TF) masks can be applied to separate speech from noise. Previous studies have shown that with appropriate parameters, ideal TF masks can extract highly intelligible speech even at very low speech-to-noise ratios (SNRs). Two psychophysical experiments provided additional information about the dependence of intelligibility on the frequency resolution and threshold criteria that define the ideal TF mask. Listeners identified AzBio Sentences in noise, before and after application of TF masks. Masks generated with 8 or 16 frequency bands per octave supported nearly-perfect identification. Word recognition accuracy was slightly lower and more variable with 4 bands per octave. When TF masks were generated with a local threshold criterion of 0 dB SNR, the mean speech reception threshold was −9.5 dB SNR, compared to −5.7 dB for unprocessed sentences in noise. Speech reception thresholds decreased by about 1 dB per dB of additional decrease in the local threshold criterion. Information reported here about the dependence of speech intelligibility on frequency and level parameters has relevance for the development of non-ideal TF masks for clinical applications such as speech processing for hearing aids. PMID:23556604

A Measure of the Auditory-perceptual Quality of Strain from Electroglottographic Analysis of Continuous Dysphonic Speech: Application to Adductor Spasmodic Dysphonia.

PubMed

Somanath, Keerthan; Mau, Ted

2016-11-01

(1) To develop an automated algorithm to analyze electroglottographic (EGG) signal in continuous dysphonic speech, and (2) to identify EGG waveform parameters that correlate with the auditory-perceptual quality of strain in the speech of patients with adductor spasmodic dysphonia (ADSD). Software development with application in a prospective controlled study. EGG was recorded from 12 normal speakers and 12 subjects with ADSD reading excerpts from the Rainbow Passage. Data were processed by a new algorithm developed with the specific goal of analyzing continuous dysphonic speech. The contact quotient, pulse width, a new parameter peak skew, and various contact closing slope quotient and contact opening slope quotient measures were extracted. EGG parameters were compared between normal and ADSD speech. Within the ADSD group, intra-subject comparison was also made between perceptually strained syllables and unstrained syllables. The opening slope quotient SO7525 distinguished strained syllables from unstrained syllables in continuous speech within individual subjects with ADSD. The standard deviations, but not the means, of contact quotient, EGGW50, peak skew, and SO7525 were different between normal and ADSD speakers. The strain-stress pattern in continuous speech can be visualized as color gradients based on the variation of EGG parameter values. EGG parameters may provide a within-subject measure of vocal strain and serve as a marker for treatment response. The addition of EGG to multidimensional assessment may lead to improved characterization of the voice disturbance in ADSD. Copyright © 2016 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

A measure of the auditory-perceptual quality of strain from electroglottographic analysis of continuous dysphonic speech: Application to adductor spasmodic dysphonia

PubMed Central

Somanath, Keerthan; Mau, Ted

2016-01-01

Objectives (1) To develop an automated algorithm to analyze electroglottographic (EGG) signal in continuous, dysphonic speech, and (2) to identify EGG waveform parameters that correlate with the auditory-perceptual quality of strain in the speech of patients with adductor spasmodic dysphonia (ADSD). Study Design Software development with application in a prospective controlled study. Methods EGG was recorded from 12 normal speakers and 12 subjects with ADSD reading excerpts from the Rainbow Passage. Data were processed by a new algorithm developed with the specific goal of analyzing continuous dysphonic speech. The contact quotient (CQ), pulse width (EGGW), a new parameter peak skew, and various contact closing slope quotient (SC) and contact opening slope quotient (SO) measures were extracted. EGG parameters were compared between normal and ADSD speech. Within the ADSD group, intra-subject comparison was also made between perceptually strained syllables and unstrained syllables. Results The opening slope quotient SO7525 distinguished strained syllables from unstrained syllables in continuous speech within individual ADSD subjects. The standard deviations, but not the means, of CQ, EGGW50, peak skew, and SO7525 were different between normal and ADSD speakers. The strain-stress pattern in continuous speech can be visualized as color gradients based on the variation of EGG parameter values. Conclusions EGG parameters may provide a within-subject measure of vocal strain and serve as a marker for treatment response. The addition of EGG to multi-dimensional assessment may lead to improved characterization of the voice disturbance in ADSD. PMID:26739857

Methods and apparatus for non-acoustic speech characterization and recognition

DOEpatents

Holzrichter, John F.

1999-01-01

By simultaneously recording EM wave reflections and acoustic speech information, the positions and velocities of the speech organs as speech is articulated can be defined for each acoustic speech unit. Well defined time frames and feature vectors describing the speech, to the degree required, can be formed. Such feature vectors can uniquely characterize the speech unit being articulated each time frame. The onset of speech, rejection of external noise, vocalized pitch periods, articulator conditions, accurate timing, the identification of the speaker, acoustic speech unit recognition, and organ mechanical parameters can be determined.

Methods and apparatus for non-acoustic speech characterization and recognition

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

Holzrichter, J.F.

By simultaneously recording EM wave reflections and acoustic speech information, the positions and velocities of the speech organs as speech is articulated can be defined for each acoustic speech unit. Well defined time frames and feature vectors describing the speech, to the degree required, can be formed. Such feature vectors can uniquely characterize the speech unit being articulated each time frame. The onset of speech, rejection of external noise, vocalized pitch periods, articulator conditions, accurate timing, the identification of the speaker, acoustic speech unit recognition, and organ mechanical parameters can be determined.

Vocoders and Speech Perception: Uses of Computer-Based Speech Analysis-Synthesis in Stimulus Generation.

ERIC Educational Resources Information Center

Tierney, Joseph; Mack, Molly

1987-01-01

Stimuli used in research on the perception of the speech signal have often been obtained from simple filtering and distortion of the speech waveform, sometimes accompanied by noise. However, for more complex stimulus generation, the parameters of speech can be manipulated, after analysis and before synthesis, using various types of algorithms to…

Low-dimensional recurrent neural network-based Kalman filter for speech enhancement.

PubMed

Xia, Youshen; Wang, Jun

2015-07-01

This paper proposes a new recurrent neural network-based Kalman filter for speech enhancement, based on a noise-constrained least squares estimate. The parameters of speech signal modeled as autoregressive process are first estimated by using the proposed recurrent neural network and the speech signal is then recovered from Kalman filtering. The proposed recurrent neural network is globally asymptomatically stable to the noise-constrained estimate. Because the noise-constrained estimate has a robust performance against non-Gaussian noise, the proposed recurrent neural network-based speech enhancement algorithm can minimize the estimation error of Kalman filter parameters in non-Gaussian noise. Furthermore, having a low-dimensional model feature, the proposed neural network-based speech enhancement algorithm has a much faster speed than two existing recurrent neural networks-based speech enhancement algorithms. Simulation results show that the proposed recurrent neural network-based speech enhancement algorithm can produce a good performance with fast computation and noise reduction. Copyright © 2015 Elsevier Ltd. All rights reserved.

Variable frame rate transmission - A review of methodology and application to narrow-band LPC speech coding

NASA Astrophysics Data System (ADS)

Viswanathan, V. R.; Makhoul, J.; Schwartz, R. M.; Huggins, A. W. F.

1982-04-01

The variable frame rate (VFR) transmission methodology developed, implemented, and tested in the years 1973-1978 for efficiently transmitting linear predictive coding (LPC) vocoder parameters extracted from the input speech at a fixed frame rate is reviewed. With the VFR method, parameters are transmitted only when their values have changed sufficiently over the interval since their preceding transmission. Two distinct approaches to automatic implementation of the VFR method are discussed. The first bases the transmission decisions on comparisons between the parameter values of the present frame and the last transmitted frame. The second, which is based on a functional perceptual model of speech, compares the parameter values of all the frames that lie in the interval between the present frame and the last transmitted frame against a linear model of parameter variation over that interval. Also considered is the application of VFR transmission to the design of narrow-band LPC speech coders with average bit rates of 2000-2400 bts/s.

Merlot Design

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

Ahern, S D

2003-06-10

We describe Merlot, a system for delivery of digital imagery over high speed networks. We describe various use cases, the client/server interaction, and the image and network codecs. We also describe some possible applications using Merlot and future work.

An Improved SEL Test of the ADV212 Video Codec

NASA Technical Reports Server (NTRS)

Wilcox, Edward P.; Campola, Michael J.; Nadendla, Seshagiri; Kadari, Madhusudhan; Gigliuto, Robert A.

2017-01-01

Single-event effect (SEE) test data is presented on the Analog Devices ADV212. Focus is given to the test setup used to improve data quality and validate single-event latch-up (SEL) protection circuitry.

An Improved SEL Test of the ADV212 Video Codec

NASA Technical Reports Server (NTRS)

Wilcox, Edward P; Campola, Michael J.; Nadendla, Seshagiri; Kadari, Madhusudhan; Gigliuto, Robert A.

2017-01-01

Single-event effect (SEE) test data is presented on the Analog Devices ADV212. Focus is given to the test setup used to improve data quality and validate single-event latchup (SEL) protection circuitry.

Digital codec for real-time processing of broadcast quality video signals at 1.8 bits/pixel

NASA Technical Reports Server (NTRS)

Shalkhauser, Mary JO; Whyte, Wayne A., Jr.

1989-01-01

The authors present the hardware implementation of a digital television bandwidth compression algorithm which processes standard NTSC (National Television Systems Committee) composite color television signals and produces broadcast-quality video in real time at an average of 1.8 b/pixel. The sampling rate used with this algorithm results in 768 samples over the active portion of each video line by 512 active video lines per video frame. The algorithm is based on differential pulse code modulation (DPCM), but additionally utilizes a nonadaptive predictor, nonuniform quantizer, and multilevel Huffman coder to reduce the data rate substantially below that achievable with straight DPCM. The nonadaptive predictor and multilevel Huffman coder combine to set this technique apart from prior-art DPCM encoding algorithms. The authors describe the data compression algorithm and the hardware implementation of the codec and provide performance results.

High-speed low-complexity video coding with EDiCTius: a DCT coding proposal for JPEG XS

NASA Astrophysics Data System (ADS)

Richter, Thomas; Fößel, Siegfried; Keinert, Joachim; Scherl, Christian

2017-09-01

In its 71th meeting, the JPEG committee issued a call for low complexity, high speed image coding, designed to address the needs of low-cost video-over-ip applications. As an answer to this call, Fraunhofer IIS and the Computing Center of the University of Stuttgart jointly developed an embedded DCT image codec requiring only minimal resources while maximizing throughput on FPGA and GPU implementations. Objective and subjective tests performed for the 73rd meeting confirmed its excellent performance and suitability for its purpose, and it was selected as one of the two key contributions for the development of a joined test model. In this paper, its authors describe the design principles of the codec, provide a high-level overview of the encoder and decoder chain and provide evaluation results on the test corpus selected by the JPEG committee.

The contrast between alveolar and velar stops with typical speech data: acoustic and articulatory analyses.

PubMed

Melo, Roberta Michelon; Mota, Helena Bolli; Berti, Larissa Cristina

2017-06-08

This study used acoustic and articulatory analyses to characterize the contrast between alveolar and velar stops with typical speech data, comparing the parameters (acoustic and articulatory) of adults and children with typical speech development. The sample consisted of 20 adults and 15 children with typical speech development. The analyzed corpus was organized through five repetitions of each target-word (/'kap ə/, /'tapə/, /'galo/ e /'daɾə/). These words were inserted into a carrier phrase and the participant was asked to name them spontaneously. Simultaneous audio and video data were recorded (tongue ultrasound images). The data was submitted to acoustic analyses (voice onset time; spectral peak and burst spectral moments; vowel/consonant transition and relative duration measures) and articulatory analyses (proportion of significant axes of the anterior and posterior tongue regions and description of tongue curves). Acoustic and articulatory parameters were effective to indicate the contrast between alveolar and velar stops, mainly in the adult group. Both speech analyses showed statistically significant differences between the two groups. The acoustic and articulatory parameters provided signals to characterize the phonic contrast of speech. One of the main findings in the comparison between adult and child speech was evidence of articulatory refinement/maturation even after the period of segment acquisition.

The design of a device for hearer and feeler differentiation, part A. [speech modulated hearing device

NASA Technical Reports Server (NTRS)

Creecy, R.

1974-01-01

A speech modulated white noise device is reported that gives the rhythmic characteristics of a speech signal for intelligible reception by deaf persons. The signal is composed of random amplitudes and frequencies as modulated by the speech envelope characteristics of rhythm and stress. Time intensity parameters of speech are conveyed through the vibro-tactile sensation stimuli.

New procedures to evaluate visually lossless compression for display systems

NASA Astrophysics Data System (ADS)

Stolitzka, Dale F.; Schelkens, Peter; Bruylants, Tim

2017-09-01

Visually lossless image coding in isochronous display streaming or plesiochronous networks reduces link complexity and power consumption and increases available link bandwidth. A new set of codecs developed within the last four years promise a new level of coding quality, but require new techniques that are sufficiently sensitive to the small artifacts or color variations induced by this new breed of codecs. This paper begins with a summary of the new ISO/IEC 29170-2, a procedure for evaluation of lossless coding and reports the new work by JPEG to extend the procedure in two important ways, for HDR content and for evaluating the differences between still images, panning images and image sequences. ISO/IEC 29170-2 relies on processing test images through a well-defined process chain for subjective, forced-choice psychophysical experiments. The procedure sets an acceptable quality level equal to one just noticeable difference. Traditional image and video coding evaluation techniques, such as, those used for television evaluation have not proven sufficiently sensitive to the small artifacts that may be induced by this breed of codecs. In 2015, JPEG received new requirements to expand evaluation of visually lossless coding for high dynamic range images, slowly moving images, i.e., panning, and image sequences. These requirements are the basis for new amendments of the ISO/IEC 29170-2 procedures described in this paper. These amendments promise to be highly useful for the new content in television and cinema mezzanine networks. The amendments passed the final ballot in April 2017 and are on track to be published in 2018.

Global Interoperability of High Definition Video Streams Via ACTS and Intelsat

NASA Technical Reports Server (NTRS)

Hsu, Eddie; Wang, Charles; Bergman, Larry; Pearman, James; Bhasin, Kul; Clark, Gilbert; Shopbell, Patrick; Gill, Mike; Tatsumi, Haruyuki; Kadowaki, Naoto

2000-01-01

In 1993, a proposal at the Japan.-U.S. Cooperation in Space Program Workshop lead to a subsequent series of satellite communications experiments and demonstrations, under the title of Trans-Pacific High Data Rate Satellite Communications Experiments. The first of which is a joint collaboration between government and industry teams in the United States and Japan that successfully demonstrated distributed high definition video (HDV) post-production on a global scale using a combination of high data rate satellites and terrestrial fiber optic asynchronous transfer mode (ATM) networks. The HDV experiment is the first GIBN experiment to establish a dual-hop broadband satellite link for the transmission of digital HDV over ATM. This paper describes the team's effort in using the NASA Advanced Communications Technology Satellite (ACTS) at rates up to OC-3 (155 Mbps) between Los Angeles and Honolulu, and using Intelsat at rates up to DS-3 (45 Mbps) between Kapolei and Tokyo, with which HDV source material was transmitted between Sony Pictures High Definition Center (SPHDC) in Los Angeles and Sony Visual Communication Center (VCC) in Shinagawa, Tokyo. The global-scale connection also used terrestrial networks in Japan, the States of Hawaii and California. The 1.2 Gbps digital HDV stream was compressed down to 22.5 Mbps using a proprietary Mitsubishi MPEG-2 codec that was ATM AAL-5 compatible. The codec: employed four-way parallel processing. Improved versions of the codec are now commercially available. The successful post-production activity performed in Tokyo with a HDV clip transmitted from Los Angeles was predicated on the seamless interoperation of all the equipment between the sites, and was an exciting example in deploying a global-scale information infrastructure involving a combination of broadband satellites and terrestrial fiber optic networks. Correlation of atmospheric effects with cell loss, codec drop-out, and picture quality were made. Current efforts in the Trans-Pacific series plan to examine the use of Internet Protocol (IP)-related technologies over such an infrastructure. The use of IP allows the general public to be an integral part of the exciting activities, helps to examine issues in constructing the solar-system internet, and affords an opportunity to tap the research results from the (reliable) multicast and distributed systems communities. The current Trans- Pacific projects, including remote astronomy and digital library (visible human) are briefly described.

Changes of some functional speech disorders after surgical correction of skeletal anterior open bite.

PubMed

Knez Ambrožič, Mojca; Hočevar Boltežar, Irena; Ihan Hren, Nataša

2015-09-01

Skeletal anterior open bite (AOB) or apertognathism is characterized by the absence of contact of the anterior teeth and affects articulation parameters, chewing, biting and voice quality. The treatment of AOB consists of orthognatic surgical procedures. The aim of this study was to evaluate the effects of treatment on voice quality, articulation and nasality in speech with respect to skeletal changes. The study was prospective; 15 patients with AOB were evaluated before and after surgery. Lateral cephalometric x-ray parameters (facial angle, interincisal distance, Wits appraisal) were measured to determine skeletal changes. Before surgery, nine patients still had articulation disorders despite speech therapy during childhood. The voice quality parameters were determined by acoustic analysis of the vowel sound /a/ (fundamental frequency-F0, jitter, shimmer). Spectral analysis of vowels /a/, /e/, /i/, /o/, /u/ was carried out by determining the mean frequency of the first (F1) and second (F2) formants. Nasality in speech was expressed as the ratio between the nasal and the oral sound energies during speech samples. After surgery, normalizations of facial skeletal parameters were observed in all patients, but no statistically significant changes in articulation and voice quality parameters occurred despite subjective observations of easier articulation. Any deterioration in velopharyngeal insufficiency was absent in all of the patients. In conclusion, the surgical treatment of skeletal AOB does not lead to deterioration in voice, resonance and articulation qualities. Despite surgical correction of the unfavourable skeletal situation of the speech apparatus, the pre-existing articulation disorder cannot improve without professional intervention.

Computer-Assisted Analysis of Spontaneous Speech: Quantification of Basic Parameters in Aphasic and Unimpaired Language

ERIC Educational Resources Information Center

Hussmann, Katja; Grande, Marion; Meffert, Elisabeth; Christoph, Swetlana; Piefke, Martina; Willmes, Klaus; Huber, Walter

2012-01-01

Although generally accepted as an important part of aphasia assessment, detailed analysis of spontaneous speech is rarely carried out in clinical practice mostly due to time limitations. The Aachener Sprachanalyse (ASPA; Aachen Speech Analysis) is a computer-assisted method for the quantitative analysis of German spontaneous speech that allows for…

Real-Time Control of an Articulatory-Based Speech Synthesizer for Brain Computer Interfaces

PubMed Central

Bocquelet, Florent; Hueber, Thomas; Girin, Laurent; Savariaux, Christophe; Yvert, Blaise

2016-01-01

Restoring natural speech in paralyzed and aphasic people could be achieved using a Brain-Computer Interface (BCI) controlling a speech synthesizer in real-time. To reach this goal, a prerequisite is to develop a speech synthesizer producing intelligible speech in real-time with a reasonable number of control parameters. We present here an articulatory-based speech synthesizer that can be controlled in real-time for future BCI applications. This synthesizer converts movements of the main speech articulators (tongue, jaw, velum, and lips) into intelligible speech. The articulatory-to-acoustic mapping is performed using a deep neural network (DNN) trained on electromagnetic articulography (EMA) data recorded on a reference speaker synchronously with the produced speech signal. This DNN is then used in both offline and online modes to map the position of sensors glued on different speech articulators into acoustic parameters that are further converted into an audio signal using a vocoder. In offline mode, highly intelligible speech could be obtained as assessed by perceptual evaluation performed by 12 listeners. Then, to anticipate future BCI applications, we further assessed the real-time control of the synthesizer by both the reference speaker and new speakers, in a closed-loop paradigm using EMA data recorded in real time. A short calibration period was used to compensate for differences in sensor positions and articulatory differences between new speakers and the reference speaker. We found that real-time synthesis of vowels and consonants was possible with good intelligibility. In conclusion, these results open to future speech BCI applications using such articulatory-based speech synthesizer. PMID:27880768

APEX/SPIN: a free test platform to measure speech intelligibility.

PubMed

Francart, Tom; Hofmann, Michael; Vanthornhout, Jonas; Van Deun, Lieselot; van Wieringen, Astrid; Wouters, Jan

2017-02-01

Measuring speech intelligibility in quiet and noise is important in clinical practice and research. An easy-to-use free software platform for conducting speech tests is presented, called APEX/SPIN. The APEX/SPIN platform allows the use of any speech material in combination with any noise. A graphical user interface provides control over a large range of parameters, such as number of loudspeakers, signal-to-noise ratio and parameters of the procedure. An easy-to-use graphical interface is provided for calibration and storage of calibration values. To validate the platform, perception of words in quiet and sentences in noise were measured both with APEX/SPIN and with an audiometer and CD player, which is a conventional setup in current clinical practice. Five normal-hearing listeners participated in the experimental evaluation. Speech perception results were similar for the APEX/SPIN platform and conventional procedures. APEX/SPIN is a freely available and open source platform that allows the administration of all kinds of custom speech perception tests and procedures.

Perceptual Consequences of Changes in Vocoded Speech Parameters in Various Reverberation Conditions

ERIC Educational Resources Information Center

Drgas, Szymon; Blaszak, Magdalena A.

2009-01-01

Purpose: To study the perceptual consequences of changes in parameters of vocoded speech in various reverberation conditions. Method: The 3 controlled variables were number of vocoder bands, instantaneous frequency change rate, and reverberation conditions. The effects were quantified in terms of (a) nonsense words' recognition scores for young…

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

None

Discussion Session - Accelerator System Design (Part II) Tutors: C. Darve, J. Weisend II, Ph. Lebrun, A. Dabrowski, U. Raich Video Conference with the CERN Control Center. Experts in the field of Accelerator science will be available to answer the students questions. This session will link the CCC and SA (using Codec VC).

Optimal bit allocation for hybrid scalable/multiple-description video transmission over wireless channels

NASA Astrophysics Data System (ADS)

Jubran, Mohammad K.; Bansal, Manu; Kondi, Lisimachos P.

2006-01-01

In this paper, we consider the problem of optimal bit allocation for wireless video transmission over fading channels. We use a newly developed hybrid scalable/multiple-description codec that combines the functionality of both scalable and multiple-description codecs. It produces a base layer and multiple-description enhancement layers. Any of the enhancement layers can be decoded (in a non-hierarchical manner) with the base layer to improve the reconstructed video quality. Two different channel coding schemes (Rate-Compatible Punctured Convolutional (RCPC)/Cyclic Redundancy Check (CRC) coding and, product code Reed Solomon (RS)+RCPC/CRC coding) are used for unequal error protection of the layered bitstream. Optimal allocation of the bitrate between source and channel coding is performed for discrete sets of source coding rates and channel coding rates. Experimental results are presented for a wide range of channel conditions. Also, comparisons with classical scalable coding show the effectiveness of using hybrid scalable/multiple-description coding for wireless transmission.

Block-based scalable wavelet image codec

NASA Astrophysics Data System (ADS)

Bao, Yiliang; Kuo, C.-C. Jay

1999-10-01

This paper presents a high performance block-based wavelet image coder which is designed to be of very low implementational complexity yet with rich features. In this image coder, the Dual-Sliding Wavelet Transform (DSWT) is first applied to image data to generate wavelet coefficients in fixed-size blocks. Here, a block only consists of wavelet coefficients from a single subband. The coefficient blocks are directly coded with the Low Complexity Binary Description (LCBiD) coefficient coding algorithm. Each block is encoded using binary context-based bitplane coding. No parent-child correlation is exploited in the coding process. There is also no intermediate buffering needed in between DSWT and LCBiD. The compressed bit stream generated by the proposed coder is both SNR and resolution scalable, as well as highly resilient to transmission errors. Both DSWT and LCBiD process the data in blocks whose size is independent of the size of the original image. This gives more flexibility in the implementation. The codec has a very good coding performance even the block size is (16,16).

Mixture block coding with progressive transmission in packet video. Appendix 1: Item 2. M.S. Thesis

NASA Technical Reports Server (NTRS)

Chen, Yun-Chung

1989-01-01

Video transmission will become an important part of future multimedia communication because of dramatically increasing user demand for video, and rapid evolution of coding algorithm and VLSI technology. Video transmission will be part of the broadband-integrated services digital network (B-ISDN). Asynchronous transfer mode (ATM) is a viable candidate for implementation of B-ISDN due to its inherent flexibility, service independency, and high performance. According to the characteristics of ATM, the information has to be coded into discrete cells which travel independently in the packet switching network. A practical realization of an ATM video codec called Mixture Block Coding with Progressive Transmission (MBCPT) is presented. This variable bit rate coding algorithm shows how a constant quality performance can be obtained according to user demand. Interactions between codec and network are emphasized including packetization, service synchronization, flow control, and error recovery. Finally, some simulation results based on MBCPT coding with error recovery are presented.

Local statistics adaptive entropy coding method for the improvement of H.26L VLC coding

NASA Astrophysics Data System (ADS)

Yoo, Kook-yeol; Kim, Jong D.; Choi, Byung-Sun; Lee, Yung Lyul

2000-05-01

In this paper, we propose an adaptive entropy coding method to improve the VLC coding efficiency of H.26L TML-1 codec. First of all, we will show that the VLC coding presented in TML-1 does not satisfy the sibling property of entropy coding. Then, we will modify the coding method into the local statistics adaptive one to satisfy the property. The proposed method based on the local symbol statistics dynamically changes the mapping relationship between symbol and bit pattern in the VLC table according to sibling property. Note that the codewords in the VLC table of TML-1 codec is not changed. Since this changed mapping relationship also derived in the decoder side by using the decoded symbols, the proposed VLC coding method does not require any overhead information. The simulation results show that the proposed method gives about 30% and 37% reduction in average bit rate for MB type and CBP information, respectively.

2-Step scalar deadzone quantization for bitplane image coding.

PubMed

Auli-Llinas, Francesc

2013-12-01

Modern lossy image coding systems generate a quality progressive codestream that, truncated at increasing rates, produces an image with decreasing distortion. Quality progressivity is commonly provided by an embedded quantizer that employs uniform scalar deadzone quantization (USDQ) together with a bitplane coding strategy. This paper introduces a 2-step scalar deadzone quantization (2SDQ) scheme that achieves same coding performance as that of USDQ while reducing the coding passes and the emitted symbols of the bitplane coding engine. This serves to reduce the computational costs of the codec and/or to code high dynamic range images. The main insights behind 2SDQ are the use of two quantization step sizes that approximate wavelet coefficients with more or less precision depending on their density, and a rate-distortion optimization technique that adjusts the distortion decreases produced when coding 2SDQ indexes. The integration of 2SDQ in current codecs is straightforward. The applicability and efficiency of 2SDQ are demonstrated within the framework of JPEG2000.

Statistical Analysis of Spectral Properties and Prosodic Parameters of Emotional Speech

NASA Astrophysics Data System (ADS)

Přibil, J.; Přibilová, A.

2009-01-01

The paper addresses reflection of microintonation and spectral properties in male and female acted emotional speech. Microintonation component of speech melody is analyzed regarding its spectral and statistical parameters. According to psychological research of emotional speech, different emotions are accompanied by different spectral noise. We control its amount by spectral flatness according to which the high frequency noise is mixed in voiced frames during cepstral speech synthesis. Our experiments are aimed at statistical analysis of cepstral coefficient values and ranges of spectral flatness in three emotions (joy, sadness, anger), and a neutral state for comparison. Calculated histograms of spectral flatness distribution are visually compared and modelled by Gamma probability distribution. Histograms of cepstral coefficient distribution are evaluated and compared using skewness and kurtosis. Achieved statistical results show good correlation comparing male and female voices for all emotional states portrayed by several Czech and Slovak professional actors.

The Interaction of Lexical Characteristics and Speech Production in Parkinson's Disease

ERIC Educational Resources Information Center

Chiu, Yi-Fang; Forrest, Karen

2017-01-01

Purpose: This study sought to investigate the interaction of speech movement execution with higher order lexical parameters. The authors examined how lexical characteristics affect speech output in individuals with Parkinson's disease (PD) and healthy control (HC) speakers. Method: Twenty speakers with PD and 12 healthy speakers read sentences…

Age and Function Differences in Shared Task Performance: Walking and Talking

ERIC Educational Resources Information Center

Williams, Kathleen; Hinton, Virginia A.; Bories, Tamara; Kovacs, Christopher R.

2006-01-01

Less is known about the effects of normal aging on speech output than other motor actions, because studies of communication integrity have focused on voice production and linguistic parameters rather than speech production characteristics. Studies investigating speech production in older adults have reported increased syllable duration (Slawinski,…

Immediate Effect of Alcohol on Voice Tremor Parameters and Speech Motor Control

ERIC Educational Resources Information Center

Krishnan, Gayathri; Ghosh, Vipin

2017-01-01

The complex neuro-muscular interplay of speech subsystems is susceptible to alcohol intoxication. Published reports have studied language formulation and fundamental frequency measures pre- and post-intoxication. This study aimed at tapping the speech motor control measure using rate, consistency, and accuracy measures of diadochokinesis and…

The Cleft Care UK study. Part 4: perceptual speech outcomes

PubMed Central

Sell, D; Mildinhall, S; Albery, L; Wills, A K; Sandy, J R; Ness, A R

2015-01-01

Structured Abstract Objectives To describe the perceptual speech outcomes from the Cleft Care UK (CCUK) study and compare them to the 1998 Clinical Standards Advisory Group (CSAG) audit. Setting and sample population A cross-sectional study of 248 children born with complete unilateral cleft lip and palate, between 1 April 2005 and 31 March 2007 who underwent speech assessment. Materials and methods Centre-based specialist speech and language therapists (SLT) took speech audio–video recordings according to nationally agreed guidelines. Two independent listeners undertook the perceptual analysis using the CAPS-A Audit tool. Intra- and inter-rater reliability were tested. Results For each speech parameter of intelligibility/distinctiveness, hypernasality, palatal/palatalization, backed to velar/uvular, glottal, weak and nasalized consonants, and nasal realizations, there was strong evidence that speech outcomes were better in the CCUK children compared to CSAG children. The parameters which did not show improvement were nasal emission, nasal turbulence, hyponasality and lateral/lateralization. Conclusion These results suggest that centralization of cleft care into high volume centres has resulted in improvements in UK speech outcomes in five-year-olds with unilateral cleft lip and palate. This may be associated with the development of a specialized workforce. Nevertheless, there still remains a group of children with significant difficulties at school entry. PMID:26567854

The Cleft Care UK study. Part 4: perceptual speech outcomes.

PubMed

Sell, D; Mildinhall, S; Albery, L; Wills, A K; Sandy, J R; Ness, A R

2015-11-01

To describe the perceptual speech outcomes from the Cleft Care UK (CCUK) study and compare them to the 1998 Clinical Standards Advisory Group (CSAG) audit. A cross-sectional study of 248 children born with complete unilateral cleft lip and palate, between 1 April 2005 and 31 March 2007 who underwent speech assessment. Centre-based specialist speech and language therapists (SLT) took speech audio-video recordings according to nationally agreed guidelines. Two independent listeners undertook the perceptual analysis using the CAPS-A Audit tool. Intra- and inter-rater reliability were tested. For each speech parameter of intelligibility/distinctiveness, hypernasality, palatal/palatalization, backed to velar/uvular, glottal, weak and nasalized consonants, and nasal realizations, there was strong evidence that speech outcomes were better in the CCUK children compared to CSAG children. The parameters which did not show improvement were nasal emission, nasal turbulence, hyponasality and lateral/lateralization. These results suggest that centralization of cleft care into high volume centres has resulted in improvements in UK speech outcomes in five-year-olds with unilateral cleft lip and palate. This may be associated with the development of a specialized workforce. Nevertheless, there still remains a group of children with significant difficulties at school entry. © The Authors. Orthodontics & Craniofacial Research Published by John Wiley & Sons Ltd.

Assessment of voice, speech, and related quality of life in advanced head and neck cancer patients 10-years+ after chemoradiotherapy.

PubMed

Kraaijenga, S A C; Oskam, I M; van Son, R J J H; Hamming-Vrieze, O; Hilgers, F J M; van den Brekel, M W M; van der Molen, L

2016-04-01

Assessment of long-term objective and subjective voice, speech, articulation, and quality of life in patients with head and neck cancer (HNC) treated with concurrent chemoradiotherapy (CRT) for advanced, stage IV disease. Twenty-two disease-free survivors, treated with cisplatin-based CRT for inoperable HNC (1999-2004), were evaluated at 10-years post-treatment. A standard Dutch text was recorded. Perceptual analysis of voice, speech, and articulation was conducted by two expert listeners (SLPs). Also an experimental expert system based on automatic speech recognition was used. Patients' perception of voice and speech and related quality of life was assessed with the Voice Handicap Index (VHI) and Speech Handicap Index (SHI) questionnaires. At a median follow-up of 11-years, perceptual evaluation showed abnormal scores in up to 64% of cases, depending on the outcome parameter analyzed. Automatic assessment of voice and speech parameters correlated moderate to strong with perceptual outcome scores. Patient-reported problems with voice (VHI>15) and speech (SHI>6) in daily life were present in 68% and 77% of patients, respectively. Patients treated with IMRT showed significantly less impairment compared to those treated with conventional radiotherapy. More than 10-years after organ-preservation treatment, voice and speech problems are common in this patient cohort, as assessed with perceptual evaluation, automatic speech recognition, and with validated structured questionnaires. There were fewer complaints in patients treated with IMRT than with conventional radiotherapy. Copyright © 2016 Elsevier Ltd. All rights reserved.

Optimal pattern synthesis for speech recognition based on principal component analysis

NASA Astrophysics Data System (ADS)

Korsun, O. N.; Poliyev, A. V.

2018-02-01

The algorithm for building an optimal pattern for the purpose of automatic speech recognition, which increases the probability of correct recognition, is developed and presented in this work. The optimal pattern forming is based on the decomposition of an initial pattern to principal components, which enables to reduce the dimension of multi-parameter optimization problem. At the next step the training samples are introduced and the optimal estimates for principal components decomposition coefficients are obtained by a numeric parameter optimization algorithm. Finally, we consider the experiment results that show the improvement in speech recognition introduced by the proposed optimization algorithm.

Putting Civility in Its Place--Free Speech in the Classroom

ERIC Educational Resources Information Center

Ben-Porath, Sigal R.

2017-01-01

Academic freedom gives professors broad discretion over expressions and interactions in the classroom. Free speech guidelines and First Amendment protections permit students to speak their minds too, but they offer very limited guidance as to how classrooms should operate. While professors should obviously work within free speech parameters in the…

Speech Spectrum's Correlation with Speakers' Eysenck Personality Traits

PubMed Central

Hu, Chao; Wang, Qiandong; Short, Lindsey A.; Fu, Genyue

2012-01-01

The current study explored the correlation between speakers' Eysenck personality traits and speech spectrum parameters. Forty-six subjects completed the Eysenck Personality Questionnaire. They were instructed to verbally answer the questions shown on a computer screen and their responses were recorded by the computer. Spectrum parameters of /sh/ and /i/ were analyzed by Praat voice software. Formant frequencies of the consonant /sh/ in lying responses were significantly lower than that in truthful responses, whereas no difference existed on the vowel /i/ speech spectrum. The second formant bandwidth of the consonant /sh/ speech spectrum was significantly correlated with the personality traits of Psychoticism, Extraversion, and Neuroticism, and the correlation differed between truthful and lying responses, whereas the first formant frequency of the vowel /i/ speech spectrum was negatively correlated with Neuroticism in both response types. The results suggest that personality characteristics may be conveyed through the human voice, although the extent to which these effects are due to physiological differences in the organs associated with speech or to a general Pygmalion effect is yet unknown. PMID:22439014

Audio visual speech source separation via improved context dependent association model

NASA Astrophysics Data System (ADS)

Kazemi, Alireza; Boostani, Reza; Sobhanmanesh, Fariborz

2014-12-01

In this paper, we exploit the non-linear relation between a speech source and its associated lip video as a source of extra information to propose an improved audio-visual speech source separation (AVSS) algorithm. The audio-visual association is modeled using a neural associator which estimates the visual lip parameters from a temporal context of acoustic observation frames. We define an objective function based on mean square error (MSE) measure between estimated and target visual parameters. This function is minimized for estimation of the de-mixing vector/filters to separate the relevant source from linear instantaneous or time-domain convolutive mixtures. We have also proposed a hybrid criterion which uses AV coherency together with kurtosis as a non-Gaussianity measure. Experimental results are presented and compared in terms of visually relevant speech detection accuracy and output signal-to-interference ratio (SIR) of source separation. The suggested audio-visual model significantly improves relevant speech classification accuracy compared to existing GMM-based model and the proposed AVSS algorithm improves the speech separation quality compared to reference ICA- and AVSS-based methods.

Changes in objective acoustic measurements and subjective voice complaints in call center customer-service advisors during one working day.

PubMed

Lehto, Laura; Laaksonen, Laura; Vilkman, Erkki; Alku, Paavo

2008-03-01

The aim of this study was to investigate how different acoustic parameters, extracted both from speech pressure waveforms and glottal flows, can be used in measuring vocal loading in modern working environments and how these parameters reflect the possible changes in the vocal function during a working day. In addition, correlations between objective acoustic parameters and subjective voice symptoms were addressed. The subjects were 24 female and 8 male customer-service advisors, who mainly use telephone during their working hours. Speech samples were recorded from continuous speech four times during a working day and voice symptom questionnaires were completed simultaneously. Among the various objective parameters, only F0 resulted in a statistically significant increase for both genders. No correlations between the changes in objective and subjective parameters appeared. However, the results encourage researchers within the field of occupational voice use to apply versatile measurement techniques in studying occupational voice loading.

Acoustic and spectral characteristics of young children's fricative productions: A developmental perspective

NASA Astrophysics Data System (ADS)

Nissen, Shawn L.; Fox, Robert Allen

2005-10-01

Scientists have made great strides toward understanding the mechanisms of speech production and perception. However, the complex relationships between the acoustic structures of speech and the resulting psychological percepts have yet to be fully and adequately explained, especially in speech produced by younger children. Thus, this study examined the acoustic structure of voiceless fricatives (/f, θ, s, /sh/) produced by adults and typically developing children from 3 to 6 years of age in terms of multiple acoustic parameters (durations, normalized amplitude, spectral slope, and spectral moments). It was found that the acoustic parameters of spectral slope and variance (commonly excluded from previous studies of child speech) were important acoustic parameters in the differentiation and classification of the voiceless fricatives, with spectral variance being the only measure to separate all four places of articulation. It was further shown that the sibilant contrast between /s/ and /sh/ was less distinguished in children than adults, characterized by a dramatic change in several spectral parameters at approximately five years of age. Discriminant analysis revealed evidence that classification models based on adult data were sensitive to these spectral differences in the five-year-old age group.

Acoustic characteristics of voice after severe traumatic brain injury.

PubMed

McHenry, M

2000-07-01

To describe the acoustic characteristics of voice in individuals with motor speech disorders after traumatic brain injury (TBI). Prospective study of 100 individuals with TBI based on consecutive referrals for motor speech evaluations. Subjects were audio tape-recorded while producing sustained vowels and single word and sentence intelligibility tests. Laryngeal airway resistance was estimated, and voice quality was rated perceptually. None of the subjects evidenced vocal parameters within normal limits. The most frequently occurring abnormal parameter across subjects was amplitude perturbation, followed by voice turbulence index. Twenty-three percent of subjects evidenced deviation in all five parameters measured. The perceptual ratings of breathiness were significantly correlated with both the amplitude perturbation quotient and the noise-to-harmonics ratio. Vocal quality deviation is common in motor speech disorders after TBI and may impact intelligibility.

Prediction of acoustic feature parameters using myoelectric signals.

PubMed

Lee, Ki-Seung

2010-07-01

It is well-known that a clear relationship exists between human voices and myoelectric signals (MESs) from the area of the speaker's mouth. In this study, we utilized this information to implement a speech synthesis scheme in which MES alone was used to predict the parameters characterizing the vocal-tract transfer function of specific speech signals. Several feature parameters derived from MES were investigated to find the optimal feature for maximization of the mutual information between the acoustic and the MES features. After the optimal feature was determined, an estimation rule for the acoustic parameters was proposed, based on a minimum mean square error (MMSE) criterion. In a preliminary study, 60 isolated words were used for both objective and subjective evaluations. The results showed that the average Euclidean distance between the original and predicted acoustic parameters was reduced by about 30% compared with the average Euclidean distance of the original parameters. The intelligibility of the synthesized speech signals using the predicted features was also evaluated. A word-level identification ratio of 65.5% and a syllable-level identification ratio of 73% were obtained through a listening test.

Speech profile of patients undergoing primary palatoplasty.

PubMed

Menegueti, Katia Ignacio; Mangilli, Laura Davison; Alonso, Nivaldo; Andrade, Claudia Regina Furquim de

2017-10-26

To characterize the profile and speech characteristics of patients undergoing primary palatoplasty in a Brazilian university hospital, considering the time of intervention (early, before two years of age; late, after two years of age). Participants were 97 patients of both genders with cleft palate and/or cleft and lip palate, assigned to the Speech-language Pathology Department, who had been submitted to primary palatoplasty and presented no prior history of speech-language therapy. Patients were divided into two groups: early intervention group (EIG) - 43 patients undergoing primary palatoplasty before 2 years of age and late intervention group (LIG) - 54 patients undergoing primary palatoplasty after 2 years of age. All patients underwent speech-language pathology assessment. The following parameters were assessed: resonance classification, presence of nasal turbulence, presence of weak intraoral air pressure, presence of audible nasal air emission, speech understandability, and compensatory articulation disorder (CAD). At statistical significance level of 5% (p≤0.05), no significant difference was observed between the groups in the following parameters: resonance classification (p=0.067); level of hypernasality (p=0.113), presence of nasal turbulence (p=0.179); presence of weak intraoral air pressure (p=0.152); presence of nasal air emission (p=0.369), and speech understandability (p=0.113). The groups differed with respect to presence of compensatory articulation disorders (p=0.020), with the LIG presenting higher occurrence of altered phonemes. It was possible to assess the general profile and speech characteristics of the study participants. Patients submitted to early primary palatoplasty present better speech profile.

Imitation of contrastive lexical stress in children with speech delay

NASA Astrophysics Data System (ADS)

Vick, Jennell C.; Moore, Christopher A.

2005-09-01

This study examined the relationship between acoustic correlates of stress in trochaic (strong-weak), spondaic (strong-strong), and iambic (weak-strong) nonword bisyllables produced by children (30-50) with normal speech acquisition and children with speech delay. Ratios comparing the acoustic measures (vowel duration, rms, and f0) of the first syllable to the second syllable were calculated to evaluate the extent to which each phonetic parameter was used to mark stress. In addition, a calculation of the variability of jaw movement in each bisyllable was made. Finally, perceptual judgments of accuracy of stress production were made. Analysis of perceptual judgments indicated a robust difference between groups: While both groups of children produced errors in imitating the contrastive lexical stress models (~40%), the children with normal speech acquisition tended to produce trochaic forms in substitution for other stress types, whereas children with speech delay showed no preference for trochees. The relationship between segmental acoustic parameters, kinematic variability, and the ratings of stress by trained listeners will be presented.

On the Use of Evolutionary Algorithms to Improve the Robustness of Continuous Speech Recognition Systems in Adverse Conditions

NASA Astrophysics Data System (ADS)

Selouani, Sid-Ahmed; O'Shaughnessy, Douglas

2003-12-01

Limiting the decrease in performance due to acoustic environment changes remains a major challenge for continuous speech recognition (CSR) systems. We propose a novel approach which combines the Karhunen-Loève transform (KLT) in the mel-frequency domain with a genetic algorithm (GA) to enhance the data representing corrupted speech. The idea consists of projecting noisy speech parameters onto the space generated by the genetically optimized principal axis issued from the KLT. The enhanced parameters increase the recognition rate for highly interfering noise environments. The proposed hybrid technique, when included in the front-end of an HTK-based CSR system, outperforms that of the conventional recognition process in severe interfering car noise environments for a wide range of signal-to-noise ratios (SNRs) varying from 16 dB to[InlineEquation not available: see fulltext.] dB. We also showed the effectiveness of the KLT-GA method in recognizing speech subject to telephone channel degradations.

A Speech Recognition-based Solution for the Automatic Detection of Mild Cognitive Impairment from Spontaneous Speech

PubMed Central

Tóth, László; Hoffmann, Ildikó; Gosztolya, Gábor; Vincze, Veronika; Szatlóczki, Gréta; Bánréti, Zoltán; Pákáski, Magdolna; Kálmán, János

2018-01-01

Background: Even today the reliable diagnosis of the prodromal stages of Alzheimer’s disease (AD) remains a great challenge. Our research focuses on the earliest detectable indicators of cognitive de-cline in mild cognitive impairment (MCI). Since the presence of language impairment has been reported even in the mild stage of AD, the aim of this study is to develop a sensitive neuropsychological screening method which is based on the analysis of spontaneous speech production during performing a memory task. In the future, this can form the basis of an Internet-based interactive screening software for the recognition of MCI. Methods: Participants were 38 healthy controls and 48 clinically diagnosed MCI patients. The provoked spontaneous speech by asking the patients to recall the content of 2 short black and white films (one direct, one delayed), and by answering one question. Acoustic parameters (hesitation ratio, speech tempo, length and number of silent and filled pauses, length of utterance) were extracted from the recorded speech sig-nals, first manually (using the Praat software), and then automatically, with an automatic speech recogni-tion (ASR) based tool. First, the extracted parameters were statistically analyzed. Then we applied machine learning algorithms to see whether the MCI and the control group can be discriminated automatically based on the acoustic features. Results: The statistical analysis showed significant differences for most of the acoustic parameters (speech tempo, articulation rate, silent pause, hesitation ratio, length of utterance, pause-per-utterance ratio). The most significant differences between the two groups were found in the speech tempo in the delayed recall task, and in the number of pauses for the question-answering task. The fully automated version of the analysis process – that is, using the ASR-based features in combination with machine learning - was able to separate the two classes with an F1-score of 78.8%. Conclusion: The temporal analysis of spontaneous speech can be exploited in implementing a new, auto-matic detection-based tool for screening MCI for the community. PMID:29165085

A Speech Recognition-based Solution for the Automatic Detection of Mild Cognitive Impairment from Spontaneous Speech.

PubMed

Toth, Laszlo; Hoffmann, Ildiko; Gosztolya, Gabor; Vincze, Veronika; Szatloczki, Greta; Banreti, Zoltan; Pakaski, Magdolna; Kalman, Janos

2018-01-01

Even today the reliable diagnosis of the prodromal stages of Alzheimer's disease (AD) remains a great challenge. Our research focuses on the earliest detectable indicators of cognitive decline in mild cognitive impairment (MCI). Since the presence of language impairment has been reported even in the mild stage of AD, the aim of this study is to develop a sensitive neuropsychological screening method which is based on the analysis of spontaneous speech production during performing a memory task. In the future, this can form the basis of an Internet-based interactive screening software for the recognition of MCI. Participants were 38 healthy controls and 48 clinically diagnosed MCI patients. The provoked spontaneous speech by asking the patients to recall the content of 2 short black and white films (one direct, one delayed), and by answering one question. Acoustic parameters (hesitation ratio, speech tempo, length and number of silent and filled pauses, length of utterance) were extracted from the recorded speech signals, first manually (using the Praat software), and then automatically, with an automatic speech recognition (ASR) based tool. First, the extracted parameters were statistically analyzed. Then we applied machine learning algorithms to see whether the MCI and the control group can be discriminated automatically based on the acoustic features. The statistical analysis showed significant differences for most of the acoustic parameters (speech tempo, articulation rate, silent pause, hesitation ratio, length of utterance, pause-per-utterance ratio). The most significant differences between the two groups were found in the speech tempo in the delayed recall task, and in the number of pauses for the question-answering task. The fully automated version of the analysis process - that is, using the ASR-based features in combination with machine learning - was able to separate the two classes with an F1-score of 78.8%. The temporal analysis of spontaneous speech can be exploited in implementing a new, automatic detection-based tool for screening MCI for the community. Copyright© Bentham Science Publishers; For any queries, please email at epub@benthamscience.org.

Deep Brain Stimulation of the Subthalamic Nucleus Parameter Optimization for Vowel Acoustics and Speech Intelligibility in Parkinson's Disease

ERIC Educational Resources Information Center

Knowles, Thea; Adams, Scott; Abeyesekera, Anita; Mancinelli, Cynthia; Gilmore, Greydon; Jog, Mandar

2018-01-01

Purpose: The settings of 3 electrical stimulation parameters were adjusted in 12 speakers with Parkinson's disease (PD) with deep brain stimulation of the subthalamic nucleus (STN-DBS) to examine their effects on vowel acoustics and speech intelligibility. Method: Participants were tested under permutations of low, mid, and high STN-DBS frequency,…

Perceptual Speech Assessment After Anterior Maxillary Distraction in Patients With Cleft Maxillary Hypoplasia.

PubMed

Richardson, Sunil; Seelan, Nikkie S; Selvaraj, Dhivakar; Khandeparker, Rakshit V; Gnanamony, Sangeetha

2016-06-01

To assess speech outcomes after anterior maxillary distraction (AMD) in patients with cleft-related maxillary hypoplasia. Fifty-eight patients at least 10 years old with cleft-related maxillary hypoplasia were included in this study irrespective of gender, type of cleft lip and palate, and amount of required advancement. AMD was carried out in all patients using a tooth-borne palatal distractor by a single oral and maxillofacial surgeon. Perceptual speech assessment was performed by 2 speech language pathologists preoperatively, before placement of the distractor device, and 6 months postoperatively using the scoring system of Perkins et al (Plast Reconstr Surg 116:72, 2005); the system evaluates velopharyngeal insufficiency (VPI), resonance, nasal air emission, articulation errors, and intelligibility. The data obtained were tabulated and subjected to statistical analysis using Wilcoxon signed rank test. A P value less than .05 was considered significant. Eight patients were lost to follow-up. At 6-month follow-up, improvements of 62% (n = 31), 64% (n = 32), 50% (n = 25), 68% (n = 34), and 70% (n = 35) in VPI, resonance, nasal air emission, articulation, and intelligibility, respectively, were observed, with worsening of all parameters in 1 patient (2%). The results for all tested parameters were highly significant (P ≤ .001). AMD offers a substantial improvement in speech for all 5 parameters of perceptual speech assessment. Copyright © 2016 The American Association of Oral and Maxillofacial Surgeons. Published by Elsevier Inc. All rights reserved.

Utility and accuracy of perceptual voice and speech distinctions in the diagnosis of Parkinson's disease, PSP and MSA-P.

PubMed

Miller, Nick; Nath, Uma; Noble, Emma; Burn, David

2017-06-01

To determine if perceptual speech measures distinguish people with Parkinson's disease (PD), multiple system atrophy with predominant parkinsonism (MSA-P) and progressive supranuclear palsy (PSP). Speech-language therapists blind to patient characteristics employed clinical rating scales to evaluate speech/voice in 24 people with clinically diagnosed PD, 17 with PSP and 9 with MSA-P, matched for disease duration (mean 4.9 years, standard deviation 2.2). No consistent intergroup differences appeared on specific speech/voice variables. People with PD were significantly less impaired on overall speech/voice severity. Analyses by severity suggested further investigation around laryngeal, resonance and fluency changes may characterize individual groups. MSA-P and PSP compared with PD were distinguished by severity of speech/voice deterioration, but individual speech/voice parameters failed to consistently differentiate groups.

Speech emotion recognition methods: A literature review

NASA Astrophysics Data System (ADS)

Basharirad, Babak; Moradhaseli, Mohammadreza

2017-10-01

Recently, attention of the emotional speech signals research has been boosted in human machine interfaces due to availability of high computation capability. There are many systems proposed in the literature to identify the emotional state through speech. Selection of suitable feature sets, design of a proper classifications methods and prepare an appropriate dataset are the main key issues of speech emotion recognition systems. This paper critically analyzed the current available approaches of speech emotion recognition methods based on the three evaluating parameters (feature set, classification of features, accurately usage). In addition, this paper also evaluates the performance and limitations of available methods. Furthermore, it highlights the current promising direction for improvement of speech emotion recognition systems.

Deep neural network and noise classification-based speech enhancement

NASA Astrophysics Data System (ADS)

Shi, Wenhua; Zhang, Xiongwei; Zou, Xia; Han, Wei

2017-07-01

In this paper, a speech enhancement method using noise classification and Deep Neural Network (DNN) was proposed. Gaussian mixture model (GMM) was employed to determine the noise type in speech-absent frames. DNN was used to model the relationship between noisy observation and clean speech. Once the noise type was determined, the corresponding DNN model was applied to enhance the noisy speech. GMM was trained with mel-frequency cepstrum coefficients (MFCC) and the parameters were estimated with an iterative expectation-maximization (EM) algorithm. Noise type was updated by spectrum entropy-based voice activity detection (VAD). Experimental results demonstrate that the proposed method could achieve better objective speech quality and smaller distortion under stationary and non-stationary conditions.

Development of an algorithm for improving quality and information processing capacity of MathSpeak synthetic speech renderings.

PubMed

Isaacson, M D; Srinivasan, S; Lloyd, L L

2010-01-01

MathSpeak is a set of rules for non speaking of mathematical expressions. These rules have been incorporated into a computerised module that translates printed mathematics into the non-ambiguous MathSpeak form for synthetic speech rendering. Differences between individual utterances produced with the translator module are difficult to discern because of insufficient pausing between utterances; hence, the purpose of this study was to develop an algorithm for improving the synthetic speech rendering of MathSpeak. To improve synthetic speech renderings, an algorithm for inserting pauses was developed based upon recordings of middle and high school math teachers speaking mathematic expressions. Efficacy testing of this algorithm was conducted with college students without disabilities and high school/college students with visual impairments. Parameters measured included reception accuracy, short-term memory retention, MathSpeak processing capacity and various rankings concerning the quality of synthetic speech renderings. All parameters measured showed statistically significant improvements when the algorithm was used. The algorithm improves the quality and information processing capacity of synthetic speech renderings of MathSpeak. This increases the capacity of individuals with print disabilities to perform mathematical activities and to successfully fulfill science, technology, engineering and mathematics academic and career objectives.

Tone classification of syllable-segmented Thai speech based on multilayer perception

NASA Astrophysics Data System (ADS)

Satravaha, Nuttavudh; Klinkhachorn, Powsiri; Lass, Norman

2002-05-01

Thai is a monosyllabic tonal language that uses tone to convey lexical information about the meaning of a syllable. Thus to completely recognize a spoken Thai syllable, a speech recognition system not only has to recognize a base syllable but also must correctly identify a tone. Hence, tone classification of Thai speech is an essential part of a Thai speech recognition system. Thai has five distinctive tones (``mid,'' ``low,'' ``falling,'' ``high,'' and ``rising'') and each tone is represented by a single fundamental frequency (F0) pattern. However, several factors, including tonal coarticulation, stress, intonation, and speaker variability, affect the F0 pattern of a syllable in continuous Thai speech. In this study, an efficient method for tone classification of syllable-segmented Thai speech, which incorporates the effects of tonal coarticulation, stress, and intonation, as well as a method to perform automatic syllable segmentation, were developed. Acoustic parameters were used as the main discriminating parameters. The F0 contour of a segmented syllable was normalized by using a z-score transformation before being presented to a tone classifier. The proposed system was evaluated on 920 test utterances spoken by 8 speakers. A recognition rate of 91.36% was achieved by the proposed system.

78 FR 57648 - Notice of Issuance of Final Determination Concerning Video Teleconferencing Server

Federal Register 2010, 2011, 2012, 2013, 2014

2013-09-19

... the Chinese- origin Video Board and the Filter Board, impart the essential character to the video... includes the codec; a network filter electronic circuit board (``Filter Board''); a housing case; a power... (``Linux software''). The Linux software allows the Filter Board to inspect each Ethernet packet of...

Objective and subjective quality assessment of geometry compression of reconstructed 3D humans in a 3D virtual room

NASA Astrophysics Data System (ADS)

Mekuria, Rufael; Cesar, Pablo; Doumanis, Ioannis; Frisiello, Antonella

2015-09-01

Compression of 3D object based video is relevant for 3D Immersive applications. Nevertheless, the perceptual aspects of the degradation introduced by codecs for meshes and point clouds are not well understood. In this paper we evaluate the subjective and objective degradations introduced by such codecs in a state of art 3D immersive virtual room. In the 3D immersive virtual room, users are captured with multiple cameras, and their surfaces are reconstructed as photorealistic colored/textured 3D meshes or point clouds. To test the perceptual effect of compression and transmission, we render degraded versions with different frame rates in different contexts (near/far) in the scene. A quantitative subjective study with 16 users shows that negligible distortion of decoded surfaces compared to the original reconstructions can be achieved in the 3D virtual room. In addition, a qualitative task based analysis in a full prototype field trial shows increased presence, emotion, user and state recognition of the reconstructed 3D Human representation compared to animated computer avatars.

Highly efficient codec based on significance-linked connected-component analysis of wavelet coefficients

NASA Astrophysics Data System (ADS)

Chai, Bing-Bing; Vass, Jozsef; Zhuang, Xinhua

1997-04-01

Recent success in wavelet coding is mainly attributed to the recognition of importance of data organization. There has been several very competitive wavelet codecs developed, namely, Shapiro's Embedded Zerotree Wavelets (EZW), Servetto et. al.'s Morphological Representation of Wavelet Data (MRWD), and Said and Pearlman's Set Partitioning in Hierarchical Trees (SPIHT). In this paper, we propose a new image compression algorithm called Significant-Linked Connected Component Analysis (SLCCA) of wavelet coefficients. SLCCA exploits both within-subband clustering of significant coefficients and cross-subband dependency in significant fields. A so-called significant link between connected components is designed to reduce the positional overhead of MRWD. In addition, the significant coefficients' magnitude are encoded in bit plane order to match the probability model of the adaptive arithmetic coder. Experiments show that SLCCA outperforms both EZW and MRWD, and is tied with SPIHT. Furthermore, it is observed that SLCCA generally has the best performance on images with large portion of texture. When applied to fingerprint image compression, it outperforms FBI's wavelet scalar quantization by about 1 dB.

Secure transport and adaptation of MC-EZBC video utilizing H.264-based transport protocols☆

PubMed Central

Hellwagner, Hermann; Hofbauer, Heinz; Kuschnig, Robert; Stütz, Thomas; Uhl, Andreas

2012-01-01

Universal Multimedia Access (UMA) calls for solutions where content is created once and subsequently adapted to given requirements. With regard to UMA and scalability, which is required often due to a wide variety of end clients, the best suited codecs are wavelet based (like the MC-EZBC) due to their inherent high number of scaling options. However, most transport technologies for delivering videos to end clients are targeted toward the H.264/AVC standard or, if scalability is required, the H.264/SVC. In this paper we will introduce a mapping of the MC-EZBC bitstream to existing H.264/SVC based streaming and scaling protocols. This enables the use of highly scalable wavelet based codecs on the one hand and the utilization of already existing network technologies without accruing high implementation costs on the other hand. Furthermore, we will evaluate different scaling options in order to choose the best option for given requirements. Additionally, we will evaluate different encryption options based on transport and bitstream encryption for use cases where digital rights management is required. PMID:26869746

High-precision two-way optic-fiber time transfer using an improved time code.

PubMed

Wu, Guiling; Hu, Liang; Zhang, Hao; Chen, Jianping

2014-11-01

We present a novel high-precision two-way optic-fiber time transfer scheme. The Inter-Range Instrumentation Group (IRIG-B) time code is modified by increasing bit rate and defining new fields. The modified time code can be transmitted directly using commercial optical transceivers and is able to efficiently suppress the effect of the Rayleigh backscattering in the optical fiber. A dedicated codec (encoder and decoder) with low delay fluctuation is developed. The synchronization issue is addressed by adopting a mask technique and combinational logic circuit. Its delay fluctuation is less than 27 ps in terms of the standard deviation. The two-way optic-fiber time transfer using the improved codec scheme is verified experimentally over 2 m to100 km fiber links. The results show that the stability over 100 km fiber link is always less than 35 ps with the minimum value of about 2 ps at the averaging time around 1000 s. The uncertainty of time difference induced by the chromatic dispersion over 100 km is less than 22 ps.

Predicting chroma from luma with frequency domain intra prediction

NASA Astrophysics Data System (ADS)

Egge, Nathan E.; Valin, Jean-Marc

2015-03-01

This paper describes a technique for performing intra prediction of the chroma planes based on the reconstructed luma plane in the frequency domain. This prediction exploits the fact that while RGB to YUV color conversion has the property that it decorrelates the color planes globally across an image, there is still some correlation locally at the block level.1 Previous proposals compute a linear model of the spatial relationship between the luma plane (Y) and the two chroma planes (U and V).2 In codecs that use lapped transforms this is not possible since transform support extends across the block boundaries3 and thus neighboring blocks are unavailable during intra- prediction. We design a frequency domain intra predictor for chroma that exploits the same local correlation with lower complexity than the spatial predictor and which works with lapped transforms. We then describe a low- complexity algorithm that directly uses luma coefficients as a chroma predictor based on gain-shape quantization and band partitioning. An experiment is performed that compares these two techniques inside the experimental Daala video codec and shows the lower complexity algorithm to be a better chroma predictor.

A forward error correction technique using a high-speed, high-rate single chip codec

NASA Astrophysics Data System (ADS)

Boyd, R. W.; Hartman, W. F.; Jones, Robert E.

The authors describe an error-correction coding approach that allows operation in either burst or continuous modes at data rates of multiple hundreds of megabits per second. Bandspreading is low since the code rate is 7/8 or greater, which is consistent with high-rate link operation. The encoder, along with a hard-decision decoder, fits on a single application-specific integrated circuit (ASIC) chip. Soft-decision decoding is possible utilizing applique hardware in conjunction with the hard-decision decoder. Expected coding gain is a function of the application and is approximately 2.5 dB for hard-decision decoding at 10-5 bit-error rate with phase-shift-keying modulation and additive Gaussian white noise interference. The principal use envisioned for this technique is to achieve a modest amount of coding gain on high-data-rate, bandwidth-constrained channels. Data rates of up to 300 Mb/s can be accommodated by the codec chip. The major objective is burst-mode communications, where code words are composed of 32 n data bits followed by 32 overhead bits.

Chinese speech intelligibility and its relationship with the speech transmission index for children in elementary school classrooms.

PubMed

Peng, Jianxin; Yan, Nanjie; Wang, Dan

2015-01-01

The present study investigated Chinese speech intelligibility in 28 classrooms from nine different elementary schools in Guangzhou, China. The subjective Chinese speech intelligibility in the classrooms was evaluated with children in grades 2, 4, and 6 (7 to 12 years old). Acoustical measurements were also performed in these classrooms. Subjective Chinese speech intelligibility scores and objective speech intelligibility parameters, such as speech transmission index (STI), were obtained at each listening position for all tests. The relationship between subjective Chinese speech intelligibility scores and STI was revealed and analyzed. The effects of age on Chinese speech intelligibility scores were compared. Results indicate high correlations between subjective Chinese speech intelligibility scores and STI for grades 2, 4, and 6 children. Chinese speech intelligibility scores increase with increase of age under the same STI condition. The differences in scores among different age groups decrease as STI increases. To achieve 95% Chinese speech intelligibility scores, the STIs required for grades 2, 4, and 6 children are 0.75, 0.69, and 0.63, respectively.

[Effects of acaoustic adaptation of classrooms on the quality of verbal communication].

PubMed

Mikulski, Witold

2013-01-01

Voice organ disorders among teachers are caused by excessive voice strain. One of the measures to reduce this strain is to decrease background noise when teaching. Increasing the acoustic absorption of the room is a technical measure for achieving this aim. The absorption level also improves speech intelligibility rated by the following parameters: room reverberation time and speech transmission index (STI). This article presents the effects of acoustic adaptation of classrooms on the quality of verbal communication, aimed at getting the speech intelligibility at the good or excellent level. The article lists the criteria for evaluating classrooms in terms of the quality of verbal communication. The parameters were defined, using the measurement methods according to PN-EN ISO 3382-2:2010 and PN-EN 60268-16:2011. Acoustic adaptations were completed in two classrooms. After completing acoustic adaptations the reverberation time for the frequency of 1 kHz was reduced: in room no. 1 from 1.45 s to 0.44 s and in room no. 2 from 1.03 s to 0.37 s (maximum 0.65 s). At the same time, the speech transmission index increased: in room no. 1 from 0.55 (satisfactory speech intelligibility) to 0.75 (speech intelligibility close to excellent); in room no. 2 from 0.63 (good speech intelligibility) to 0.80 (excellent speech intelligibility). Therefore, it can be stated that prior to completing acoustic adaptations room no. 1 did not comply and room no. 2 barely complied with the criterion (speech transmission index of 0.62). After completing acoustic adaptations both rooms meet the requirements.

Statistical properties of Chinese phonemic networks

NASA Astrophysics Data System (ADS)

Yu, Shuiyuan; Liu, Haitao; Xu, Chunshan

2011-04-01

The study of properties of speech sound systems is of great significance in understanding the human cognitive mechanism and the working principles of speech sound systems. Some properties of speech sound systems, such as the listener-oriented feature and the talker-oriented feature, have been unveiled with the statistical study of phonemes in human languages and the research of the interrelations between human articulatory gestures and the corresponding acoustic parameters. With all the phonemes of speech sound systems treated as a coherent whole, our research, which focuses on the dynamic properties of speech sound systems in operation, investigates some statistical parameters of Chinese phoneme networks based on real text and dictionaries. The findings are as follows: phonemic networks have high connectivity degrees and short average distances; the degrees obey normal distribution and the weighted degrees obey power law distribution; vowels enjoy higher priority than consonants in the actual operation of speech sound systems; the phonemic networks have high robustness against targeted attacks and random errors. In addition, for investigating the structural properties of a speech sound system, a statistical study of dictionaries is conducted, which shows the higher frequency of shorter words and syllables and the tendency that the longer a word is, the shorter the syllables composing it are. From these structural properties and dynamic properties one can derive the following conclusion: the static structure of a speech sound system tends to promote communication efficiency and save articulation effort while the dynamic operation of this system gives preference to reliable transmission and easy recognition. In short, a speech sound system is an effective, efficient and reliable communication system optimized in many aspects.

Hearing AIDS and music.

PubMed

Chasin, Marshall; Russo, Frank A

2004-01-01

Historically, the primary concern for hearing aid design and fitting is optimization for speech inputs. However, increasingly other types of inputs are being investigated and this is certainly the case for music. Whether the hearing aid wearer is a musician or merely someone who likes to listen to music, the electronic and electro-acoustic parameters described can be optimized for music as well as for speech. That is, a hearing aid optimally set for music can be optimally set for speech, even though the converse is not necessarily true. Similarities and differences between speech and music as inputs to a hearing aid are described. Many of these lead to the specification of a set of optimal electro-acoustic characteristics. Parameters such as the peak input-limiting level, compression issues-both compression ratio and knee-points-and number of channels all can deleteriously affect music perception through hearing aids. In other cases, it is not clear how to set other parameters such as noise reduction and feedback control mechanisms. Regardless of the existence of a "music program,'' unless the various electro-acoustic parameters are available in a hearing aid, music fidelity will almost always be less than optimal. There are many unanswered questions and hypotheses in this area. Future research by engineers, researchers, clinicians, and musicians will aid in the clarification of these questions and their ultimate solutions.

JND measurements of the speech formants parameters and its implication in the LPC pole quantization

NASA Astrophysics Data System (ADS)

Orgad, Yaakov

1988-08-01

The inherent sensitivity of auditory perception is explicitly used with the objective of designing an efficient speech encoder. Speech can be modelled by a filter representing the vocal tract shape that is driven by an excitation signal representing glottal air flow. This work concentrates on the filter encoding problem, assuming that excitation signal encoding is optimal. Linear predictive coding (LPC) techniques were used to model a short speech segment by an all-pole filter; each pole was directly related to the speech formants. Measurements were made of the auditory just noticeable difference (JND) corresponding to the natural speech formants, with the LPC filter poles as the best candidates to represent the speech spectral envelope. The JND is the maximum precision required in speech quantization; it was defined on the basis of the shift of one pole parameter of a single frame of a speech segment, necessary to induce subjective perception of the distortion, with .75 probability. The average JND in LPC filter poles in natural speech was found to increase with increasing pole bandwidth and, to a lesser extent, frequency. The JND measurements showed a large spread of the residuals around the average values, indicating that inter-formant coupling and, perhaps, other, not yet fully understood, factors were not taken into account at this stage of the research. A future treatment should consider these factors. The average JNDs obtained in this work were used to design pole quantization tables for speech coding and provided a better bit-rate than the standard quantizer of reflection coefficient; a 30-bits-per-frame pole quantizer yielded a speech quality similar to that obtained with a standard 41-bits-per-frame reflection coefficient quantizer. Owing to the complexity of the numerical root extraction system, the practical implementation of the pole quantization approach remains to be proved.

A universal multilingual weightless neural network tagger via quantitative linguistics.

PubMed

Carneiro, Hugo C C; Pedreira, Carlos E; França, Felipe M G; Lima, Priscila M V

2017-07-01

In the last decade, given the availability of corpora in several distinct languages, research on multilingual part-of-speech tagging started to grow. Amongst the novelties there is mWANN-Tagger (multilingual weightless artificial neural network tagger), a weightless neural part-of-speech tagger capable of being used for mostly-suffix-oriented languages. The tagger was subjected to corpora in eight languages of quite distinct natures and had a remarkable accuracy with very low sample deviation in every one of them, indicating the robustness of weightless neural systems for part-of-speech tagging tasks. However, mWANN-Tagger needed to be tuned for every new corpus, since each one required a different parameter configuration. For mWANN-Tagger to be truly multilingual, it should be usable for any new language with no need of parameter tuning. This article proposes a study that aims to find a relation between the lexical diversity of a language and the parameter configuration that would produce the best performing mWANN-Tagger instance. Preliminary analyses suggested that a single parameter configuration may be applied to the eight aforementioned languages. The mWANN-Tagger instance produced by this configuration was as accurate as the language-dependent ones obtained through tuning. Afterwards, the weightless neural tagger was further subjected to new corpora in languages that range from very isolating to polysynthetic ones. The best performing instances of mWANN-Tagger are again the ones produced by the universal parameter configuration. Hence, mWANN-Tagger can be applied to new corpora with no need of parameter tuning, making it a universal multilingual part-of-speech tagger. Further experiments with Universal Dependencies treebanks reveal that mWANN-Tagger may be extended and that it has potential to outperform most state-of-the-art part-of-speech taggers if better word representations are provided. Copyright © 2017 Elsevier Ltd. All rights reserved.

Computerized design of speech prostheses.

PubMed

Leonard, R J

1991-08-01

The use of computerized techniques to assist in the design of palatal and/or glossal prostheses is described. Patients with oropharyngeal resection and associated speech impairment are candidates for such prostheses. Procedures discussed allow for the design of some features of the prosthesis, such as shape, location, and tests of its effect on certain speech parameters, prior to actual fabrication. Advantages and current limitations of the techniques are also discussed.

Speech fluency profile on different tasks for individuals with Parkinson's disease.

PubMed

Juste, Fabiola Staróbole; Andrade, Claudia Regina Furquim de

2017-07-20

To characterize the speech fluency profile of patients with Parkinson's disease. Study participants were 40 individuals of both genders aged 40 to 80 years divided into 2 groups: Research Group - RG (20 individuals with diagnosis of Parkinson's disease) and Control Group - CG (20 individuals with no communication or neurological disorders). For all of the participants, three speech samples involving different tasks were collected: monologue, individual reading, and automatic speech. The RG presented a significant larger number of speech disruptions, both stuttering-like and typical dysfluencies, and higher percentage of speech discontinuity in the monologue and individual reading tasks compared with the CG. Both groups presented reduced number of speech disruptions (stuttering-like and typical dysfluencies) in the automatic speech task; the groups presented similar performance in this task. Regarding speech rate, individuals in the RG presented lower number of words and syllables per minute compared with those in the CG in all speech tasks. Participants of the RG presented altered parameters of speech fluency compared with those of the CG; however, this change in fluency cannot be considered a stuttering disorder.

Speech perception: Some new directions in research and theory

PubMed Central

Pisoni, David B.

2012-01-01

The perception of speech is one of the most fascinating attributes of human behavior; both the auditory periphery and higher centers help define the parameters of sound perception. In this paper some of the fundamental perceptual problems facing speech sciences are described. The paper focuses on several of the new directions speech perception research is taking to solve these problems. Recent developments suggest that major breakthroughs in research and theory will soon be possible. The current study of segmentation, invariance, and normalization are described. The paper summarizes some of the new techniques used to understand auditory perception of speech signals and their linguistic significance to the human listener. PMID:4031245

Scalable Video Transmission Over Multi-Rate Multiple Access Channels

DTIC Science & Technology

2007-06-01

Rate - compatible punctured convolutional codes (RCPC codes ) and their ap- plications,” IEEE...source encoded using the MPEG-4 video codec. The source encoded bitstream is then channel encoded with Rate Compatible Punctured Convolutional (RCPC...Clark, and J. M. Geist, “ Punctured convolutional codes or rate (n-1)/n and simplified maximum likelihood decoding,” IEEE Transactions on

AT2 DS II - Accelerator System Design (Part II) - CCC Video Conference

ScienceCinema

None

2017-12-09

Discussion Session - Accelerator System Design (Part II) Tutors: C. Darve, J. Weisend II, Ph. Lebrun, A. Dabrowski, U. Raich Video Conference with the CERN Control Center. Experts in the field of Accelerator science will be available to answer the students questions. This session will link the CCC and SA (using Codec VC).

Systematic studies of modified vocalization: effects of speech rate and instatement style during metronome stimulation.

PubMed

Davidow, Jason H; Bothe, Anne K; Richardson, Jessica D; Andreatta, Richard D

2010-12-01

This study introduces a series of systematic investigations intended to clarify the parameters of the fluency-inducing conditions (FICs) in stuttering. Participants included 11 adults, aged 20-63 years, with typical speech-production skills. A repeated measures design was used to examine the relationships between several speech production variables (vowel duration, voice onset time, fundamental frequency, intraoral pressure, pressure rise time, transglottal airflow, and phonated intervals) and speech rate and instatement style during metronome-entrained rhythmic speech. Measures of duration (vowel duration, voice onset time, and pressure rise time) differed across different metronome conditions. When speech rates were matched between the control condition and metronome condition, voice onset time was the only variable that changed. Results confirm that speech rate and instatement style can influence speech production variables during the production of fluency-inducing conditions. Future studies of normally fluent speech and of stuttered speech must control both features and should further explore the importance of voice onset time, which may be influenced by rate during metronome stimulation in a way that the other variables are not.

Hearing Aids and Music

PubMed Central

Chasin, Marshall; Russo, Frank A.

2004-01-01

Historically, the primary concern for hearing aid design and fitting is optimization for speech inputs. However, increasingly other types of inputs are being investigated and this is certainly the case for music. Whether the hearing aid wearer is a musician or merely someone who likes to listen to music, the electronic and electro-acoustic parameters described can be optimized for music as well as for speech. That is, a hearing aid optimally set for music can be optimally set for speech, even though the converse is not necessarily true. Similarities and differences between speech and music as inputs to a hearing aid are described. Many of these lead to the specification of a set of optimal electro-acoustic characteristics. Parameters such as the peak input-limiting level, compression issues—both compression ratio and knee-points—and number of channels all can deleteriously affect music perception through hearing aids. In other cases, it is not clear how to set other parameters such as noise reduction and feedback control mechanisms. Regardless of the existence of a “music program,” unless the various electro-acoustic parameters are available in a hearing aid, music fidelity will almost always be less than optimal. There are many unanswered questions and hypotheses in this area. Future research by engineers, researchers, clinicians, and musicians will aid in the clarification of these questions and their ultimate solutions. PMID:15497032

Irregular vocal fold dynamics incited by asymmetric fluid loading in a model of recurrent laryngeal nerve paralysis

NASA Astrophysics Data System (ADS)

Sommer, David; Erath, Byron D.; Zanartu, Matias; Peterson, Sean D.

2011-11-01

Voiced speech is produced by dynamic fluid-structure interactions in the larynx. Traditionally, reduced order models of speech have relied upon simplified inviscid flow solvers to prescribe the fluid loadings that drive vocal fold motion, neglecting viscous flow effects that occur naturally in voiced speech. Viscous phenomena, such as skewing of the intraglottal jet, have the most pronounced effect on voiced speech in cases of vocal fold paralysis where one vocal fold loses some, or all, muscular control. The impact of asymmetric intraglottal flow in pathological speech is captured in a reduced order two-mass model of speech by coupling a boundary-layer estimation of the asymmetric pressures with asymmetric tissue parameters that are representative of recurrent laryngeal nerve paralysis. Nonlinear analysis identifies the emergence of irregular and chaotic vocal fold dynamics at values representative of pathological speech conditions.

Differences in early speech patterns between Parkinson variant of multiple system atrophy and Parkinson's disease.

PubMed

Huh, Young Eun; Park, Jongkyu; Suh, Mee Kyung; Lee, Sang Eun; Kim, Jumin; Jeong, Yuri; Kim, Hee-Tae; Cho, Jin Whan

2015-08-01

In Parkinson variant of multiple system atrophy (MSA-P), patterns of early speech impairment and their distinguishing features from Parkinson's disease (PD) require further exploration. Here, we compared speech data among patients with early-stage MSA-P, PD, and healthy subjects using quantitative acoustic and perceptual analyses. Variables were analyzed for men and women in view of gender-specific features of speech. Acoustic analysis revealed that male patients with MSA-P exhibited more profound speech abnormalities than those with PD, regarding increased voice pitch, prolonged pause time, and reduced speech rate. This might be due to widespread pathology of MSA-P in nigrostriatal or extra-striatal structures related to speech production. Although several perceptual measures were mildly impaired in MSA-P and PD patients, none of these parameters showed a significant difference between patient groups. Detailed speech analysis using acoustic measures may help distinguish between MSA-P and PD early in the disease process. Copyright © 2015 Elsevier Inc. All rights reserved.

Affective state and voice: cross-cultural assessment of speaking behavior and voice sound characteristics--a normative multicenter study of 577 + 36 healthy subjects.

PubMed

Braun, Silke; Botella, Cristina; Bridler, René; Chmetz, Florian; Delfino, Juan Pablo; Herzig, Daniela; Kluckner, Viktoria J; Mohr, Christine; Moragrega, Ines; Schrag, Yann; Seifritz, Erich; Soler, Carla; Stassen, Hans H

2014-01-01

Human speech is greatly influenced by the speakers' affective state, such as sadness, happiness, grief, guilt, fear, anger, aggression, faintheartedness, shame, sexual arousal, love, amongst others. Attentive listeners discover a lot about the affective state of their dialog partners with no great effort, and without having to talk about it explicitly during a conversation or on the phone. On the other hand, speech dysfunctions, such as slow, delayed or monotonous speech, are prominent features of affective disorders. This project was comprised of four studies with healthy volunteers from Bristol (English: n = 117), Lausanne (French: n = 128), Zurich (German: n = 208), and Valencia (Spanish: n = 124). All samples were stratified according to gender, age, and education. The specific study design with different types of spoken text along with repeated assessments at 14-day intervals allowed us to estimate the 'natural' variation of speech parameters over time, and to analyze the sensitivity of speech parameters with respect to form and content of spoken text. Additionally, our project included a longitudinal self-assessment study with university students from Zurich (n = 18) and unemployed adults from Valencia (n = 18) in order to test the feasibility of the speech analysis method in home environments. The normative data showed that speaking behavior and voice sound characteristics can be quantified in a reproducible and language-independent way. The high resolution of the method was verified by a computerized assignment of speech parameter patterns to languages at a success rate of 90%, while the correct assignment to texts was 70%. In the longitudinal self-assessment study we calculated individual 'baselines' for each test person along with deviations thereof. The significance of such deviations was assessed through the normative reference data. Our data provided gender-, age-, and language-specific thresholds that allow one to reliably distinguish between 'natural fluctuations' and 'significant changes'. The longitudinal self-assessment study with repeated assessments at 1-day intervals over 14 days demonstrated the feasibility and efficiency of the speech analysis method in home environments, thus clearing the way to a broader range of applications in psychiatry. © 2014 S. Karger AG, Basel.

Nonlinear frequency compression: effects on sound quality ratings of speech and music.

PubMed

Parsa, Vijay; Scollie, Susan; Glista, Danielle; Seelisch, Andreas

2013-03-01

Frequency lowering technologies offer an alternative amplification solution for severe to profound high frequency hearing losses. While frequency lowering technologies may improve audibility of high frequency sounds, the very nature of this processing can affect the perceived sound quality. This article reports the results from two studies that investigated the impact of a nonlinear frequency compression (NFC) algorithm on perceived sound quality. In the first study, the cutoff frequency and compression ratio parameters of the NFC algorithm were varied, and their effect on the speech quality was measured subjectively with 12 normal hearing adults, 12 normal hearing children, 13 hearing impaired adults, and 9 hearing impaired children. In the second study, 12 normal hearing and 8 hearing impaired adult listeners rated the quality of speech in quiet, speech in noise, and music after processing with a different set of NFC parameters. Results showed that the cutoff frequency parameter had more impact on sound quality ratings than the compression ratio, and that the hearing impaired adults were more tolerant to increased frequency compression than normal hearing adults. No statistically significant differences were found in the sound quality ratings of speech-in-noise and music stimuli processed through various NFC settings by hearing impaired listeners. These findings suggest that there may be an acceptable range of NFC settings for hearing impaired individuals where sound quality is not adversely affected. These results may assist an Audiologist in clinical NFC hearing aid fittings for achieving a balance between high frequency audibility and sound quality.

Study of wavelet packet energy entropy for emotion classification in speech and glottal signals

NASA Astrophysics Data System (ADS)

He, Ling; Lech, Margaret; Zhang, Jing; Ren, Xiaomei; Deng, Lihua

2013-07-01

The automatic speech emotion recognition has important applications in human-machine communication. Majority of current research in this area is focused on finding optimal feature parameters. In recent studies, several glottal features were examined as potential cues for emotion differentiation. In this study, a new type of feature parameter is proposed, which calculates energy entropy on values within selected Wavelet Packet frequency bands. The modeling and classification tasks are conducted using the classical GMM algorithm. The experiments use two data sets: the Speech Under Simulated Emotion (SUSE) data set annotated with three different emotions (angry, neutral and soft) and Berlin Emotional Speech (BES) database annotated with seven different emotions (angry, bored, disgust, fear, happy, sad and neutral). The average classification accuracy achieved for the SUSE data (74%-76%) is significantly higher than the accuracy achieved for the BES data (51%-54%). In both cases, the accuracy was significantly higher than the respective random guessing levels (33% for SUSE and 14.3% for BES).

Emotion to emotion speech conversion in phoneme level

NASA Astrophysics Data System (ADS)

Bulut, Murtaza; Yildirim, Serdar; Busso, Carlos; Lee, Chul Min; Kazemzadeh, Ebrahim; Lee, Sungbok; Narayanan, Shrikanth

2004-10-01

Having an ability to synthesize emotional speech can make human-machine interaction more natural in spoken dialogue management. This study investigates the effectiveness of prosodic and spectral modification in phoneme level on emotion-to-emotion speech conversion. The prosody modification is performed with the TD-PSOLA algorithm (Moulines and Charpentier, 1990). We also transform the spectral envelopes of source phonemes to match those of target phonemes using LPC-based spectral transformation approach (Kain, 2001). Prosodic speech parameters (F0, duration, and energy) for target phonemes are estimated from the statistics obtained from the analysis of an emotional speech database of happy, angry, sad, and neutral utterances collected from actors. Listening experiments conducted with native American English speakers indicate that the modification of prosody only or spectrum only is not sufficient to elicit targeted emotions. The simultaneous modification of both prosody and spectrum results in higher acceptance rates of target emotions, suggesting that not only modeling speech prosody but also modeling spectral patterns that reflect underlying speech articulations are equally important to synthesize emotional speech with good quality. We are investigating suprasegmental level modifications for further improvement in speech quality and expressiveness.

Standardization of Freeze Frame TV Codecs

DTIC Science & Technology

1990-06-01

Kodak SV9600 Still Video Transceiver Colorado Video, Inc.286 Digital Transceiver Image Data Corp. CP-200 Photophone Interand Corp. DISCON Imagephone...error recovery Proprietary Proprby retransmission errorIMAGE BUILD-UP Sequential Sequential PHOTOPHONE Video Teleconferenc- DISCON Imaqephone GENERIC...and information transfer is effected among terminals. An indication of the function and power of these commands can be obtained by reviewing Table

An 8-PSK TDMA uplink modulation and coding system

NASA Technical Reports Server (NTRS)

Ames, S. A.

1992-01-01

The combination of 8-phase shift keying (8PSK) modulation and greater than 2 bits/sec/Hz drove the design of the Nyquist filter to one specified to have a rolloff factor of 0.2. This filter when built and tested was found to produce too much intersymbol interference and was abandoned for a design with a rolloff factor of 0.4. The preamble is limited to 100 bit periods of the uncoded bit period of 5 ns for a maximum preamble length of 500 ns or 40 8PSK symbol times at 12.5 ns per symbol. For 8PSK modulation, the required maximum degradation of 1 dB in -20 dB cochannel interference (CCI) drove the requirement for forward error correction coding. In this contract, the funding was not sufficient to develop the proposed codec so the codec was limited to a paper design during the preliminary design phase. The mechanization of the demodulator is digital, starting from the output of the analog to digital converters which quantize the outputs of the quadrature phase detectors. This approach is amenable to an application specific integrated circuit (ASIC) replacement in the next phase of development.

Improved inter-layer prediction for light field content coding with display scalability

NASA Astrophysics Data System (ADS)

Conti, Caroline; Ducla Soares, Luís.; Nunes, Paulo

2016-09-01

Light field imaging based on microlens arrays - also known as plenoptic, holoscopic and integral imaging - has recently risen up as feasible and prospective technology due to its ability to support functionalities not straightforwardly available in conventional imaging systems, such as: post-production refocusing and depth of field changing. However, to gradually reach the consumer market and to provide interoperability with current 2D and 3D representations, a display scalable coding solution is essential. In this context, this paper proposes an improved display scalable light field codec comprising a three-layer hierarchical coding architecture (previously proposed by the authors) that provides interoperability with 2D (Base Layer) and 3D stereo and multiview (First Layer) representations, while the Second Layer supports the complete light field content. For further improving the compression performance, novel exemplar-based inter-layer coding tools are proposed here for the Second Layer, namely: (i) an inter-layer reference picture construction relying on an exemplar-based optimization algorithm for texture synthesis, and (ii) a direct prediction mode based on exemplar texture samples from lower layers. Experimental results show that the proposed solution performs better than the tested benchmark solutions, including the authors' previous scalable codec.

Broadband set-top box using MAP-CA processor

NASA Astrophysics Data System (ADS)

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

2001-12-01

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

On-chip frame memory reduction using a high-compression-ratio codec in the overdrives of liquid-crystal displays

NASA Astrophysics Data System (ADS)

Wang, Jun; Min, Kyeong-Yuk; Chong, Jong-Wha

2010-11-01

Overdrive is commonly used to reduce the liquid-crystal response time and motion blur in liquid-crystal displays (LCDs). However, overdrive requires a large frame memory in order to store the previous frame for reference. In this paper, a high-compression-ratio codec is presented to compress the image data stored in the on-chip frame memory so that only 1 Mbit of on-chip memory is required in the LCD overdrives of mobile devices. The proposed algorithm further compresses the color bitmaps and representative values (RVs) resulting from the block truncation coding (BTC). The color bitmaps are represented by a luminance bitmap, which is further reduced and reconstructed using median filter interpolation in the decoder, while the RVs are compressed using adaptive quantization coding (AQC). Interpolation and AQC can provide three-level compression, which leads to 16 combinations. Using a rate-distortion analysis, we select the three optimal schemes to compress the image data for video graphics array (VGA), wide-VGA LCD, and standard-definitionTV applications. Our simulation results demonstrate that the proposed schemes outperform interpolation BTC both in PSNR (by 1.479 to 2.205 dB) and in subjective visual quality.

Effect of bilateral stimulation of the subthalamic nucleus on different speech subsystems in patients with Parkinson's disease.

PubMed

Pützer, Manfred; Barry, William J; Moringlane, Jean Richard

2008-12-01

The effect of deep brain stimulation on the two speech-production subsystems, articulation and phonation, of nine Parkinsonian patients is examined. Production parameters (stop closure voicing; stop closure, VOT, vowel) in fast syllable-repetitions were defined and measured and quantitative, objective metrics of vocal fold function were obtained during vowel production. Speech material was recorded for patients (with and without stimulation) and for a reference group of healthy control speakers. With stimulation, precision of the glottal and supraglottal articulation as well as the phonatory function is reduced for some individuals, whereas for other individuals an improvement is observed. Importantly, the improvement or deterioration is determined not only on the basis of the direction of parameter change but also on the individuals' position relative to the healthy control data. This study also notes differences within an individual in the effects of stimulation on the two speech subsystems. These findings qualify the value of global statements about the effect of neurostimulatory operations on Parkinsonian patients. They also underline the importance of careful consideration of individual differences in the effect of deep brain stimulation on different speech subsystems.

Visual and Auditory Components in the Perception of Asynchronous Audiovisual Speech

PubMed Central

Alcalá-Quintana, Rocío

2015-01-01

Research on asynchronous audiovisual speech perception manipulates experimental conditions to observe their effects on synchrony judgments. Probabilistic models establish a link between the sensory and decisional processes underlying such judgments and the observed data, via interpretable parameters that allow testing hypotheses and making inferences about how experimental manipulations affect such processes. Two models of this type have recently been proposed, one based on independent channels and the other using a Bayesian approach. Both models are fitted here to a common data set, with a subsequent analysis of the interpretation they provide about how experimental manipulations affected the processes underlying perceived synchrony. The data consist of synchrony judgments as a function of audiovisual offset in a speech stimulus, under four within-subjects manipulations of the quality of the visual component. The Bayesian model could not accommodate asymmetric data, was rejected by goodness-of-fit statistics for 8/16 observers, and was found to be nonidentifiable, which renders uninterpretable parameter estimates. The independent-channels model captured asymmetric data, was rejected for only 1/16 observers, and identified how sensory and decisional processes mediating asynchronous audiovisual speech perception are affected by manipulations that only alter the quality of the visual component of the speech signal. PMID:27551361

Emotionally conditioning the target-speech voice enhances recognition of the target speech under "cocktail-party" listening conditions.

PubMed

Lu, Lingxi; Bao, Xiaohan; Chen, Jing; Qu, Tianshu; Wu, Xihong; Li, Liang

2018-05-01

Under a noisy "cocktail-party" listening condition with multiple people talking, listeners can use various perceptual/cognitive unmasking cues to improve recognition of the target speech against informational speech-on-speech masking. One potential unmasking cue is the emotion expressed in a speech voice, by means of certain acoustical features. However, it was unclear whether emotionally conditioning a target-speech voice that has none of the typical acoustical features of emotions (i.e., an emotionally neutral voice) can be used by listeners for enhancing target-speech recognition under speech-on-speech masking conditions. In this study we examined the recognition of target speech against a two-talker speech masker both before and after the emotionally neutral target voice was paired with a loud female screaming sound that has a marked negative emotional valence. The results showed that recognition of the target speech (especially the first keyword in a target sentence) was significantly improved by emotionally conditioning the target speaker's voice. Moreover, the emotional unmasking effect was independent of the unmasking effect of the perceived spatial separation between the target speech and the masker. Also, (skin conductance) electrodermal responses became stronger after emotional learning when the target speech and masker were perceptually co-located, suggesting an increase of listening efforts when the target speech was informationally masked. These results indicate that emotionally conditioning the target speaker's voice does not change the acoustical parameters of the target-speech stimuli, but the emotionally conditioned vocal features can be used as cues for unmasking target speech.

Measuring the rate of change of voice fundamental frequency in fluent speech during mental depression.

PubMed

Nilsonne, A; Sundberg, J; Ternström, S; Askenfelt, A

1988-02-01

A method of measuring the rate of change of fundamental frequency has been developed in an effort to find acoustic voice parameters that could be useful in psychiatric research. A minicomputer program was used to extract seven parameters from the fundamental frequency contour of tape-recorded speech samples: (1) the average rate of change of the fundamental frequency and (2) its standard deviation, (3) the absolute rate of fundamental frequency change, (4) the total reading time, (5) the percent pause time of the total reading time, (6) the mean, and (7) the standard deviation of the fundamental frequency distribution. The method is demonstrated on (a) a material consisting of synthetic speech and (b) voice recordings of depressed patients who were examined during depression and after improvement.

A Computational Model Quantifies the Effect of Anatomical Variability on Velopharyngeal Function

PubMed Central

Inouye, Joshua M.; Perry, Jamie L.; Lin, Kant Y.

2015-01-01

Purpose This study predicted the effects of velopharyngeal (VP) anatomical parameters on VP function to provide a greater understanding of speech mechanics and aid in the treatment of speech disorders. Method We created a computational model of the VP mechanism using dimensions obtained from magnetic resonance imaging measurements of 10 healthy adults. The model components included the levator veli palatini (LVP), the velum, and the posterior pharyngeal wall, and the simulations were based on material parameters from the literature. The outcome metrics were the VP closure force and LVP muscle activation required to achieve VP closure. Results Our average model compared favorably with experimental data from the literature. Simulations of 1,000 random anatomies reflected the large variability in closure forces observed experimentally. VP distance had the greatest effect on both outcome metrics when considering the observed anatomic variability. Other anatomical parameters were ranked by their predicted influences on the outcome metrics. Conclusions Our results support the implication that interventions for VP dysfunction that decrease anterior to posterior VP portal distance, increase velar length, and/or increase LVP cross-sectional area may be very effective. Future modeling studies will help to further our understanding of speech mechanics and optimize treatment of speech disorders. PMID:26049120

[Simulation of speech perception with cochlear implants : Influence of frequency and level of fundamental frequency components with electronic acoustic stimulation].

PubMed

Rader, T; Fastl, H; Baumann, U

2017-03-01

After implantation of cochlear implants with hearing preservation for combined electronic acoustic stimulation (EAS), the residual acoustic hearing ability relays fundamental speech frequency information in the low frequency range. With the help of acoustic simulation of EAS hearing perception the impact of frequency and level fine structure of speech signals can be systematically examined. The aim of this study was to measure the speech reception threshold (SRT) under various noise conditions with acoustic EAS simulation by variation of the frequency and level information of the fundamental frequency f0 of speech. The study was carried out to determine to what extent the SRT is impaired by modification of the f0 fine structure. Using partial tone time pattern analysis an acoustic EAS simulation of the speech material from the Oldenburg sentence test (OLSA) was generated. In addition, determination of the f0 curve of the speech material was conducted. Subsequently, either the parameter frequency or level of f0 was fixed in order to remove one of the two fine contour information of the speech signal. The processed OLSA sentences were used to determine the SRT in background noise under various test conditions. The conditions "f0 fixed frequency" and "f0 fixed level" were tested under two different situations, under "amplitude modulated background noise" and "continuous background noise" conditions. A total of 24 subjects with normal hearing participated in the study. The SRT in background noise for the condition "f0 fixed frequency" was more favorable in continuous noise with 2.7 dB and in modulated noise with 0.8 dB compared to the condition "f0 fixed level" with 3.7 dB and 2.9 dB, respectively. In the simulation of speech perception with cochlear implants and acoustic components, the level information of the fundamental frequency had a stronger impact on speech intelligibility than the frequency information. The method of simulation of transmission of cochlear implants allows investigation of how various parameters influence speech intelligibility in subjects with normal hearing.

Voice Preprocessor for Digital Voice Applications

DTIC Science & Technology

1989-09-11

helit tralnsforniers are marketed for use kll ith multitonle MI( LMS and are acceptable for w ice appl ica- tiotns. 3. Automatic Gain (iontro: A...variations of speech spectral tilt to improve the quaiit\\ of the extracted speech parameters. Nlore imnportantly, the onlN analoii circuit "e use is a

Five-year speech and language outcomes in children with cleft lip-palate.

PubMed

Prathanee, Benjamas; Pumnum, Tawitree; Seepuaham, Cholada; Jaiyong, Pechcharat

2016-10-01

To investigate 5-year speech and language outcomes in children with cleft lip/palate (CLP). Thirty-eight children aged 4-7 years and 8 months were recruited for this study. Speech abilities including articulation, resonance, voice, and intelligibility were assessed based on Thai Universal Parameters of Speech Outcomes. Language ability was assessed by the Language Screening Test. The findings revealed that children with clefts had speech and language delay, abnormal understandability, resonance abnormality, and voice disturbance; articulation defects that were 8.33 (1.75, 22.47), 50.00 (32.92, 67.08), 36.11 (20.82, 53.78), 30.56 (16.35, 48.11), and 94.44 (81.34, 99.32). Articulation errors were the most common speech and language defects in children with clefts, followed by abnormal understandability, resonance abnormality, and voice disturbance. These results should be of critical concern. Protocol reviewing and early intervention programs are needed for improved speech outcomes. Copyright © 2016 European Association for Cranio-Maxillo-Facial Surgery. Published by Elsevier Ltd. All rights reserved.

Effects of human fatigue on speech signals

NASA Astrophysics Data System (ADS)

Stamoulis, Catherine

2004-05-01

Cognitive performance may be significantly affected by fatigue. In the case of critical personnel, such as pilots, monitoring human fatigue is essential to ensure safety and success of a given operation. One of the modalities that may be used for this purpose is speech, which is sensitive to respiratory changes and increased muscle tension of vocal cords, induced by fatigue. Age, gender, vocal tract length, physical and emotional state may significantly alter speech intensity, duration, rhythm, and spectral characteristics. In addition to changes in speech rhythm, fatigue may also affect the quality of speech, such as articulation. In a noisy environment, detecting fatigue-related changes in speech signals, particularly subtle changes at the onset of fatigue, may be difficult. Therefore, in a performance-monitoring system, speech parameters which are significantly affected by fatigue need to be identified and extracted from input signals. For this purpose, a series of experiments was performed under slowly varying cognitive load conditions and at different times of the day. The results of the data analysis are presented here.

Noise suppression methods for robust speech processing

NASA Astrophysics Data System (ADS)

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

1980-05-01

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

[Detection of Weak Speech Signals from Strong Noise Background Based on Adaptive Stochastic Resonance].

PubMed

Lu, Huanhuan; Wang, Fuzhong; Zhang, Huichun

2016-04-01

Traditional speech detection methods regard the noise as a jamming signal to filter,but under the strong noise background,these methods lost part of the original speech signal while eliminating noise.Stochastic resonance can use noise energy to amplify the weak signal and suppress the noise.According to stochastic resonance theory,a new method based on adaptive stochastic resonance to extract weak speech signals is proposed.This method,combined with twice sampling,realizes the detection of weak speech signals from strong noise.The parameters of the systema,b are adjusted adaptively by evaluating the signal-to-noise ratio of the output signal,and then the weak speech signal is optimally detected.Experimental simulation analysis showed that under the background of strong noise,the output signal-to-noise ratio increased from the initial value-7dB to about 0.86 dB,with the gain of signalto-noise ratio is 7.86 dB.This method obviously raises the signal-to-noise ratio of the output speech signals,which gives a new idea to detect the weak speech signals in strong noise environment.

An evaluation of the effectiveness of PROMPT therapy in improving speech production accuracy in six children with cerebral palsy.

PubMed

Ward, Roslyn; Leitão, Suze; Strauss, Geoff

2014-08-01

This study evaluates perceptual changes in speech production accuracy in six children (3-11 years) with moderate-to-severe speech impairment associated with cerebral palsy before, during, and after participation in a motor-speech intervention program (Prompts for Restructuring Oral Muscular Phonetic Targets). An A1BCA2 single subject research design was implemented. Subsequent to the baseline phase (phase A1), phase B targeted each participant's first intervention priority on the PROMPT motor-speech hierarchy. Phase C then targeted one level higher. Weekly speech probes were administered, containing trained and untrained words at the two levels of intervention, plus an additional level that served as a control goal. The speech probes were analysed for motor-speech-movement-parameters and perceptual accuracy. Analysis of the speech probe data showed all participants recorded a statistically significant change. Between phases A1-B and B-C 6/6 and 4/6 participants, respectively, recorded a statistically significant increase in performance level on the motor speech movement patterns targeted during the training of that intervention. The preliminary data presented in this study make a contribution to providing evidence that supports the use of a treatment approach aligned with dynamic systems theory to improve the motor-speech movement patterns and speech production accuracy in children with cerebral palsy.

The Role of Efficient XML Interchange (EXI) in Navy Wide-Area Network (WAN) Optimization

DTIC Science & Technology

2015-03-01

compress, and re-encrypt data to continue providing optimization through compression; however, that capability requires careful consideration of...optimization 23 of encrypted data requires a careful analysis and comparison of performance improvements and IA vulnerabilities. It is important...Contained EXI capitalizes on multiple techniques to improve compression, and they vary depending on a set of EXI options passed to the codec

Architecture design of motion estimation for ITU-T H.263

NASA Astrophysics Data System (ADS)

Ku, Chung-Wei; Lin, Gong-Sheng; Chen, Liang-Gee; Lee, Yung-Ping

1997-01-01

Digitalized video and audio system has become the trend of the progress in multimedia, because it provides great performance in quality and feasibility of processing. However, as the huge amount of information is needed while the bandwidth is limitted, data compression plays an important role in the system. Say, for a 176 x 144 monochromic sequence with 10 frames/sec frame rate, the bandwidth is about 2Mbps. This wastes much channel resource and limits the applications. MPEG (moving picttre ezpert groip) standardizes the video codec scheme, and it performs high compression ratio while providing good quality. MPEG-i is used for the frame size about 352 x 240 and 30 frames per second, and MPEG-2 provides scalibility and can be applied on scenes with higher definition, say HDTV (high definition television). On the other hand, some applications concerns the very low bit-rate, such as videophone and video-conferencing. Because the channel bandwidth is much limitted in telephone network, a very high compression ratio must be required. ITU-T announced the H.263 video coding standards to meet the above requirements.8 According to the simulation results of TMN-5,22 it outperforms 11.263 with little overhead of complexity. Since wireless communication is the trend in the near future, low power design of the video codec is an important issue for portable visual telephone. Motion estimation is the most computation consuming parts in the whole video codec. About 60% of the computation is spent on this parts for the encoder. Several architectures were proposed for efficient processing of block matching algorithms. In this paper, in order to meet the requirements of 11.263 and the expectation of low power consumption, a modified sandwich architecture in21 is proposed. Based on the parallel processing philosophy, low power is expected and the generation of either one motion vector or four motion vectors with half-pixel accuracy is achieved concurrently. In addition, we will present our solution how to solve the other addition modes in 11.263 with the proposed architecture.

Thermal welding vs. cold knife tonsillectomy: a comparison of voice and speech.

PubMed

Celebi, Saban; Yelken, Kursat; Celik, Oner; Taskin, Umit; Topak, Murat

2011-01-01

To compare acoustic, aerodynamic and perceptual voice and speech parameters in thermal welding system tonsillectomy and cold knife tonsillectomy patients in order to determine the impact of operation technique on voice and speech. Thirty tonsillectomy patients (22 children, 8 adults) participated in this study. The preferred technique was cold knife tonsillectomy in 15 patients and thermal welding system tonsillectomy in the remaining 15 patients. One week before and 1 month after surgery the following parameters were estimated: average of fundamental frequency, Jitter, Shimmer, harmonic to noise ratio, formant frequency analyses of sustained vowels. Perceptual speech analysis and aerodynamic measurements (maximum phonation time and s/z ratio) were also conducted. There was no significant difference in any of the parameters between cold knife tonsillectomy and thermal welding system tonsillectomy groups (p>0.05). When the groups were contrasted among themselves with regards to preoperative and postoperative rates, fundamental frequency was found to be significantly decreased after tonsillectomy in both of the groups (p<0.001). First formant for the vowel /a/ in the cold knife tonsillectomy group and for the vowel /i/ in the thermal welding system tonsillectomy group, second formant for the vowel /u/ in the thermal welding system tonsillectomy group and third formant for the vowel /u/ in the cold knife tonsillectomy group were found to be significantly decreased (p<0.05). The surgical technique, whether it is cold knife or thermal welding system, does not appear to affect voice and speech in tonsillectomy patients. Copyright © 2010 Elsevier Ireland Ltd. All rights reserved.

Speech rate in Parkinson's disease: A controlled study.

PubMed

Martínez-Sánchez, F; Meilán, J J G; Carro, J; Gómez Íñiguez, C; Millian-Morell, L; Pujante Valverde, I M; López-Alburquerque, T; López, D E

2016-09-01

Speech disturbances will affect most patients with Parkinson's disease (PD) over the course of the disease. The origin and severity of these symptoms are of clinical and diagnostic interest. To evaluate the clinical pattern of speech impairment in PD patients and identify significant differences in speech rate and articulation compared to control subjects. Speech rate and articulation in a reading task were measured using an automatic analytical method. A total of 39 PD patients in the 'on' state and 45 age-and sex-matched asymptomatic controls participated in the study. None of the patients experienced dyskinesias or motor fluctuations during the test. The patients with PD displayed a significant reduction in speech and articulation rates; there were no significant correlations between the studied speech parameters and patient characteristics such as L-dopa dose, duration of the disorder, age, and UPDRS III scores and Hoehn & Yahr scales. Patients with PD show a characteristic pattern of declining speech rate. These results suggest that in PD, disfluencies are the result of the movement disorder affecting the physiology of speech production systems. Copyright © 2014 Sociedad Española de Neurología. Publicado por Elsevier España, S.L.U. All rights reserved.

Improved Speech Coding Based on Open-Loop Parameter Estimation

NASA Technical Reports Server (NTRS)

Juang, Jer-Nan; Chen, Ya-Chin; Longman, Richard W.

2000-01-01

A nonlinear optimization algorithm for linear predictive speech coding was developed early that not only optimizes the linear model coefficients for the open loop predictor, but does the optimization including the effects of quantization of the transmitted residual. It also simultaneously optimizes the quantization levels used for each speech segment. In this paper, we present an improved method for initialization of this nonlinear algorithm, and demonstrate substantial improvements in performance. In addition, the new procedure produces monotonically improving speech quality with increasing numbers of bits used in the transmitted error residual. Examples of speech encoding and decoding are given for 8 speech segments and signal to noise levels as high as 47 dB are produced. As in typical linear predictive coding, the optimization is done on the open loop speech analysis model. Here we demonstrate that minimizing the error of the closed loop speech reconstruction, instead of the simpler open loop optimization, is likely to produce negligible improvement in speech quality. The examples suggest that the algorithm here is close to giving the best performance obtainable from a linear model, for the chosen order with the chosen number of bits for the codebook.

Video transmission on ATM networks. Ph.D. Thesis

NASA Technical Reports Server (NTRS)

Chen, Yun-Chung

1993-01-01

The broadband integrated services digital network (B-ISDN) is expected to provide high-speed and flexible multimedia applications. Multimedia includes data, graphics, image, voice, and video. Asynchronous transfer mode (ATM) is the adopted transport techniques for B-ISDN and has the potential for providing a more efficient and integrated environment for multimedia. It is believed that most broadband applications will make heavy use of visual information. The prospect of wide spread use of image and video communication has led to interest in coding algorithms for reducing bandwidth requirements and improving image quality. The major results of a study on the bridging of network transmission performance and video coding are: Using two representative video sequences, several video source models are developed. The fitness of these models are validated through the use of statistical tests and network queuing performance. A dual leaky bucket algorithm is proposed as an effective network policing function. The concept of the dual leaky bucket algorithm can be applied to a prioritized coding approach to achieve transmission efficiency. A mapping of the performance/control parameters at the network level into equivalent parameters at the video coding level is developed. Based on that, a complete set of principles for the design of video codecs for network transmission is proposed.

Speech fluency profile in Williams-Beuren syndrome: a preliminary study.

PubMed

Rossi, Natalia Freitas; Souza, Deise Helena de; Moretti-Ferreira, Danilo; Giacheti, Célia Maria

2009-01-01

the speech fluency pattern attributed to individuals with Williams-Beuren syndrome (WBS) is supported by the effectiveness of the phonological loop. Some studies have reported the occurrence of speech disruptions caused by lexical and semantic deficits. However, the type and frequency of such speech disruptions has not been well elucidated. to determine the speech fluency profile of individuals with WBS and to compare the speech performance of these individuals to a control group matched by gender and mental age. Twelve subjects with Williams-Beuren syndrome, chronologically aged between 6.6 and 23.6 years and mental age ranging from 4.8 to 14.3 years, were evaluated. They were compared with another group consisting of 12 subjects with similar mental age and with no speech or learning difficulties. Speech fluency parameters were assessed according to the ABFW Language Test: type and frequency of speech disruptions and speech rate. The obtained results were compared between the groups. In comparison with individuals of similar mental age and typical speech and language development, the group with Williams-Beuren syndrome showed a greater percentage of speech discontinuity, and an increased frequency of common hesitations and word repetition. The speech fluency profile presented by individuals with WBS in this study suggests that the presence of disfluencies can be caused by deficits in the lexical, semantic, and syntactic processing of verbal information. The authors stress that further systematic investigations on the subject are warranted.

A Computational Model Quantifies the Effect of Anatomical Variability on Velopharyngeal Function

ERIC Educational Resources Information Center

Inouye, Joshua M.; Perry, Jamie L.; Lin, Kant Y.; Blemker, Silvia S.

2015-01-01

Purpose: This study predicted the effects of velopharyngeal (VP) anatomical parameters on VP function to provide a greater understanding of speech mechanics and aid in the treatment of speech disorders. Method: We created a computational model of the VP mechanism using dimensions obtained from magnetic resonance imaging measurements of 10 healthy…

Vector adaptive predictive coder for speech and audio

NASA Technical Reports Server (NTRS)

Chen, Juin-Hwey (Inventor); Gersho, Allen (Inventor)

1990-01-01

A real-time vector adaptive predictive coder which approximates each vector of K speech samples by using each of M fixed vectors in a first codebook to excite a time-varying synthesis filter and picking the vector that minimizes distortion. Predictive analysis for each frame determines parameters used for computing from vectors in the first codebook zero-state response vectors that are stored at the same address (index) in a second codebook. Encoding of input speech vectors s.sub.n is then carried out using the second codebook. When the vector that minimizes distortion is found, its index is transmitted to a decoder which has a codebook identical to the first codebook of the decoder. There the index is used to read out a vector that is used to synthesize an output speech vector s.sub.n. The parameters used in the encoder are quantized, for example by using a table, and the indices are transmitted to the decoder where they are decoded to specify transfer characteristics of filters used in producing the vector s.sub.n from the receiver codebook vector selected by the vector index transmitted.

Speech recovery and language plasticity can be facilitated by Sensori-Motor Fusion training in chronic non-fluent aphasia. A case report study.

PubMed

Haldin, Célise; Acher, Audrey; Kauffmann, Louise; Hueber, Thomas; Cousin, Emilie; Badin, Pierre; Perrier, Pascal; Fabre, Diandra; Perennou, Dominic; Detante, Olivier; Jaillard, Assia; Lœvenbruck, Hélène; Baciu, Monica

2017-11-17

The rehabilitation of speech disorders benefits from providing visual information which may improve speech motor plans in patients. We tested the proof of concept of a rehabilitation method (Sensori-Motor Fusion, SMF; Ultraspeech player) in one post-stroke patient presenting chronic non-fluent aphasia. SMF allows visualisation by the patient of target tongue and lips movements using high-speed ultrasound and video imaging. This can improve the patient's awareness of his/her own lingual and labial movements, which can, in turn, improve the representation of articulatory movements and increase the ability to coordinate and combine articulatory gestures. The auditory and oro-sensory feedback received by the patient as a result of his/her own pronunciation can be integrated with the target articulatory movements they watch. Thus, this method is founded on sensorimotor integration during speech. The SMF effect on this patient was assessed through qualitative comparison of language scores and quantitative analysis of acoustic parameters measured in a speech production task, before and after rehabilitation. We also investigated cerebral patterns of language reorganisation for rhyme detection and syllable repetition, to evaluate the influence of SMF on phonological-phonetic processes. Our results showed that SMF had a beneficial effect on this patient who qualitatively improved in naming, reading, word repetition and rhyme judgment tasks. Quantitative measurements of acoustic parameters indicate that the patient's production of vowels and syllables also improved. Compared with pre-SMF, the fMRI data in the post-SMF session revealed the activation of cerebral regions related to articulatory, auditory and somatosensory processes, which were expected to be recruited by SMF. We discuss neurocognitive and linguistic mechanisms which may explain speech improvement after SMF, as well as the advantages of using this speech rehabilitation method.

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

PubMed Central

Souza, Pamela E.

2002-01-01

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

Sparse gammatone signal model optimized for English speech does not match the human auditory filters.

PubMed

Strahl, Stefan; Mertins, Alfred

2008-07-18

Evidence that neurosensory systems use sparse signal representations as well as improved performance of signal processing algorithms using sparse signal models raised interest in sparse signal coding in the last years. For natural audio signals like speech and environmental sounds, gammatone atoms have been derived as expansion functions that generate a nearly optimal sparse signal model (Smith, E., Lewicki, M., 2006. Efficient auditory coding. Nature 439, 978-982). Furthermore, gammatone functions are established models for the human auditory filters. Thus far, a practical application of a sparse gammatone signal model has been prevented by the fact that deriving the sparsest representation is, in general, computationally intractable. In this paper, we applied an accelerated version of the matching pursuit algorithm for gammatone dictionaries allowing real-time and large data set applications. We show that a sparse signal model in general has advantages in audio coding and that a sparse gammatone signal model encodes speech more efficiently in terms of sparseness than a sparse modified discrete cosine transform (MDCT) signal model. We also show that the optimal gammatone parameters derived for English speech do not match the human auditory filters, suggesting for signal processing applications to derive the parameters individually for each applied signal class instead of using psychometrically derived parameters. For brain research, it means that care should be taken with directly transferring findings of optimality for technical to biological systems.

Soft context clustering for F0 modeling in HMM-based speech synthesis

NASA Astrophysics Data System (ADS)

Khorram, Soheil; Sameti, Hossein; King, Simon

2015-12-01

This paper proposes the use of a new binary decision tree, which we call a soft decision tree, to improve generalization performance compared to the conventional `hard' decision tree method that is used to cluster context-dependent model parameters in statistical parametric speech synthesis. We apply the method to improve the modeling of fundamental frequency, which is an important factor in synthesizing natural-sounding high-quality speech. Conventionally, hard decision tree-clustered hidden Markov models (HMMs) are used, in which each model parameter is assigned to a single leaf node. However, this `divide-and-conquer' approach leads to data sparsity, with the consequence that it suffers from poor generalization, meaning that it is unable to accurately predict parameters for models of unseen contexts: the hard decision tree is a weak function approximator. To alleviate this, we propose the soft decision tree, which is a binary decision tree with soft decisions at the internal nodes. In this soft clustering method, internal nodes select both their children with certain membership degrees; therefore, each node can be viewed as a fuzzy set with a context-dependent membership function. The soft decision tree improves model generalization and provides a superior function approximator because it is able to assign each context to several overlapped leaves. In order to use such a soft decision tree to predict the parameters of the HMM output probability distribution, we derive the smoothest (maximum entropy) distribution which captures all partial first-order moments and a global second-order moment of the training samples. Employing such a soft decision tree architecture with maximum entropy distributions, a novel speech synthesis system is trained using maximum likelihood (ML) parameter re-estimation and synthesis is achieved via maximum output probability parameter generation. In addition, a soft decision tree construction algorithm optimizing a log-likelihood measure is developed. Both subjective and objective evaluations were conducted and indicate a considerable improvement over the conventional method.

Gigabit Network Communications Research

DTIC Science & Technology

1992-12-31

additional BPF channels, raw bytesync support for video codecs, and others. All source file modifications were logged with RCS. Source and object trees were...34 (RFCs). 20 RFCs were published this quarter: RFC 1366: Gerich, E., " Guidelines for Management of IP Address Space", Merit, October 1992. RFC 1367...Topolcic, C., "Schedule for IP Address Space Management Guidelines ", CNRI, October 1992. RFC 1368: McMaster, D. (Synoptics Communications, Inc.), K

CAGE IIIA Distributed Simulation Design Methodology

DTIC Science & Technology

2014-05-01

2 VHF Very High Frequency VLC Video LAN Codec – an Open-source cross-platform multimedia player and framework VM Virtual Machine VOIP Voice Over...Implementing Defence Experimentation (GUIDEx). The key challenges for this methodology are with understanding how to: • design it o define the...operation and to be available in the other nation’s simulations. The challenge for the CAGE campaign of experiments is to continue to build upon this

Coprime and Nested Arrays: A New Paradigm for Sampling in Space and Time

DTIC Science & Technology

2015-09-14

International Conference on Computers and Devices for Communication (CODEC), Kolkata, India , December, 2015. 2) Plenary speaker at the Asia Pacific...National Symposium on Mathematical Methods and Applications, Chennai, India , Dec. 2013. This is held to honor the late Srinivasa Ramanujan every year...4) Plenary speaker at the International Conf. on Comm. and Signal Processing (ICCSP), Calicut, India , 2011. List of publications during the above

An improvement analysis on video compression using file segmentation

NASA Astrophysics Data System (ADS)

Sharma, Shubhankar; Singh, K. John; Priya, M.

2017-11-01

From the past two decades the extreme evolution of the Internet has lead a massive rise in video technology and significantly video consumption over the Internet which inhabits the bulk of data traffic in general. Clearly, video consumes that so much data size on the World Wide Web, to reduce the burden on the Internet and deduction of bandwidth consume by video so that the user can easily access the video data.For this, many video codecs are developed such as HEVC/H.265 and V9. Although after seeing codec like this one gets a dilemma of which would be improved technology in the manner of rate distortion and the coding standard.This paper gives a solution about the difficulty for getting low delay in video compression and video application e.g. ad-hoc video conferencing/streaming or observation by surveillance. Also this paper describes the benchmark of HEVC and V9 technique of video compression on subjective oral estimations of High Definition video content, playback on web browsers. Moreover, this gives the experimental ideology of dividing the video file into several segments for compression and putting back together to improve the efficiency of video compression on the web as well as on the offline mode.

Relationship Among Signal Fidelity, Hearing Loss, and Working Memory for Digital Noise Suppression.

PubMed

Arehart, Kathryn; Souza, Pamela; Kates, James; Lunner, Thomas; Pedersen, Michael Syskind

2015-01-01

This study considered speech modified by additive babble combined with noise-suppression processing. The purpose was to determine the relative importance of the signal modifications, individual peripheral hearing loss, and individual cognitive capacity on speech intelligibility and speech quality. The participant group consisted of 31 individuals with moderate high-frequency hearing loss ranging in age from 51 to 89 years (mean = 69.6 years). Speech intelligibility and speech quality were measured using low-context sentences presented in babble at several signal-to-noise ratios. Speech stimuli were processed with a binary mask noise-suppression strategy with systematic manipulations of two parameters (error rate and attenuation values). The cumulative effects of signal modification produced by babble and signal processing were quantified using an envelope-distortion metric. Working memory capacity was assessed with a reading span test. Analysis of variance was used to determine the effects of signal processing parameters on perceptual scores. Hierarchical linear modeling was used to determine the role of degree of hearing loss and working memory capacity in individual listener response to the processed noisy speech. The model also considered improvements in envelope fidelity caused by the binary mask and the degradations to envelope caused by error and noise. The participants showed significant benefits in terms of intelligibility scores and quality ratings for noisy speech processed by the ideal binary mask noise-suppression strategy. This benefit was observed across a range of signal-to-noise ratios and persisted when up to a 30% error rate was introduced into the processing. Average intelligibility scores and average quality ratings were well predicted by an objective metric of envelope fidelity. Degree of hearing loss and working memory capacity were significant factors in explaining individual listener's intelligibility scores for binary mask processing applied to speech in babble. Degree of hearing loss and working memory capacity did not predict listeners' quality ratings. The results indicate that envelope fidelity is a primary factor in determining the combined effects of noise and binary mask processing for intelligibility and quality of speech presented in babble noise. Degree of hearing loss and working memory capacity are significant factors in explaining variability in listeners' speech intelligibility scores but not in quality ratings.

A generalized time-frequency subtraction method for robust speech enhancement based on wavelet filter banks modeling of human auditory system.

PubMed

Shao, Yu; Chang, Chip-Hong

2007-08-01

We present a new speech enhancement scheme for a single-microphone system to meet the demand for quality noise reduction algorithms capable of operating at a very low signal-to-noise ratio. A psychoacoustic model is incorporated into the generalized perceptual wavelet denoising method to reduce the residual noise and improve the intelligibility of speech. The proposed method is a generalized time-frequency subtraction algorithm, which advantageously exploits the wavelet multirate signal representation to preserve the critical transient information. Simultaneous masking and temporal masking of the human auditory system are modeled by the perceptual wavelet packet transform via the frequency and temporal localization of speech components. The wavelet coefficients are used to calculate the Bark spreading energy and temporal spreading energy, from which a time-frequency masking threshold is deduced to adaptively adjust the subtraction parameters of the proposed method. An unvoiced speech enhancement algorithm is also integrated into the system to improve the intelligibility of speech. Through rigorous objective and subjective evaluations, it is shown that the proposed speech enhancement system is capable of reducing noise with little speech degradation in adverse noise environments and the overall performance is superior to several competitive methods.

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

NASA Astrophysics Data System (ADS)

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

1982-04-01

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

Speech rate and fluency in children with phonological disorder.

PubMed

Novaes, Priscila Maronezi; Nicolielo-Carrilho, Ana Paola; Lopes-Herrera, Simone Aparecida

2015-01-01

To identify and describe the speech rate and fluency of children with phonological disorder (PD) with and without speech-language therapy. Thirty children, aged 5-8 years old, both genders, were divided into three groups: experimental group 1 (G1) — 10 children with PD in intervention; experimental group 2 (G2) — 10 children with PD without intervention; and control group (CG) — 10 children with typical development. Speech samples were collected and analyzed according to parameters of specific protocol. The children in CG had higher number of words per minute compared to those in G1, which, in turn, performed better in this aspect compared to children in G2. Regarding the number of syllables per minute, the CG showed the best result. In this aspect, the children in G1 showed better results than those in G2. Comparing children's performance in the assessed groups regarding the tests, those with PD in intervention had higher time of speech sample and adequate speech rate, which may be indicative of greater auditory monitoring of their own speech as a result of the intervention.

Speech prosody impairment predicts cognitive decline in Parkinson's disease.

PubMed

Rektorova, Irena; Mekyska, Jiri; Janousova, Eva; Kostalova, Milena; Eliasova, Ilona; Mrackova, Martina; Berankova, Dagmar; Necasova, Tereza; Smekal, Zdenek; Marecek, Radek

2016-08-01

Impairment of speech prosody is characteristic for Parkinson's disease (PD) and does not respond well to dopaminergic treatment. We assessed whether baseline acoustic parameters, alone or in combination with other predominantly non-dopaminergic symptoms may predict global cognitive decline as measured by the Addenbrooke's cognitive examination (ACE-R) and/or worsening of cognitive status as assessed by a detailed neuropsychological examination. Forty-four consecutive non-depressed PD patients underwent clinical and cognitive testing, and acoustic voice analysis at baseline and at the two-year follow-up. Influence of speech and other clinical parameters on worsening of the ACE-R and of the cognitive status was analyzed using linear and logistic regression. The cognitive status (classified as normal cognition, mild cognitive impairment and dementia) deteriorated in 25% of patients during the follow-up. The multivariate linear regression model consisted of the variation in range of the fundamental voice frequency (F0VR) and the REM Sleep Behavioral Disorder Screening Questionnaire (RBDSQ). These parameters explained 37.2% of the variability of the change in ACE-R. The most significant predictors in the univariate logistic regression were the speech index of rhythmicity (SPIR; p = 0.012), disease duration (p = 0.019), and the RBDSQ (p = 0.032). The multivariate regression analysis revealed that SPIR alone led to 73.2% accuracy in predicting a change in cognitive status. Combining SPIR with RBDSQ improved the prediction accuracy of SPIR alone by 7.3%. Impairment of speech prosody together with symptoms of RBD predicted rapid cognitive decline and worsening of PD cognitive status during a two-year period. Copyright © 2016 Elsevier Ltd. All rights reserved.

Systematic Studies of Modified Vocalization: Effects of Speech Rate and Instatement Style during Metronome Stimulation

ERIC Educational Resources Information Center

Davidow, Jason H.; Bothe, Anne K.; Richardson, Jessica D.; Andreatta, Richard D.

2010-01-01

Purpose: This study introduces a series of systematic investigations intended to clarify the parameters of the fluency-inducing conditions (FICs) in stuttering. Method: Participants included 11 adults, aged 20-63 years, with typical speech-production skills. A repeated measures design was used to examine the relationships between several speech…

The Use of Artificial Neural Networks to Estimate Speech Intelligibility from Acoustic Variables: A Preliminary Analysis.

ERIC Educational Resources Information Center

Metz, Dale Evan; And Others

1992-01-01

A preliminary scheme for estimating the speech intelligibility of hearing-impaired speakers from acoustic parameters, using a computerized artificial neural network to process mathematically the acoustic input variables, is outlined. Tests with 60 hearing-impaired speakers found the scheme to be highly accurate in identifying speakers separated by…

Voice and Fluency Changes as a Function of Speech Task and Deep Brain Stimulation

ERIC Educational Resources Information Center

Van Lancker Sidtis, Diana; Rogers, Tiffany; Godier, Violette; Tagliati, Michele; Sidtis, John J.

2010-01-01

Purpose: Speaking, which naturally occurs in different modes or "tasks" such as conversation and repetition, relies on intact basal ganglia nuclei. Recent studies suggest that voice and fluency parameters are differentially affected by speech task. In this study, the authors examine the effects of subcortical functionality on voice and fluency,…

Effect of Bilateral Stimulation of the Subthalamic Nucleus on Different Speech Subsystems in Patients with Parkinson's Disease

ERIC Educational Resources Information Center

Putzer, Manfred; Barry, William J.; Moringlane, Jean Richard

2008-01-01

The effect of deep brain stimulation on the two speech-production subsystems, articulation and phonation, of nine Parkinsonian patients is examined. Production parameters (stop closure voicing; stop closure, VOT, vowel) in fast syllable-repetitions were defined and measured and quantitative, objective metrics of vocal fold function were obtained…

Actor Vocal Training for the Habilitation of Speech in Adolescent Users of Cochlear Implants

ERIC Educational Resources Information Center

Holt, Colleen M.; Dowell, Richard C.

2011-01-01

This study examined changes to speech production in adolescents with hearing impairment following a period of actor vocal training. In addition to vocal parameters, the study also investigated changes to psychosocial factors such as confidence, self-esteem, and anxiety. The group were adolescent users of cochlear implants (mean age at commencement…

IQ, Non-Cognitive and Social-Emotional Parameters Influencing Education in Speech- and Language-Impaired Children

ERIC Educational Resources Information Center

Ullrich, Dieter; Ullrich, Katja; Marten, Magret

2017-01-01

Speech-/language-impaired (SL)-children face problems in school and later life. The significance of "non-cognitive, social-emotional skills" (NCSES) in these children is often underestimated. Aim: Present study of affected SL-children was assessed to analyse the influence of NCSES for long-term school education. Methods: Nineteen…

Effect of Parkinson's disease on the production of structured and unstructured speaking tasks: Respiratory physiologic and linguistic considerations

PubMed Central

Huber, Jessica E.; Darling, Meghan

2012-01-01

Purpose The purpose of the present study was to examine the effects of cognitive-linguistic deficits and respiratory physiologic changes on respiratory support for speech in PD, using two speech tasks, reading and extemporaneous speech. Methods Five women with PD, 9 men with PD, and 14 age- and sex-matched control participants read a passage and spoke extemporaneously on a topic of their choice at comfortable loudness. Sound pressure level, syllables per breath group, speech rate, and lung volume parameters were measured. Number of formulation errors, disfluencies, and filled pauses were counted. Results Individuals with PD produced shorter utterances as compared to control participants. The relationships between utterance length and lung volume initiation and inspiratory duration were weaker in individuals with PD than for control participants, particularly for the extemporaneous speech task. These results suggest less consistent planning for utterance length by individuals with PD in extemporaneous speech. Individuals with PD produced more formulation errors in both tasks and significantly fewer filled pauses in extemporaneous speech. Conclusions Both respiratory physiologic and cognitive-linguistic issues affected speech production by individuals with PD. Overall, individuals with PD had difficulty planning or coordinating language formulation and respiratory support, in particular during extemporaneous speech. PMID:20844256

Hybrid simulated annealing and its application to optimization of hidden Markov models for visual speech recognition.

PubMed

Lee, Jong-Seok; Park, Cheol Hoon

2010-08-01

We propose a novel stochastic optimization algorithm, hybrid simulated annealing (SA), to train hidden Markov models (HMMs) for visual speech recognition. In our algorithm, SA is combined with a local optimization operator that substitutes a better solution for the current one to improve the convergence speed and the quality of solutions. We mathematically prove that the sequence of the objective values converges in probability to the global optimum in the algorithm. The algorithm is applied to train HMMs that are used as visual speech recognizers. While the popular training method of HMMs, the expectation-maximization algorithm, achieves only local optima in the parameter space, the proposed method can perform global optimization of the parameters of HMMs and thereby obtain solutions yielding improved recognition performance. The superiority of the proposed algorithm to the conventional ones is demonstrated via isolated word recognition experiments.

Spectral analysis method and sample generation for real time visualization of speech

NASA Astrophysics Data System (ADS)

Hobohm, Klaus

A method for translating speech signals into optical models, characterized by high sound discrimination and learnability and designed to provide to deaf persons a feedback towards control of their way of speaking, is presented. Important properties of speech production and perception processes and organs involved in these mechanisms are recalled in order to define requirements for speech visualization. It is established that the spectral representation of time, frequency and amplitude resolution of hearing must be fair and continuous variations of acoustic parameters of speech signal must be depicted by a continuous variation of images. A color table was developed for dynamic illustration and sonograms were generated with five spectral analysis methods such as Fourier transformations and linear prediction coding. For evaluating sonogram quality, test persons had to recognize consonant/vocal/consonant words and an optimized analysis method was achieved with a fast Fourier transformation and a postprocessor. A hardware concept of a real time speech visualization system, based on multiprocessor technology in a personal computer, is presented.

Evaluation of Spectral and Prosodic Features of Speech Affected by Orthodontic Appliances Using the Gmm Classifier

NASA Astrophysics Data System (ADS)

Přibil, Jiří; Přibilová, Anna; Ďuračkoá, Daniela

2014-01-01

The paper describes our experiment with using the Gaussian mixture models (GMM) for classification of speech uttered by a person wearing orthodontic appliances. For the GMM classification, the input feature vectors comprise the basic and the complementary spectral properties as well as the supra-segmental parameters. Dependence of classification correctness on the number of the parameters in the input feature vector and on the computation complexity is also evaluated. In addition, an influence of the initial setting of the parameters for GMM training process was analyzed. Obtained recognition results are compared visually in the form of graphs as well as numerically in the form of tables and confusion matrices for tested sentences uttered using three configurations of orthodontic appliances.

Vocal effectiveness of speech-language pathology students: Before and after voice use during service delivery.

PubMed

Couch, Stephanie; Zieba, Dominique; Van der Linde, Jeannie; Van der Merwe, Anita

2015-03-26

As a professional voice user, it is imperative that a speech-language pathologist's(SLP) vocal effectiveness remain consistent throughout the day. Many factors may contribute to reduced vocal effectiveness, including prolonged voice use, vocally abusive behaviours,poor vocal hygiene and environmental factors. To determine the effect of service delivery on the perceptual and acoustic features of voice. A quasi-experimental., pre-test-post-test research design was used. Participants included third- and final-year speech-language pathology students at the University of Pretoria(South Africa). Voice parameters were evaluated in a pre-test measurement, after which the participants provided two consecutive hours of therapy. A post-test measurement was then completed. Data analysis consisted of an instrumental analysis in which the multidimensional voice programme (MDVP) and the voice range profile (VRP) were used to measure vocal parameters and then calculate the dysphonia severity index (DSI). The GRBASI scale was used to conduct a perceptual analysis of voice quality. Data were processed using descriptive statistics to determine change in each measured parameter after service delivery. A change of clinical significance was observed in the acoustic and perceptual parameters of voice. Guidelines for SLPs in order to maintain optimal vocal effectiveness were suggested.

Vocal effectiveness of speech-language pathology students: Before and after voice use during service delivery

PubMed Central

Couch, Stephanie; Zieba, Dominique; van der Merwe, Anita

2015-01-01

Background As a professional voice user, it is imperative that a speech-language pathologist's (SLP) vocal effectiveness remain consistent throughout the day. Many factors may contribute to reduced vocal effectiveness, including prolonged voice use, vocally abusive behaviours, poor vocal hygiene and environmental factors. Objectives To determine the effect of service delivery on the perceptual and acoustic features of voice. Method A quasi-experimental., pre-test–post-test research design was used. Participants included third- and final-year speech-language pathology students at the University of Pretoria (South Africa). Voice parameters were evaluated in a pre-test measurement, after which the participants provided two consecutive hours of therapy. A post-test measurement was then completed. Data analysis consisted of an instrumental analysis in which the multidimensional voice programme (MDVP) and the voice range profile (VRP) were used to measure vocal parameters and then calculate the dysphonia severity index (DSI). The GRBASI scale was used to conduct a perceptual analysis of voice quality. Data were processed using descriptive statistics to determine change in each measured parameter after service delivery. Results A change of clinical significance was observed in the acoustic and perceptual parameters of voice. Conclusion Guidelines for SLPs in order to maintain optimal vocal effectiveness were suggested. PMID:26304213

[Relevance of psychosocial factors in speech rehabilitation after laryngectomy].

PubMed

Singer, S; Fuchs, M; Dietz, A; Klemm, E; Kienast, U; Meyer, A; Oeken, J; Täschner, R; Wulke, C; Schwarz, R

2007-12-01

Often it is assumed that psychosocial and sociodemographic factors cause the success of voice rehabilitation after laryngectomy. Aim of this study was to analyze the association between these parameters. Based on tumor registries of six ENT-clinics all patients were surveyed, who were laryngectomized in the years before (N = 190). Success of voice rehabilitation has been assessed as speech intelligibility measured with the postlaryngectomy-telephone-intelligibility-test. For the assessment of the psychosocial parameters validated and standardized instruments were used if possible. Statistical analysis was done by multiple logistic regression analysis. Low speech intelligibility is associated with reduced conversations (OR 0.970) and social activity (OR 1.049). Patients are more likely to talk with esophageal voice when their motivation for learning the new voice was high (OR 7.835) and when they assessed their speech therapist as important for their motivation (OR 4.794). The risk to communicate merely by whispering is higher when patients live together with a partner (OR 5.293), when they talk seldomly (OR 1.017) and when they are not very active in social contexts (OR 0.966). Psychosocial factors can only partly explain how voice rehabilitation after laryngectomy becomes a success. Speech intelligibility is associated with active communication behaviour, whereas the use of an esophageal voice is correlated with motivation. It seems that the gaining of tracheoesophageal puncture voice is independent of psychosocial factors.

Top-down modulation of auditory processing: effects of sound context, musical expertise and attentional focus.

PubMed

Tervaniemi, M; Kruck, S; De Baene, W; Schröger, E; Alter, K; Friederici, A D

2009-10-01

By recording auditory electrical brain potentials, we investigated whether the basic sound parameters (frequency, duration and intensity) are differentially encoded among speech vs. music sounds by musicians and non-musicians during different attentional demands. To this end, a pseudoword and an instrumental sound of comparable frequency and duration were presented. The accuracy of neural discrimination was tested by manipulations of frequency, duration and intensity. Additionally, the subjects' attentional focus was manipulated by instructions to ignore the sounds while watching a silent movie or to attentively discriminate the different sounds. In both musicians and non-musicians, the pre-attentively evoked mismatch negativity (MMN) component was larger to slight changes in music than in speech sounds. The MMN was also larger to intensity changes in music sounds and to duration changes in speech sounds. During attentional listening, all subjects more readily discriminated changes among speech sounds than among music sounds as indexed by the N2b response strength. Furthermore, during attentional listening, musicians displayed larger MMN and N2b than non-musicians for both music and speech sounds. Taken together, the data indicate that the discriminative abilities in human audition differ between music and speech sounds as a function of the sound-change context and the subjective familiarity of the sound parameters. These findings provide clear evidence for top-down modulatory effects in audition. In other words, the processing of sounds is realized by a dynamically adapting network considering type of sound, expertise and attentional demands, rather than by a strictly modularly organized stimulus-driven system.

Immediate effects of AAF devices on the characteristics of stuttering: a clinical analysis.

PubMed

Unger, Julia P; Glück, Christian W; Cholewa, Jürgen

2012-06-01

The present study investigated the immediate effects of altered auditory feedback (AAF) and one Inactive Condition (AAF parameters set to 0) on clinical attributes of stuttering during scripted and spontaneous speech. Two commercially available, portable AAF devices were used to create the combined delayed auditory feedback (DAF) and frequency altered feedback (FAF) effects. Thirty adults, who stutter, aged 18-68 years (M=36.5; SD=15.2), participated in this investigation. Each subject produced four sets of 5-min of oral reading, three sets of 5-min monologs as well as 10-min dialogs. These speech samples were analyzed to detect changes in descriptive features of stuttering (frequency, duration, speech/articulatory rate, core behaviors) across the various speech samples and within two SSI-4 (Riley, 2009) based severity ratings. A statistically significant difference was found in the frequency of stuttered syllables (%SS) during both Active Device conditions (p=.000) for all speech samples. The most sizable reductions in %SS occurred within scripted speech. In the analysis of stuttering type, it was found that blocks were reduced significantly (Device A: p=.017; Device B: p=.049). To evaluate the impact on severe and mild stuttering, participants were grouped into two SSI-4 based categories; mild and moderate-severe. During the Inactive Condition those participants within the moderate-severe group (p=.024) showed a statistically significant reduction in overall disfluencies. This result indicates, that active AAF parameters alone may not be the sole cause of a fluency-enhancement when using a technical speech aid. The reader will learn and be able to describe: (1) currently available scientific evidence on the use of altered auditory feedback (AAF) during scripted and spontaneous speech, (2) which characteristics of stuttering are impacted by an AAF device (frequency, duration, core behaviors, speech & articulatory rate, stuttering severity), (3) the effects of an Inactive Condition on people who stutter (PWS) falling into two severity groups, and (4) how the examined participants perceived the use of AAF devices. Copyright © 2012 Elsevier Inc. All rights reserved.

An experimental version of the MZT (speech-from-text) system with external F(sub 0) control

NASA Astrophysics Data System (ADS)

Nowak, Ignacy

1994-12-01

The version of a Polish speech from text system described in this article was developed using the speech-from-text system. The new system has additional functions which make it possible to enter commands in edited orthographic text to control the phrase component and accentuation parameters. This makes it possible to generate a series of modified intonation contours in the texts spoken by the system. The effects obtained are made easier to control by a graphic illustration of the base frequency pattern in phrases that were last 'spoken' by the system. This version of the system was designed as a test prototype which will help us expand and refine our set of rules for automatic generation of intonation contours, which in turn will enable the fully automated speech-from-text system to generate speech with a more varied and precisely formed fundamental frequency pattern.

Cerebellar mutism--report of four cases.

PubMed

Ozimek, A; Richter, S; Hein-Kropp, C; Schoch, B; Gorissen, B; Kaiser, O; Gizewski, E; Ziegler, W; Timmann, D

2004-08-01

The aim of the present study was to investigate the manifestations of mutism after surgery in children with cerebellar tumors. Speech impairment following cerebellar mutism in children was investigated based on standardized acoustic speech parameters and perceptual criteria. Mutistic and non-mutistic children after cerebellar surgery as well as orthopedic controls were tested pre-and postoperatively. Speech impairment was compared with the localization of cerebellar lesions (i. e. affected lobules and nuclei). Whereas both control groups showed no abnormalities in speech and behavior, the mutistic group could be divided into children with dysarthria in post mutistic phase and children with mainly behavioral disturbances. In the mutistic children involvement of dentate and fastigial nuclei tended to be more frequent and extended than in the nonmutistic cerebellar children. Cerebellar mutism is a complex phenomenon of at least two types. Dysarthric symptoms during resolution of mutism support the anarthria hypothesis, while mainly behavioral changes suggest an explanation independent from speech motor control.

MPCM: a hardware coder for super slow motion video sequences

NASA Astrophysics Data System (ADS)

Alcocer, Estefanía; López-Granado, Otoniel; Gutierrez, Roberto; Malumbres, Manuel P.

2013-12-01

In the last decade, the improvements in VLSI levels and image sensor technologies have led to a frenetic rush to provide image sensors with higher resolutions and faster frame rates. As a result, video devices were designed to capture real-time video at high-resolution formats with frame rates reaching 1,000 fps and beyond. These ultrahigh-speed video cameras are widely used in scientific and industrial applications, such as car crash tests, combustion research, materials research and testing, fluid dynamics, and flow visualization that demand real-time video capturing at extremely high frame rates with high-definition formats. Therefore, data storage capability, communication bandwidth, processing time, and power consumption are critical parameters that should be carefully considered in their design. In this paper, we propose a fast FPGA implementation of a simple codec called modulo-pulse code modulation (MPCM) which is able to reduce the bandwidth requirements up to 1.7 times at the same image quality when compared with PCM coding. This allows current high-speed cameras to capture in a continuous manner through a 40-Gbit Ethernet point-to-point access.

Motor Proficiency of 6- to 9-Year-Old Children with Speech and Language Problems

ERIC Educational Resources Information Center

Visscher, Chris; Houwen, Suzanne; Moolenaar, Ben; Lyons, Jim; Scherder, Erik J. A.; Hartman, Esther

2010-01-01

Aim: This study compared the gross motor skills of school-age children (mean age 7y 8mo, range 6-9y) with developmental speech and language disorders (DSLDs; n = 105; 76 males, 29 females) and typically developing children (n = 105; 76 males, 29 females). The relationship between the performance parameters and the children's age was investigated…

The Effects of Hearing Aid Compression Parameters on the Short-Term Dynamic Range of Continuous Speech

ERIC Educational Resources Information Center

Henning, Rebecca L. Warner; Bentler, Ruth A.

2008-01-01

Purpose: The purpose of this study was to evaluate and quantitatively model the independent and interactive effects of compression ratio, number of compression channels, and release time on the dynamic range of continuous speech. Method: A CD of the Rainbow Passage (J. E. Bernthal & N. W. Bankson, 1993) was used. The hearing aid was a…

Acoustic analysis in Mudejar-Gothic churches: Experimental results

NASA Astrophysics Data System (ADS)

Galindo, Miguel; Zamarreño, Teófilo; Girón, Sara

2005-05-01

This paper describes the preliminary results of research work in acoustics, conducted in a set of 12 Mudejar-Gothic churches in the city of Seville in the south of Spain. Despite common architectural style, the churches feature individual characteristics and have volumes ranging from 3947 to 10 708 m3. Acoustic parameters were measured in unoccupied churches according to the ISO-3382 standard. An extensive experimental study was carried out using impulse response analysis through a maximum length sequence measurement system in each church. It covered aspects such as reverberation (reverberation times, early decay times), distribution of sound levels (sound strength); early to late sound energy parameters derived from the impulse responses (center time, clarity for speech, clarity, definition, lateral energy fraction), and speech intelligibility (rapid speech transmission index), which all take both spectral and spatial distribution into account. Background noise was also measured to obtain the NR indices. The study describes the acoustic field inside each temple and establishes a discussion for each one of the acoustic descriptors mentioned by using the theoretical models available and the principles of architectural acoustics. Analysis of the quality of the spaces for music and speech is carried out according to the most widespread criteria for auditoria. .

Acoustic analysis in Mudejar-Gothic churches: experimental results.

PubMed

Galindo, Miguel; Zamarreño, Teófilo; Girón, Sara

2005-05-01

This paper describes the preliminary results of research work in acoustics, conducted in a set of 12 Mudejar-Gothic churches in the city of Seville in the south of Spain. Despite common architectural style, the churches feature individual characteristics and have volumes ranging from 3947 to 10 708 m3. Acoustic parameters were measured in unoccupied churches according to the ISO-3382 standard. An extensive experimental study was carried out using impulse response analysis through a maximum length sequence measurement system in each church. It covered aspects such as reverberation (reverberation times, early decay times), distribution of sound levels (sound strength); early to late sound energy parameters derived from the impulse responses (center time, clarity for speech, clarity, definition, lateral energy fraction), and speech intelligibility (rapid speech transmission index), which all take both spectral and spatial distribution into account. Background noise was also measured to obtain the NR indices. The study describes the acoustic field inside each temple and establishes a discussion for each one of the acoustic descriptors mentioned by using the theoretical models available and the principles of architectural acoustics. Analysis of the quality of the spaces for music and speech is carried out according to the most widespread criteria for auditoria.

The Americleft Speech Project: A Training and Reliability Study.

PubMed

Chapman, Kathy L; Baylis, Adriane; Trost-Cardamone, Judith; Cordero, Kelly Nett; Dixon, Angela; Dobbelsteyn, Cindy; Thurmes, Anna; Wilson, Kristina; Harding-Bell, Anne; Sweeney, Triona; Stoddard, Gregory; Sell, Debbie

2016-01-01

To describe the results of two reliability studies and to assess the effect of training on interrater reliability scores. The first study (1) examined interrater and intrarater reliability scores (weighted and unweighted kappas) and (2) compared interrater reliability scores before and after training on the use of the Cleft Audit Protocol for Speech-Augmented (CAPS-A) with British English-speaking children. The second study examined interrater and intrarater reliability on a modified version of the CAPS-A (CAPS-A Americleft Modification) with American and Canadian English-speaking children. Finally, comparisons were made between the interrater and intrarater reliability scores obtained for Study 1 and Study 2. The participants were speech-language pathologists from the Americleft Speech Project. In Study 1, interrater reliability scores improved for 6 of the 13 parameters following training on the CAPS-A protocol. Comparison of the reliability results for the two studies indicated lower scores for Study 2 compared with Study 1. However, this appeared to be an artifact of the kappa statistic that occurred due to insufficient variability in the reliability samples for Study 2. When percent agreement scores were also calculated, the ratings appeared similar across Study 1 and Study 2. The findings of this study suggested that improvements in interrater reliability could be obtained following a program of systematic training. However, improvements were not uniform across all parameters. Acceptable levels of reliability were achieved for those parameters most important for evaluation of velopharyngeal function.

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

NASA Astrophysics Data System (ADS)

Gao, Pei-pei; Liu, Feng

2016-10-01

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

Speech auditory brainstem response (speech ABR) characteristics depending on recording conditions, and hearing status: an experimental parametric study.

PubMed

Akhoun, Idrick; Moulin, Annie; Jeanvoine, Arnaud; Ménard, Mikael; Buret, François; Vollaire, Christian; Scorretti, Riccardo; Veuillet, Evelyne; Berger-Vachon, Christian; Collet, Lionel; Thai-Van, Hung

2008-11-15

Speech elicited auditory brainstem responses (Speech ABR) have been shown to be an objective measurement of speech processing in the brainstem. Given the simultaneous stimulation and recording, and the similarities between the recording and the speech stimulus envelope, there is a great risk of artefactual recordings. This study sought to systematically investigate the source of artefactual contamination in Speech ABR response. In a first part, we measured the sound level thresholds over which artefactual responses were obtained, for different types of transducers and experimental setup parameters. A watermelon model was used to model the human head susceptibility to electromagnetic artefact. It was found that impedances between the electrodes had a great effect on electromagnetic susceptibility and that the most prominent artefact is due to the transducer's electromagnetic leakage. The only artefact-free condition was obtained with insert-earphones shielded in a Faraday cage linked to common ground. In a second part of the study, using the previously defined artefact-free condition, we recorded speech ABR in unilateral deaf subjects and bilateral normal hearing subjects. In an additional control condition, Speech ABR was recorded with the insert-earphones used to deliver the stimulation, unplugged from the ears, so that the subjects did not perceive the stimulus. No responses were obtained from the deaf ear of unilaterally hearing impaired subjects, nor in the insert-out-of-the-ear condition in all the subjects, showing that Speech ABR reflects the functioning of the auditory pathways.

Two different phenomena in basic motor speech performance in premanifest Huntington disease.

PubMed

Skodda, Sabine; Grönheit, Wenke; Lukas, Carsten; Bellenberg, Barbara; von Hein, Sarah M; Hoffmann, Rainer; Saft, Carsten

2016-03-09

Dysarthria is a common feature in Huntington disease (HD). The aim of this cross-sectional pilot study was the description and objective analysis of different speech parameters with special emphasis on the aspect of speech timing of connected speech and nonspeech verbal utterances in premanifest HD (preHD). A total of 28 preHD mutation carriers and 28 age- and sex-matched healthy speakers had to perform a reading task and several syllable repetition tasks. Results of computerized acoustic analysis of different variables for the measurement of speech rate and regularity were correlated with clinical measures and MRI-based brain atrophy assessment by voxel-based morphometry. An impaired capacity to steadily repeat single syllables with higher variations in preHD compared to healthy controls was found (variance 1: Cohen d = 1.46). Notably, speech rate was increased compared to controls and showed correlations to the volume of certain brain areas known to be involved in the sensory-motor speech networks (net speech rate: Cohen d = 1.19). Furthermore, speech rate showed correlations to disease burden score, probability of disease onset, the estimated years to onset, and clinical measures like the cognitive score. Measurement of speech rate and regularity might be helpful additional tools for the monitoring of subclinical functional disability in preHD. As one of the possible causes for higher performance in preHD, we discuss huntingtin-dependent temporarily advantageous development processes of the brain. © 2016 American Academy of Neurology.

Study of acoustic correlates associate with emotional speech

NASA Astrophysics Data System (ADS)

Yildirim, Serdar; Lee, Sungbok; Lee, Chul Min; Bulut, Murtaza; Busso, Carlos; Kazemzadeh, Ebrahim; Narayanan, Shrikanth

2004-10-01

This study investigates the acoustic characteristics of four different emotions expressed in speech. The aim is to obtain detailed acoustic knowledge on how a speech signal is modulated by changes from neutral to a certain emotional state. Such knowledge is necessary for automatic emotion recognition and classification and emotional speech synthesis. Speech data obtained from two semi-professional actresses are analyzed and compared. Each subject produces 211 sentences with four different emotions; neutral, sad, angry, happy. We analyze changes in temporal and acoustic parameters such as magnitude and variability of segmental duration, fundamental frequency and the first three formant frequencies as a function of emotion. Acoustic differences among the emotions are also explored with mutual information computation, multidimensional scaling and acoustic likelihood comparison with normal speech. Results indicate that speech associated with anger and happiness is characterized by longer duration, shorter interword silence, higher pitch and rms energy with wider ranges. Sadness is distinguished from other emotions by lower rms energy and longer interword silence. Interestingly, the difference in formant pattern between [happiness/anger] and [neutral/sadness] are better reflected in back vowels such as /a/(/father/) than in front vowels. Detailed results on intra- and interspeaker variability will be reported.

Glove-talk II - a neural-network interface which maps gestures to parallel formant speech synthesizer controls.

PubMed

Fels, S S; Hinton, G E

1997-01-01

Glove-Talk II is a system which translates hand gestures to speech through an adaptive interface. Hand gestures are mapped continuously to ten control parameters of a parallel formant speech synthesizer. The mapping allows the hand to act as an artificial vocal tract that produces speech in real time. This gives an unlimited vocabulary in addition to direct control of fundamental frequency and volume. Currently, the best version of Glove-Talk II uses several input devices, a parallel formant speech synthesizer, and three neural networks. The gesture-to-speech task is divided into vowel and consonant production by using a gating network to weight the outputs of a vowel and a consonant neural network. The gating network and the consonant network are trained with examples from the user. The vowel network implements a fixed user-defined relationship between hand position and vowel sound and does not require any training examples from the user. Volume, fundamental frequency, and stop consonants are produced with a fixed mapping from the input devices. With Glove-Talk II, the subject can speak slowly but with far more natural sounding pitch variations than a text-to-speech synthesizer.

Toward dynamic magnetic resonance imaging of the vocal tract during speech production.

PubMed

Ventura, Sandra M Rua; Freitas, Diamantino Rui S; Tavares, João Manuel R S

2011-07-01

The most recent and significant magnetic resonance imaging (MRI) improvements allow for the visualization of the vocal tract during speech production, which has been revealed to be a powerful tool in dynamic speech research. However, a synchronization technique with enhanced temporal resolution is still required. The study design was transversal in nature. Throughout this work, a technique for the dynamic study of the vocal tract with MRI by using the heart's signal to synchronize and trigger the imaging-acquisition process is presented and described. The technique in question is then used in the measurement of four speech articulatory parameters to assess three different syllables (articulatory gestures) of European Portuguese Language. The acquired MR images are automatically reconstructed so as to result in a variable sequence of images (slices) of different vocal tract shapes in articulatory positions associated with Portuguese speech sounds. The knowledge obtained as a result of the proposed technique represents a direct contribution to the improvement of speech synthesis algorithms, thereby allowing for novel perceptions in coarticulation studies, in addition to providing further efficient clinical guidelines in the pursuit of more proficient speech rehabilitation processes. Copyright © 2011 The Voice Foundation. Published by Mosby, Inc. All rights reserved.

Evidence for a perception of prosodic cues in bat communication: contact call classification by Megaderma lyra.

PubMed

Janssen, Simone; Schmidt, Sabine

2009-07-01

The perception of prosodic cues in human speech may be rooted in mechanisms common to mammals. The present study explores to what extent bats use rhythm and frequency, typically carrying prosodic information in human speech, for the classification of communication call series. Using a two-alternative, forced choice procedure, we trained Megaderma lyra to discriminate between synthetic contact call series differing in frequency, rhythm on level of calls and rhythm on level of call series, and measured the classification performance for stimuli differing in only one, or two, of the above parameters. A comparison with predictions from models based on one, combinations of two, or all, parameters revealed that the bats based their decision predominantly on frequency and in addition on rhythm on the level of call series, whereas rhythm on level of calls was not taken into account in this paradigm. Moreover, frequency and rhythm on the level of call series were evaluated independently. Our results show that parameters corresponding to prosodic cues in human languages are perceived and evaluated by bats. Thus, these necessary prerequisites for a communication via prosodic structures in mammals have evolved far before human speech.

Cochlear implanted children present vocal parameters within normal standards.

PubMed

de Souza, Lourdes Bernadete Rocha; Bevilacqua, Maria Cecília; Brasolotto, Alcione Ghedini; Coelho, Ana Cristina

2012-08-01

to compare acoustic and perceptual parameters regarding the voice of cochlear implanted children, with normal hearing children. this is a cross-sectional, quantitative and qualitative study. Thirty six cochlear implanted children aged between 3 y and 3 m to 5 y and 9 m and 25 children with normal hearing, aged between 3 y and 11 m and 6 y and 6 m, participated in this study. The recordings and the acoustics analysis of the sustained vowel/a/and spontaneous speech were performed using the PRAAT program. The parameters analyzed for the sustained vowel were the mean of the fundamental frequency, jitter, shimmer and harmonic-to-noise ratio (HNR). For the spontaneous speech, the minimum and maximum frequencies and the number of semitones were extracted. The perceptual analysis of the speech material was analyzed using visual-analogical scales of 100 points, composing the aspects related to the overall severity of the vocal deviation, roughness, breathiness, strain, pitch, loudness and resonance deviation, and instability. This last parameter was only analyzed for the sustained vowel. The results demonstrated that the majority of the vocal parameters analyzed in the samples of the implanted children disclosed values similar to those obtained by the group of children with normal hearing. implanted children who participate in a (re) habilitation and follow-up program, can present vocal characteristics similar to those vocal characteristics of children with normal hearing. Copyright © 2012 Elsevier Ireland Ltd. All rights reserved.

[Perception of emotional intonation of noisy speech signal with different acoustic parameters by adults of different age and gender].

PubMed

Dmitrieva, E S; Gel'man, V Ia

2011-01-01

The listener-distinctive features of recognition of different emotional intonations (positive, negative and neutral) of male and female speakers in the presence or absence of background noise were studied in 49 adults aged 20-79 years. In all the listeners noise produced the most pronounced decrease in recognition accuracy for positive emotional intonation ("joy") as compared to other intonations, whereas it did not influence the recognition accuracy of "anger" in 65-79-year-old listeners. The higher emotion recognition rates of a noisy signal were observed for speech emotional intonations expressed by female speakers. Acoustic characteristics of noisy and clear speech signals underlying perception of speech emotional prosody were found for adult listeners of different age and gender.

The effect of speaking style on a locus equation characterization of stop place of articulation.

PubMed

Sussman, H M; Dalston, E; Gumbert, S

1998-01-01

Locus equations were employed to assess the phonetic stability and distinctiveness of stop place categories in reduced speech. Twenty-two speakers produced stop consonant + vowel utterances in citation and spontaneous speech. Coarticulatory increases in hypoarticulated speech were documented only for /dV/ and [gV] productions in front vowel contexts. Coarticulatory extents for /bV/ and [gV] in back vowel contexts remained stable across style changes. Discriminant analyses showed equivalent levels of correct classification across speaking styles. CV reduction was quantified by use of Euclidean distances separating stop place categories. Despite sensitivity of locus equation parameters to articulatory differences encountered in informal speech, stop place categories still maintained a clear separability when plotted in a higher-order slope x y-intercept acoustic space.

Selective left, right and bilateral stimulation of subthalamic nuclei in Parkinson's disease: differential effects on motor, speech and language function.

PubMed

Schulz, Geralyn M; Hosey, Lara A; Bradberry, Trent J; Stager, Sheila V; Lee, Li-Ching; Pawha, Rajesh; Lyons, Kelly E; Metman, Leo Verhagen; Braun, Allen R

2012-01-01

Deep brain stimulation (DBS) of the subthalamic nucleus improves the motor symptoms of Parkinson's disease, but may produce a worsening of speech and language performance at rates and amplitudes typically selected in clinical practice. The possibility that these dissociated effects might be modulated by selective stimulation of left and right STN has never been systematically investigated. To address this issue, we analyzed motor, speech and language functions of 12 patients implanted with bilateral stimulators configured for optimal motor responses. Behavioral responses were quantified under four stimulator conditions: bilateral DBS, right-only DBS, left-only DBS and no DBS. Under bilateral and left-only DBS conditions, our results exhibited a significant improvement in motor symptoms but worsening of speech and language. These findings contribute to the growing body of literature demonstrating that bilateral STN DBS compromises speech and language function and suggests that these negative effects may be principally due to left-sided stimulation. These findings may have practical clinical consequences, suggesting that clinicians might optimize motor, speech and language functions by carefully adjusting left- and right-sided stimulation parameters.

Intelligent acoustic data fusion technique for information security analysis

NASA Astrophysics Data System (ADS)

Jiang, Ying; Tang, Yize; Lu, Wenda; Wang, Zhongfeng; Wang, Zepeng; Zhang, Luming

2017-08-01

Tone is an essential component of word formation in all tonal languages, and it plays an important role in the transmission of information in speech communication. Therefore, tones characteristics study can be applied into security analysis of acoustic signal by the means of language identification, etc. In speech processing, fundamental frequency (F0) is often viewed as representing tones by researchers of speech synthesis. However, regular F0 values may lead to low naturalness in synthesized speech. Moreover, F0 and tone are not equivalent linguistically; F0 is just a representation of a tone. Therefore, the Electroglottography (EGG) signal is collected for deeper tones characteristics study. In this paper, focusing on the Northern Kam language, which has nine tonal contours and five level tone types, we first collected EGG and speech signals from six natural male speakers of the Northern Kam language, and then achieved the clustering distributions of the tone curves. After summarizing the main characteristics of tones of Northern Kam, we analyzed the relationship between EGG and speech signal parameters, and laid the foundation for further security analysis of acoustic signal.

Factors prognostic for phonetic development after cleft palate repair.

PubMed

Lee, Joon Seok; Kim, Jae Bong; Lee, Jeong Woo; Yang, Jung Dug; Chung, Ho Yun; Cho, Byung Chae; Choi, Kang Young

2015-10-01

Palatoplasty is aimed to achieve normal speech, improve food intake, and ensure successful maxillary growth. However, the velopharyngeal function is harder to control than other functions. Therefore, many studies on the prognostic factor of velopharyngeal insufficiency have been conducted. This study aimed to evaluate the relationships between speech outcomes and multimodality based on intraoral and preoperative three-dimensional computerized tomographic (CT) findings. Among 73 children with cleft palate who underwent palatoplasty between April 2011 and August 2014 at Kyungpook National University Hospital (KNUH), 27 were retrospectively evaluated. The 27 cases were non-syndromic, for which successful speech evaluation was conducted by a single speech-language pathologist (Table 1). Successful speech evaluation was defined as performing the test three times in 6-month intervals. Three intraoral parameters were measured before and immediately after operation (Fig. 1). On axial- and coronal-view preoperative facial CT, 5 and 2 different parameters were analyzed, respectively (Figs. 2 and 3). Regression analysis (SPSS IBM 22.0) was used in the statistical analysis. Two-flap palatoplasty and Furlow's double opposing Z-plasty were performed in 15 and 12 patients, respectively. The operation was performed 11 months after birth on average. Children with a higher palatal arch and wider maxillary tuberosity distance showed hypernasality (p < 0.05; Table 2). The useful prognostic factors of velopharyngeal function after palatoplasty were palate width and height, rather than initial diagnosis, treatment method, or palate length. Therefore, a more active intervention is needed, such as orthopedic appliance, posterior pharyngeal wall augmentation, or early speech training. Copyright © 2015 European Association for Cranio-Maxillo-Facial Surgery. Published by Elsevier Ltd. All rights reserved.

Prediction of Beck Depression Inventory (BDI-II) Score Using Acoustic Measurements in a Sample of Iium Engineering Students

NASA Astrophysics Data System (ADS)

Fikri Zanil, Muhamad; Nur Wahidah Nik Hashim, Nik; Azam, Huda

2017-11-01

Psychiatrist currently relies on questionnaires and interviews for psychological assessment. These conservative methods often miss true positives and might lead to death, especially in cases where a patient might be experiencing suicidal predisposition but was only diagnosed as major depressive disorder (MDD). With modern technology, an assessment tool might aid psychiatrist with a more accurate diagnosis and thus hope to reduce casualty. This project will explore on the relationship between speech features of spoken audio signal (reading) in Bahasa Malaysia with the Beck Depression Inventory scores. The speech features used in this project were Power Spectral Density (PSD), Mel-frequency Ceptral Coefficients (MFCC), Transition Parameter, formant and pitch. According to analysis, the optimum combination of speech features to predict BDI-II scores include PSD, MFCC and Transition Parameters. The linear regression approach with sequential forward/backward method was used to predict the BDI-II scores using reading speech. The result showed 0.4096 mean absolute error (MAE) for female reading speech. For male, the BDI-II scores successfully predicted 100% less than 1 scores difference with MAE of 0.098437. A prediction system called Depression Severity Evaluator (DSE) was developed. The DSE managed to predict one out of five subjects. Although the prediction rate was low, the system precisely predict the score within the maximum difference of 4.93 for each person. This demonstrates that the scores are not random numbers.

Sentence Recognition Prediction for Hearing-impaired Listeners in Stationary and Fluctuation Noise With FADE

PubMed Central

Schädler, Marc René; Warzybok, Anna; Meyer, Bernd T.; Brand, Thomas

2016-01-01

To characterize the individual patient’s hearing impairment as obtained with the matrix sentence recognition test, a simulation Framework for Auditory Discrimination Experiments (FADE) is extended here using the Attenuation and Distortion (A+D) approach by Plomp as a blueprint for setting the individual processing parameters. FADE has been shown to predict the outcome of both speech recognition tests and psychoacoustic experiments based on simulations using an automatic speech recognition system requiring only few assumptions. It builds on the closed-set matrix sentence recognition test which is advantageous for testing individual speech recognition in a way comparable across languages. Individual predictions of speech recognition thresholds in stationary and in fluctuating noise were derived using the audiogram and an estimate of the internal level uncertainty for modeling the individual Plomp curves fitted to the data with the Attenuation (A-) and Distortion (D-) parameters of the Plomp approach. The “typical” audiogram shapes from Bisgaard et al with or without a “typical” level uncertainty and the individual data were used for individual predictions. As a result, the individualization of the level uncertainty was found to be more important than the exact shape of the individual audiogram to accurately model the outcome of the German Matrix test in stationary or fluctuating noise for listeners with hearing impairment. The prediction accuracy of the individualized approach also outperforms the (modified) Speech Intelligibility Index approach which is based on the individual threshold data only. PMID:27604782

Threats and Challenges in Reconfigurable Hardware Security

DTIC Science & Technology

2008-07-01

Same as Report (SAR) 18. NUMBER OF PAGES 12 19a. NAME OF RESPONSIBLE PERSON a. REPORT unclassified b . ABSTRACT unclassified c. THIS PAGE...g e F P G A MemoryMemory Codec A/DD/A F P G A B o a rd System Developers Avnet, Digilent, NuHorizons Application Development Stage Deployment...Rohatgi, and B . Sunar. Trojan detection using IC fingerprinting. In Proceedings of IEEE Symposium on Security and Privacy, 2007. [3] K. Austin. Data

Encryption for confidentiality of the network and influence of this to the quality of streaming video through network

NASA Astrophysics Data System (ADS)

Sevcik, L.; Uhrin, D.; Frnda, J.; Voznak, M.; Toral-Cruz, Homer; Mikulec, M.; Jakovlev, Sergej

2015-05-01

Nowadays, the interest in real-time services, like audio and video, is growing. These services are mostly transmitted over packet networks, which are based on IP protocol. It leads to analyses of these services and their behavior in such networks which are becoming more frequent. Video has become the significant part of all data traffic sent via IP networks. In general, a video service is one-way service (except e.g. video calls) and network delay is not such an important factor as in a voice service. Dominant network factors that influence the final video quality are especially packet loss, delay variation and the capacity of the transmission links. Analysis of video quality concentrates on the resistance of video codecs to packet loss in the network, which causes artefacts in the video. IPsec provides confidentiality in terms of safety, integrity and non-repudiation (using HMAC-SHA1 and 3DES encryption for confidentiality and AES in CBC mode) with an authentication header and ESP (Encapsulating Security Payload). The paper brings a detailed view of the performance of video streaming over an IP-based network. We compared quality of video with packet loss and encryption as well. The measured results demonstrated the relation between the video codec type and bitrate to the final video quality.

Sensitivity to Rhythmic Parameters in Dyslexic Children: A Comparison of Hungarian and English

ERIC Educational Resources Information Center

Suranyi, Zsuzsanna; Csepe, Valeria; Richardson, Ulla; Thomson, Jennifer M.; Honbolygo, Ferenc; Goswami, Usha

2009-01-01

It has been proposed that sensitivity to the parameters underlying speech rhythm may be important in setting up well-specified phonological representations in the mental lexicon. However, different acoustic parameters may contribute differentially to rhythm and stress in different languages. Here we contrast sensitivity to one such cue, amplitude…

Different Vocal Parameters Predict Perceptions of Dominance and Attractiveness.

PubMed

Hodges-Simeon, Carolyn R; Gaulin, Steven J C; Puts, David A

2010-12-01

Low mean fundamental frequency (F(0)) in men's voices has been found to positively influence perceptions of dominance by men and attractiveness by women using standardized speech. Using natural speech obtained during an ecologically valid social interaction, we examined relationships between multiple vocal parameters and dominance and attractiveness judgments. Male voices from an unscripted dating game were judged by men for physical and social dominance and by women in fertile and non-fertile menstrual cycle phases for desirability in short-term and long-term relationships. Five vocal parameters were analyzed: mean F(0) (an acoustic correlate of vocal fold size), F(0) variation, intensity (loudness), utterance duration, and formant dispersion (D(f), an acoustic correlate of vocal tract length). Parallel but separate ratings of speech transcripts served as controls for content. Multiple regression analyses were used to examine the independent contributions of each of the predictors. Physical dominance was predicted by low F(0) variation and physically dominant word content. Social dominance was predicted only by socially dominant word content. Ratings of attractiveness by women were predicted by low mean F(0), low D(f), high intensity, and attractive word content across cycle phase and mating context. Low D(f) was perceived as attractive by fertile-phase women only. We hypothesize that competitors and potential mates may attend more strongly to different components of men's voices because of the different types of information these vocal parameters provide.

Actor vocal training for the habilitation of speech in adolescent users of cochlear implants.

PubMed

Holt, Colleen M; Dowell, Richard C

2011-01-01

This study examined changes to speech production in adolescents with hearing impairment following a period of actor vocal training. In addition to vocal parameters, the study also investigated changes to psychosocial factors such as confidence, self-esteem, and anxiety. The group were adolescent users of cochlear implants (mean age at commencement of training 15.9 years), with approximately half of the group wearing a hearing aid in the contralateral ear. The mean age of implantation of the group was 7.6 years and the participants displayed a range of speech production abilities. Evaluation of posttraining outcomes was performed via a combination of perceptual and acoustic analyses. Significant posttraining changes to vocal parameters included increased pitch range and variability and decreased speaking rate. From a psychosocial perspective, posttraining stress levels were significantly lowered. This study suggested that actor vocal training may benefit young people with hearing impairment, both in the way in which they use their voices and in the way in which they view themselves.

Sinusoidal transform coding

NASA Technical Reports Server (NTRS)

Mcaulay, Robert J.; Quatieri, Thomas F.

1988-01-01

It has been shown that an analysis/synthesis system based on a sinusoidal representation of speech leads to synthetic speech that is essentially perceptually indistinguishable from the original. Strategies for coding the amplitudes, frequencies and phases of the sine waves have been developed that have led to a multirate coder operating at rates from 2400 to 9600 bps. The encoded speech is highly intelligible at all rates with a uniformly improving quality as the data rate is increased. A real-time fixed-point implementation has been developed using two ADSP2100 DSP chips. The methods used for coding and quantizing the sine-wave parameters for operation at the various frame rates are described.

Effect of deep brain stimulation on different speech subsystems in patients with multiple sclerosis.

PubMed

Pützer, Manfred; Barry, William John; Moringlane, Jean Richard

2007-11-01

The effect of deep brain stimulation on articulation and phonation subsystems in seven patients with multiple sclerosis (MS) was examined. Production parameters in fast syllable-repetitions were defined and measured, and the phonation quality during vowel productions was analyzed. Speech material was recorded for patients (with and without stimulation) and for a group of healthy control speakers. With stimulation, the precision of glottal and supraglottal articulatory gestures is reduced, whereas phonation has a greater tendency to be hyperfunctional in comparison with the healthy control data. Different effects on the two speech subsystems are induced by electrical stimulation of the thalamus in patients with MS.

A software tool for analyzing multichannel cochlear implant signals.

PubMed

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

2003-10-01

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

The minor third communicates sadness in speech, mirroring its use in music.

PubMed

Curtis, Meagan E; Bharucha, Jamshed J

2010-06-01

There is a long history of attempts to explain why music is perceived as expressing emotion. The relationship between pitches serves as an important cue for conveying emotion in music. The musical interval referred to as the minor third is generally thought to convey sadness. We reveal that the minor third also occurs in the pitch contour of speech conveying sadness. Bisyllabic speech samples conveying four emotions were recorded by 9 actresses. Acoustic analyses revealed that the relationship between the 2 salient pitches of the sad speech samples tended to approximate a minor third. Participants rated the speech samples for perceived emotion, and the use of numerous acoustic parameters as cues for emotional identification was modeled using regression analysis. The minor third was the most reliable cue for identifying sadness. Additional participants rated musical intervals for emotion, and their ratings verified the historical association between the musical minor third and sadness. These findings support the theory that human vocal expressions and music share an acoustic code for communicating sadness.

Design and performance of an analysis-by-synthesis class of predictive speech coders

NASA Technical Reports Server (NTRS)

Rose, Richard C.; Barnwell, Thomas P., III

1990-01-01

The performance of a broad class of analysis-by-synthesis linear predictive speech coders is quantified experimentally. The class of coders includes a number of well-known techniques as well as a very large number of speech coders which have not been named or studied. A general formulation for deriving the parametric representation used in all of the coders in the class is presented. A new coder, named the self-excited vocoder, is discussed because of its good performance with low complexity, and because of the insight this coder gives to analysis-by-synthesis coders in general. The results of a study comparing the performances of different members of this class are presented. The study takes the form of a series of formal subjective and objective speech quality tests performed on selected coders. The results of this study lead to some interesting and important observations concerning the controlling parameters for analysis-by-synthesis speech coders.

On mobile wireless ad hoc IP video transports

NASA Astrophysics Data System (ADS)

Kazantzidis, Matheos

2006-05-01

Multimedia transports in wireless, ad-hoc, multi-hop or mobile networks must be capable of obtaining information about the network and adaptively tune sending and encoding parameters to the network response. Obtaining meaningful metrics to guide a stable congestion control mechanism in the transport (i.e. passive, simple, end-to-end and network technology independent) is a complex problem. Equally difficult is obtaining a reliable QoS metrics that agrees with user perception in a client/server or distributed environment. Existing metrics, objective or subjective, are commonly used after or before to test or report on a transmission and require access to both original and transmitted frames. In this paper, we propose that an efficient and successful video delivery and the optimization of overall network QoS requires innovation in a) a direct measurement of available and bottleneck capacity for its congestion control and b) a meaningful subjective QoS metric that is dynamically reported to video sender. Once these are in place, a binomial -stable, fair and TCP friendly- algorithm can be used to determine the sending rate and other packet video parameters. An adaptive mpeg codec can then continually test and fit its parameters and temporal-spatial data-error control balance using the perceived QoS dynamic feedback. We suggest a new measurement based on a packet dispersion technique that is independent of underlying network mechanisms. We then present a binomial control based on direct measurements. We implement a QoS metric that is known to agree with user perception (MPQM) in a client/server, distributed environment by using predetermined table lookups and characterization of video content.

Dereverberation and denoising based on generalized spectral subtraction by multi-channel LMS algorithm using a small-scale microphone array

NASA Astrophysics Data System (ADS)

Wang, Longbiao; Odani, Kyohei; Kai, Atsuhiko

2012-12-01

A blind dereverberation method based on power spectral subtraction (SS) using a multi-channel least mean squares algorithm was previously proposed to suppress the reverberant speech without additive noise. The results of isolated word speech recognition experiments showed that this method achieved significant improvements over conventional cepstral mean normalization (CMN) in a reverberant environment. In this paper, we propose a blind dereverberation method based on generalized spectral subtraction (GSS), which has been shown to be effective for noise reduction, instead of power SS. Furthermore, we extend the missing feature theory (MFT), which was initially proposed to enhance the robustness of additive noise, to dereverberation. A one-stage dereverberation and denoising method based on GSS is presented to simultaneously suppress both the additive noise and nonstationary multiplicative noise (reverberation). The proposed dereverberation method based on GSS with MFT is evaluated on a large vocabulary continuous speech recognition task. When the additive noise was absent, the dereverberation method based on GSS with MFT using only 2 microphones achieves a relative word error reduction rate of 11.4 and 32.6% compared to the dereverberation method based on power SS and the conventional CMN, respectively. For the reverberant and noisy speech, the dereverberation and denoising method based on GSS achieves a relative word error reduction rate of 12.8% compared to the conventional CMN with GSS-based additive noise reduction method. We also analyze the effective factors of the compensation parameter estimation for the dereverberation method based on SS, such as the number of channels (the number of microphones), the length of reverberation to be suppressed, and the length of the utterance used for parameter estimation. The experimental results showed that the SS-based method is robust in a variety of reverberant environments for both isolated and continuous speech recognition and under various parameter estimation conditions.

Can we perceptually rate alaryngeal voice? Developing the Sunderland Tracheoesophageal Voice Perceptual Scale.

PubMed

Hurren, A; Hildreth, A J; Carding, P N

2009-12-01

To investigate the inter and intra reliability of raters (in relation to both profession and expertise) when judging two alaryngeal voice parameters: 'Overall Grade' and 'Neoglottal Tonicity'. Reliable perceptual assessment is essential for surgical and therapeutic outcome measurement but has been minimally researched to date. Test of inter and intra rater agreement from audio recordings of 55 tracheoesophageal speakers. Cancer Unit. Twelve speech and language therapists and ten Ear, Nose and Throat surgeons. Perceptual voice parameters of 'Overall Grade' rated with a 0-3 equally appearing interval scale and 'Neoglottal Tonicity' with an 11-point bipolar semantic scale. All raters achieved 'good' agreement for 'Overall Grade' with mean weighted kappa coefficients of 0.78 for intra and 0.70 for inter-rater agreement. All raters achieved 'good' intra-rater agreement for 'Neoglottal Tonicity' (0.64) but inter-rater agreement was only 'moderate' (0.40). However, the expert speech and language therapists sub-group attained 'good' inter-rater agreement with this parameter (0.63). The effect of 'Neoglottal Tonicity' on 'Overall Grade' was examined utilising only expert speech and language therapists data. Linear regression analysis resulted in an r-squared coefficient of 0.67. Analysis of the perceptual impression of hypotonicity and hypertonicity in relation to mean 'Overall Grade' score demonstrated neither tone was linked to a more favourable grade (P = 0.42). Expert speech and language therapist raters may be the optimal judges for tracheoesophageal voice assessment. Tonicity appears to be a good predictor of 'Overall Grade'. These scales have clinical applicability to investigate techniques that facilitate optotonic neoglottal voice quality.

Adaptive method of recognition of signals for one and two-frequency signal system in the telephony on the background of speech

NASA Astrophysics Data System (ADS)

Kuznetsov, Michael V.

2006-05-01

For reliable teamwork of various systems of automatic telecommunication including transferring systems of optical communication networks it is necessary authentic recognition of signals for one- or two-frequency service signal system. The analysis of time parameters of an accepted signal allows increasing reliability of detection and recognition of the service signal system on a background of speech.

Speech intelligibility in cerebral palsy children attending an art therapy program.

PubMed

Wilk, Magdalena; Pachalska, Maria; Lipowska, Małgorzata; Herman-Sucharska, Izabela; Makarowski, Ryszard; Mirski, Andrzej; Jastrzebowska, Grazyna

2010-05-01

Dysarthia is a common sequela of cerebral palsy (CP), directly affecting both the intelligibility of speech and the child's psycho-social adjustment. Speech therapy focused exclusively on the articulatory organs does not always help CP children to speak more intelligibly. The program of art therapy described here has proven to be helpful for these children. From among all the CP children enrolled in our art therapy program from 2005 to 2009, we selected a group of 14 boys and girls (average age 15.3) with severe dysarthria at baseline but no other language or cognitive disturbances. Our retrospective study was based on results from the Auditory Dysarthria Scale and neuropsychological tests for fluency, administered routinely over the 4 months of art therapy. All 14 children in the study group showed some degree of improvement after art therapy in all tested parameters. On the Auditory Dysarthia Scale, highly significant improvements were noted in overall intelligibility (p<0.0001), with significant improvement (p<0.001) in volume, tempo, and control of pauses. The least improvement was noted in the most purely motor parameters. All 14 children also exhibited significant improvement in fluency. Art therapy improves the intelligibility of speech in children with cerebral palsy, even when language functions are not as such the object of therapeutic intervention.

Test-retest reliability of speech-evoked auditory brainstem response in healthy children at a low sensation level.

PubMed

Zakaria, Mohd Normani; Jalaei, Bahram

2017-11-01

Auditory brainstem responses evoked by complex stimuli such as speech syllables have been studied in normal subjects and subjects with compromised auditory functions. The stability of speech-evoked auditory brainstem response (speech-ABR) when tested over time has been reported but the literature is limited. The present study was carried out to determine the test-retest reliability of speech-ABR in healthy children at a low sensation level. Seventeen healthy children (6 boys, 11 girls) aged from 5 to 9 years (mean = 6.8 ± 3.3 years) were tested in two sessions separated by a 3-month period. The stimulus used was a 40-ms syllable /da/ presented at 30 dB sensation level. As revealed by pair t-test and intra-class correlation (ICC) analyses, peak latencies, peak amplitudes and composite onset measures of speech-ABR were found to be highly replicable. Compared to other parameters, higher ICC values were noted for peak latencies of speech-ABR. The present study was the first to report the test-retest reliability of speech-ABR recorded at low stimulation levels in healthy children. Due to its good stability, it can be used as an objective indicator for assessing the effectiveness of auditory rehabilitation in hearing-impaired children in future studies. Copyright © 2017 Elsevier B.V. All rights reserved.

Exploring expressivity and emotion with artificial voice and speech technologies.

PubMed

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

2013-10-01

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

Proximate factors associated with speech intelligibility in children with cochlear implants: A preliminary study.

PubMed

Chin, Steven B; Kuhns, Matthew J

2014-01-01

The purpose of this descriptive pilot study was to examine possible relationships among speech intelligibility and structural characteristics of speech in children who use cochlear implants. The Beginners Intelligibility Test (BIT) was administered to 10 children with cochlear implants, and the intelligibility of the words in the sentences was judged by panels of naïve adult listeners. Additionally, several qualitative and quantitative measures of word omission, segment correctness, duration, and intonation variability were applied to the sentences used to assess intelligibility. Correlational analyses were conducted to determine if BIT scores and the other speech parameters were related. There was a significant correlation between BIT score and percent words omitted, but no other variables correlated significantly with BIT score. The correlation between intelligibility and word omission may be task-specific as well as reflective of memory limitations.

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

PubMed

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

2012-05-01

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

Glove-TalkII--a neural-network interface which maps gestures to parallel formant speech synthesizer controls.

PubMed

Fels, S S; Hinton, G E

1998-01-01

Glove-TalkII is a system which translates hand gestures to speech through an adaptive interface. Hand gestures are mapped continuously to ten control parameters of a parallel formant speech synthesizer. The mapping allows the hand to act as an artificial vocal tract that produces speech in real time. This gives an unlimited vocabulary in addition to direct control of fundamental frequency and volume. Currently, the best version of Glove-TalkII uses several input devices (including a Cyberglove, a ContactGlove, a three-space tracker, and a foot pedal), a parallel formant speech synthesizer, and three neural networks. The gesture-to-speech task is divided into vowel and consonant production by using a gating network to weight the outputs of a vowel and a consonant neural network. The gating network and the consonant network are trained with examples from the user. The vowel network implements a fixed user-defined relationship between hand position and vowel sound and does not require any training examples from the user. Volume, fundamental frequency, and stop consonants are produced with a fixed mapping from the input devices. One subject has trained to speak intelligibly with Glove-TalkII. He speaks slowly but with far more natural sounding pitch variations than a text-to-speech synthesizer.

Effects of WDRC release time and number of channels on output SNR and speech recognition

PubMed Central

Alexander, Joshua M.; Masterson, Katie

2014-01-01

Objectives The purpose of this study was to investigate the joint effects that wide dynamic range compression (WDRC) release time (RT) and number of channels have on recognition of sentences in the presence of steady and modulated maskers at different signal-to-noise ratios (SNRs). How the different combinations of WDRC parameters affect output SNR and the role this plays in the observed findings was also investigated. Design Twenty-four listeners with mild to moderate sensorineural hearing loss identified sentences mixed with steady or modulated maskers at 3 SNRs (−5, 0, +5 dB) that had been processed using a hearing aid simulator with 6 combinations of RT (40 and 640 ms) and number of channels (4, 8, and 16). Compression parameters were set using the Desired Sensation Level v5.0a prescriptive fitting method. For each condition, amplified speech and masker levels and the resultant long-term output SNR were measured. Results Speech recognition with WDRC depended on the combination of RT and number of channels, with the greatest effects observed at 0 dB input SNR, in which mean speech recognition scores varied by 10–12% across WDRC manipulations. Overall, effect sizes were generally small. Across both masker types and the three SNRs tested, the best speech recognition was obtained with 8 channels, regardless of RT. Increased speech levels, which favor audibility, were associated with the short RT and with an increase in the number of channels. These same conditions also increased masker levels by an even greater amount, for a net decrease in the long-term output SNR. Changes in long-term SNR across WDRC conditions were found to be strongly associated with changes in the temporal envelope shape as quantified by the Envelope Difference Index, however, neither of these factors fully explained the observed differences in speech recognition. Conclusions A primary finding of this study was that the number of channels had a modest effect when analyzed at each level of RT, with results suggesting that selecting 8 channels for a given RT might be the safest choice. Effects were smaller for RT, with results suggesting that short RT was slightly better when only 4 channels were used and that long RT was better when 16 channels were used. Individual differences in how listeners were influenced by audibility, output SNR, temporal distortion, and spectral distortion may have contributed to the size of the effects found in this study. Because only general suppositions could made for how each of these factors may have influenced the overall results of this study, future research would benefit from exploring the predictive value of these and other factors in selecting the processing parameters that maximize speech recognition for individuals. PMID:25470368

Impact of auditory training for perceptual assessment of voice executed by undergraduate students in Speech-Language Pathology.

PubMed

Silva, Regiane Serafim Abreu; Simões-Zenari, Marcia; Nemr, Nair Kátia

2012-01-01

To analyze the impact of auditory training for auditory-perceptual assessment carried out by Speech-Language Pathology undergraduate students. During two semesters, 17 undergraduate students enrolled in theoretical subjects regarding phonation (Phonation/Phonation Disorders) analyzed samples of altered and unaltered voices (selected for this purpose), using the GRBAS scale. All subjects received auditory training during nine 15-minute meetings. In each meeting, a different parameter was presented using the different voices sample, with predominance of the trained aspect in each session. Sample assessment using the scale was carried out before and after training, and in other four opportunities throughout the meetings. Students' assessments were compared to an assessment carried out by three voice-experts speech-language pathologists who were the judges. To verify training effectiveness, the Friedman's test and the Kappa index were used. The rate of correct answers in the pre-training was considered between regular and good. It was observed maintenance of the number of correct answers throughout assessments, for most of the scale parameters. In the post-training moment, the students showed improvements in the analysis of asthenia, a parameter that was emphasized during training after the students reported difficulties analyzing it. There was a decrease in the number of correct answers for the roughness parameter after it was approached segmented into hoarseness and harshness, and observed in association with different diagnoses and acoustic parameters. Auditory training enhances students' initial abilities to perform the evaluation, aside from guiding adjustments in the dynamics of the university subject.

Propagation considerations in the American Mobile Satellite system design

NASA Technical Reports Server (NTRS)

Kittiver, Charles; Sigler, Charles E., Jr.

1993-01-01

An overview of the American Mobile Satellite Corporation (AMSC) mobile satellite services (MSS) system with special emphasis given to the propagation issues that were considered in the design is presented. The aspects of the voice codec design that effect system performance in a shadowed environment are discussed. The strategies for overcoming Ku-Band rain fades in the uplink and downlink paths of the gateway station are presented. A land mobile propagation study that has both measurement and simulation activities is described.

Study of Potential Standardization of Digital Freeze Frame Video Codecs.

DTIC Science & Technology

1984-01-01

and MAR track an input clock over a very wide range. These are dependent on the modem used in any specific application. Interface connectors are those...terminals, 56K bit digital transmission sets). We have a limited custan capability and are not in the custom unit business. 1.,o .2e e.. , , 4g..2. . j...will) are designed for narrowband operation. We build our own modems which send .’e- pixels at a rate of 1969 pixels/second. Grey scale information is

Region of interest video coding for low bit-rate transmission of carotid ultrasound videos over 3G wireless networks.

PubMed

Tsapatsoulis, Nicolas; Loizou, Christos; Pattichis, Constantinos

2007-01-01

Efficient medical video transmission over 3G wireless is of great importance for fast diagnosis and on site medical staff training purposes. In this paper we present a region of interest based ultrasound video compression study which shows that significant reduction of the required, for transmission, bit rate can be achieved without altering the design of existing video codecs. Simple preprocessing of the original videos to define visually and clinically important areas is the only requirement.

Investigation of in-vehicle speech intelligibility metrics for normal hearing and hearing impaired listeners

NASA Astrophysics Data System (ADS)

Samardzic, Nikolina

The effectiveness of in-vehicle speech communication can be a good indicator of the perception of the overall vehicle quality and customer satisfaction. Currently available speech intelligibility metrics do not account in their procedures for essential parameters needed for a complete and accurate evaluation of in-vehicle speech intelligibility. These include the directivity and the distance of the talker with respect to the listener, binaural listening, hearing profile of the listener, vocal effort, and multisensory hearing. In the first part of this research the effectiveness of in-vehicle application of these metrics is investigated in a series of studies to reveal their shortcomings, including a wide range of scores resulting from each of the metrics for a given measurement configuration and vehicle operating condition. In addition, the nature of a possible correlation between the scores obtained from each metric is unknown. The metrics and the subjective perception of speech intelligibility using, for example, the same speech material have not been compared in literature. As a result, in the second part of this research, an alternative method for speech intelligibility evaluation is proposed for use in the automotive industry by utilizing a virtual reality driving environment for ultimately setting targets, including the associated statistical variability, for future in-vehicle speech intelligibility evaluation. The Speech Intelligibility Index (SII) was evaluated at the sentence Speech Receptions Threshold (sSRT) for various listening situations and hearing profiles using acoustic perception jury testing and a variety of talker and listener configurations and background noise. In addition, the effect of individual sources and transfer paths of sound in an operating vehicle to the vehicle interior sound, specifically their effect on speech intelligibility was quantified, in the framework of the newly developed speech intelligibility evaluation method. Lastly, as an example of the significance of speech intelligibility evaluation in the context of an applicable listening environment, as indicated in this research, it was found that the jury test participants required on average an approximate 3 dB increase in sound pressure level of speech material while driving and listening compared to when just listening, for an equivalent speech intelligibility performance and the same listening task.

The emerging phenotype of long-term survivors with infantile Pompe disease

PubMed Central

Prater, Sean N.; Banugaria, Suhrad G.; DeArmey, Stephanie M.; Botha, Eleanor G.; Stege, Erin M.; Case, Laura E.; Jones, Harrison N.; Phornphutkul, Chanika; Wang, Raymond Y.; Young, Sarah P.; Kishnani, Priya S.

2013-01-01

Purpose Enzyme replacement therapy with alglucosidase alfa for infantile Pompe disease has improved survival creating new management challenges. We describe an emerging phenotype in a retrospective review of long-term survivors. Methods Inclusion criteria included ventilator-free status and age ≤6 months at treatment initiation, and survival to age ≥5 years. Clinical outcome measures included invasive ventilator-free survival and parameters for cardiac, pulmonary, musculoskeletal, gross motor and ambulatory status; growth; speech, hearing, and swallowing; and gastrointestinal and nutritional status. Results Eleven of 17 patients met study criteria. All were cross-reactive immunologic material-positive, alive, and invasive ventilator-free at most recent assessment, with a median age of 8.0 years (range: 5.4 to 12.0 years). All had marked improvements in cardiac parameters. Commonly present were gross motor weakness, motor speech deficits, sensorineural and/or conductive hearing loss, osteopenia, gastroesophageal reflux disease, and dysphagia with aspiration risk. Seven of 11 patients were independently ambulatory and four required the use of assistive ambulatory devices. All long-term survivors had low or undetectable anti-alglucosidase alfa antibody titers. Conclusions Long-term survivors exhibited sustained improvements in cardiac parameters and gross motor function. Residual muscle weakness, hearing loss, risk for arrhythmias, hypernasal speech, dysphagia with risk for aspiration, and osteopenia were commonly observed findings. PMID:22538254

A novel probabilistic framework for event-based speech recognition

NASA Astrophysics Data System (ADS)

Juneja, Amit; Espy-Wilson, Carol

2003-10-01

One of the reasons for unsatisfactory performance of the state-of-the-art automatic speech recognition (ASR) systems is the inferior acoustic modeling of low-level acoustic-phonetic information in the speech signal. An acoustic-phonetic approach to ASR, on the other hand, explicitly targets linguistic information in the speech signal, but such a system for continuous speech recognition (CSR) is not known to exist. A probabilistic and statistical framework for CSR based on the idea of the representation of speech sounds by bundles of binary valued articulatory phonetic features is proposed. Multiple probabilistic sequences of linguistically motivated landmarks are obtained using binary classifiers of manner phonetic features-syllabic, sonorant and continuant-and the knowledge-based acoustic parameters (APs) that are acoustic correlates of those features. The landmarks are then used for the extraction of knowledge-based APs for source and place phonetic features and their binary classification. Probabilistic landmark sequences are constrained using manner class language models for isolated or connected word recognition. The proposed method could overcome the disadvantages encountered by the early acoustic-phonetic knowledge-based systems that led the ASR community to switch to systems highly dependent on statistical pattern analysis methods and probabilistic language or grammar models.

Automatic detection of obstructive sleep apnea using speech signals.

PubMed

Goldshtein, Evgenia; Tarasiuk, Ariel; Zigel, Yaniv

2011-05-01

Obstructive sleep apnea (OSA) is a common disorder associated with anatomical abnormalities of the upper airways that affects 5% of the population. Acoustic parameters may be influenced by the vocal tract structure and soft tissue properties. We hypothesize that speech signal properties of OSA patients will be different than those of control subjects not having OSA. Using speech signal processing techniques, we explored acoustic speech features of 93 subjects who were recorded using a text-dependent speech protocol and a digital audio recorder immediately prior to polysomnography study. Following analysis of the study, subjects were divided into OSA (n=67) and non-OSA (n=26) groups. A Gaussian mixture model-based system was developed to model and classify between the groups; discriminative features such as vocal tract length and linear prediction coefficients were selected using feature selection technique. Specificity and sensitivity of 83% and 79% were achieved for the male OSA and 86% and 84% for the female OSA patients, respectively. We conclude that acoustic features from speech signals during wakefulness can detect OSA patients with good specificity and sensitivity. Such a system can be used as a basis for future development of a tool for OSA screening. © 2011 IEEE

Voice recognition through phonetic features with Punjabi utterances

NASA Astrophysics Data System (ADS)

Kaur, Jasdeep; Juglan, K. C.; Sharma, Vishal; Upadhyay, R. K.

2017-07-01

This paper deals with perception and disorders of speech in view of Punjabi language. Visualizing the importance of voice identification, various parameters of speaker identification has been studied. The speech material was recorded with a tape recorder in their normal and disguised mode of utterances. Out of the recorded speech materials, the utterances free from noise, etc were selected for their auditory and acoustic spectrographic analysis. The comparison of normal and disguised speech of seven subjects is reported. The fundamental frequency (F0) at similar places, Plosive duration at certain phoneme, Amplitude ratio (A1:A2) etc. were compared in normal and disguised speech. It was found that the formant frequency of normal and disguised speech remains almost similar only if it is compared at the position of same vowel quality and quantity. If the vowel is more closed or more open in the disguised utterance the formant frequency will be changed in comparison to normal utterance. The ratio of the amplitude (A1: A2) is found to be speaker dependent. It remains unchanged in the disguised utterance. However, this value may shift in disguised utterance if cross sectioning is not done at the same location.

Determination of preferred parameters for multichannel compression using individually fitted simulated hearing AIDS and paired comparisons.

PubMed

Moore, Brian C J; Füllgrabe, Christian; Stone, Michael A

2011-01-01

To determine preferred parameters of multichannel compression using individually fitted simulated hearing aids and a method of paired comparisons. Fourteen participants with mild to moderate hearing loss listened via a simulated five-channel compression hearing aid fitted using the CAMEQ2-HF method to pairs of speech sounds (a male talker and a female talker) and musical sounds (a percussion instrument, orchestral classical music, and a jazz trio) presented sequentially and indicated which sound of the pair was preferred and by how much. The sounds in each pair were derived from the same token and differed along a single dimension in the type of processing applied. For the speech sounds, participants judged either pleasantness or clarity; in the latter case, the speech was presented in noise at a 2-dB signal-to-noise ratio. For musical sounds, they judged pleasantness. The parameters explored were time delay of the audio signal relative to the gain control signal (the alignment delay), compression speed (attack and release times), bandwidth (5, 7.5, or 10 kHz), and gain at high frequencies relative to that prescribed by CAMEQ2-HF. Pleasantness increased with increasing alignment delay only for the percussive musical sound. Clarity was not affected by alignment delay. There was a trend for pleasantness to decrease slightly with increasing bandwidth, but this was significant only for female speech with fast compression. Judged clarity was significantly higher for the 7.5- and 10-kHz bandwidths than for the 5-kHz bandwidth for both slow and fast compression and for both talker genders. Compression speed had little effect on pleasantness for 50- or 65-dB SPL input levels, but slow compression was generally judged as slightly more pleasant than fast compression for an 80-dB SPL input level. Clarity was higher for slow than for fast compression for input levels of 80 and 65 dB SPL but not for a level of 50 dB SPL. Preferences for pleasantness were approximately equal with CAMEQ2-HF gains and with gains slightly reduced at high frequencies and were lower when gains were slightly increased at high frequencies. Speech clarity was not affected by changing the gain at high frequencies. Effects of alignment delay were small except for the percussive sound. A wider bandwidth was slightly preferred for speech clarity. Speech clarity was slightly greater with slow compression, especially at high levels. Preferred high-frequency gains were close to or a little below those prescribed by CAMEQ2-HF.

A randomized controlled trial comparing two techniques for unilateral cleft lip and palate: Growth and speech outcomes during mixed dentition.

PubMed

Ganesh, Praveen; Murthy, Jyotsna; Ulaghanathan, Navitha; Savitha, V H

2015-07-01

To study the growth and speech outcomes in children who were operated on for unilateral cleft lip and palate (UCLP) by a single surgeon using two different treatment protocols. A total of 200 consecutive patients with nonsyndromic UCLP were randomly allocated to two different treatment protocols. Of the 200 patients, 179 completed the protocol. However, only 85 patients presented for follow-up during the mixed dentition period (7-10 years of age). The following treatment protocol was followed. Protocol 1 consisted of the vomer flap (VF), whereby patients underwent primary lip nose repair and vomer flap for hard palate single-layer closure, followed by soft palate repair 6 months later; Protocol 2 consisted of the two-flap technique (TF), whereby the cleft palate (CP) was repaired by two-flap technique after primary lip and nose repair. GOSLON Yardstick scores for dental arch relation, and speech outcomes based on universal reporting parameters, were noted. A total of 40 patients in the VF group and 45 in the TF group completed the treatment protocols. The GOSLON scores showed marginally better outcomes in the VF group compared to the TF group. Statistically significant differences were found only in two speech parameters, with better outcomes in the TF group. Our results showed marginally better growth outcome in the VF group compared to the TF group. However, the speech outcomes were better in the TF group. Copyright © 2015 European Association for Cranio-Maxillo-Facial Surgery. Published by Elsevier Ltd. All rights reserved.

Sentence Recognition Prediction for Hearing-impaired Listeners in Stationary and Fluctuation Noise With FADE: Empowering the Attenuation and Distortion Concept by Plomp With a Quantitative Processing Model.

PubMed

Kollmeier, Birger; Schädler, Marc René; Warzybok, Anna; Meyer, Bernd T; Brand, Thomas

2016-09-07

To characterize the individual patient's hearing impairment as obtained with the matrix sentence recognition test, a simulation Framework for Auditory Discrimination Experiments (FADE) is extended here using the Attenuation and Distortion (A+D) approach by Plomp as a blueprint for setting the individual processing parameters. FADE has been shown to predict the outcome of both speech recognition tests and psychoacoustic experiments based on simulations using an automatic speech recognition system requiring only few assumptions. It builds on the closed-set matrix sentence recognition test which is advantageous for testing individual speech recognition in a way comparable across languages. Individual predictions of speech recognition thresholds in stationary and in fluctuating noise were derived using the audiogram and an estimate of the internal level uncertainty for modeling the individual Plomp curves fitted to the data with the Attenuation (A-) and Distortion (D-) parameters of the Plomp approach. The "typical" audiogram shapes from Bisgaard et al with or without a "typical" level uncertainty and the individual data were used for individual predictions. As a result, the individualization of the level uncertainty was found to be more important than the exact shape of the individual audiogram to accurately model the outcome of the German Matrix test in stationary or fluctuating noise for listeners with hearing impairment. The prediction accuracy of the individualized approach also outperforms the (modified) Speech Intelligibility Index approach which is based on the individual threshold data only. © The Author(s) 2016.

Effects of cognitive impairment on prosodic parameters of speech production planning in multiple sclerosis.

PubMed

De Looze, Céline; Moreau, Noémie; Renié, Laurent; Kelly, Finnian; Ghio, Alain; Rico, Audrey; Audoin, Bertrand; Viallet, François; Pelletier, Jean; Petrone, Caterina

2017-05-24

Cognitive impairment (CI) affects 40-65% of patients with multiple sclerosis (MS). CI can have a negative impact on a patient's everyday activities, such as engaging in conversations. Speech production planning ability is crucial for successful verbal interactions and thus for preserving social and occupational skills. This study investigates the effect of cognitive-linguistic demand and CI on speech production planning in MS, as reflected in speech prosody. A secondary aim is to explore the clinical potential of prosodic features for the prediction of an individual's cognitive status in MS. A total of 45 subjects, that is 22 healthy controls (HC) and 23 patients in early stages of relapsing-remitting MS, underwent neuropsychological tests probing specific cognitive processes involved in speech production planning. All subjects also performed a read speech task, in which they had to read isolated sentences manipulated as for phonological length. Results show that the speech of MS patients with CI is mainly affected at the temporal level (articulation and speech rate, pause duration). Regression analyses further indicate that rate measures are correlated with working memory scores. In addition, linear discriminant analysis shows the ROC AUC of identifying MS patients with CI is 0.70 (95% confidence interval: 0.68-0.73). Our findings indicate that prosodic planning is deficient in patients with MS-CI and that the scope of planning depends on patients' cognitive abilities. We discuss how speech-based approaches could be used as an ecological method for the assessment and monitoring of CI in MS. © 2017 The British Psychological Society.

Real-time spectrum estimation–based dual-channel speech-enhancement algorithm for cochlear implant

PubMed Central

2012-01-01

Background Improvement of the cochlear implant (CI) front-end signal acquisition is needed to increase speech recognition in noisy environments. To suppress the directional noise, we introduce a speech-enhancement algorithm based on microphone array beamforming and spectral estimation. The experimental results indicate that this method is robust to directional mobile noise and strongly enhances the desired speech, thereby improving the performance of CI devices in a noisy environment. Methods The spectrum estimation and the array beamforming methods were combined to suppress the ambient noise. The directivity coefficient was estimated in the noise-only intervals, and was updated to fit for the mobile noise. Results The proposed algorithm was realized in the CI speech strategy. For actual parameters, we use Maxflat filter to obtain fractional sampling points and cepstrum method to differentiate the desired speech frame and the noise frame. The broadband adjustment coefficients were added to compensate the energy loss in the low frequency band. Discussions The approximation of the directivity coefficient is tested and the errors are discussed. We also analyze the algorithm constraint for noise estimation and distortion in CI processing. The performance of the proposed algorithm is analyzed and further be compared with other prevalent methods. Conclusions The hardware platform was constructed for the experiments. The speech-enhancement results showed that our algorithm can suppresses the non-stationary noise with high SNR. Excellent performance of the proposed algorithm was obtained in the speech enhancement experiments and mobile testing. And signal distortion results indicate that this algorithm is robust with high SNR improvement and low speech distortion. PMID:23006896

Deep learning

NASA Astrophysics Data System (ADS)

Lecun, Yann; Bengio, Yoshua; Hinton, Geoffrey

2015-05-01

Deep learning allows computational models that are composed of multiple processing layers to learn representations of data with multiple levels of abstraction. These methods have dramatically improved the state-of-the-art in speech recognition, visual object recognition, object detection and many other domains such as drug discovery and genomics. Deep learning discovers intricate structure in large data sets by using the backpropagation algorithm to indicate how a machine should change its internal parameters that are used to compute the representation in each layer from the representation in the previous layer. Deep convolutional nets have brought about breakthroughs in processing images, video, speech and audio, whereas recurrent nets have shone light on sequential data such as text and speech.

Deep learning.

PubMed

LeCun, Yann; Bengio, Yoshua; Hinton, Geoffrey

2015-05-28

Deep learning allows computational models that are composed of multiple processing layers to learn representations of data with multiple levels of abstraction. These methods have dramatically improved the state-of-the-art in speech recognition, visual object recognition, object detection and many other domains such as drug discovery and genomics. Deep learning discovers intricate structure in large data sets by using the backpropagation algorithm to indicate how a machine should change its internal parameters that are used to compute the representation in each layer from the representation in the previous layer. Deep convolutional nets have brought about breakthroughs in processing images, video, speech and audio, whereas recurrent nets have shone light on sequential data such as text and speech.

[Perception features of emotional intonation of short pseudowords].

PubMed

Dmitrieva, E S; Gel'man, V Ia; Zaĭtseva, K A; Orlov, A M

2012-01-01

Reaction time and recognition accuracy of speech emotional intonations in short meaningless words that differed only in one phoneme with background noise and without it were studied in 49 adults of 20-79 years old. The results were compared with the same parameters of emotional intonations in intelligent speech utterances under similar conditions. Perception of emotional intonations at different linguistic levels (phonological and lexico-semantic) was found to have both common features and certain peculiarities. Recognition characteristics of emotional intonations depending on gender and age of listeners appeared to be invariant with regard to linguistic levels of speech stimuli. Phonemic composition of pseudowords was found to influence the emotional perception, especially against the background noise. The most significant stimuli acoustic characteristic responsible for the perception of speech emotional prosody in short meaningless words under the two experimental conditions, i.e. with and without background noise, was the fundamental frequency variation.

Cleft audit protocol for speech (CAPS-A): a comprehensive training package for speech analysis.

PubMed

Sell, D; John, A; Harding-Bell, A; Sweeney, T; Hegarty, F; Freeman, J

2009-01-01

The previous literature has largely focused on speech analysis systems and ignored process issues, such as the nature of adequate speech samples, data acquisition, recording and playback. Although there has been recognition of the need for training on tools used in speech analysis associated with cleft palate, little attention has been paid to this issue. To design, execute, and evaluate a training programme for speech and language therapists on the systematic and reliable use of the Cleft Audit Protocol for Speech-Augmented (CAPS-A), addressing issues of standardized speech samples, data acquisition, recording, playback, and listening guidelines. Thirty-six specialist speech and language therapists undertook the training programme over four days. This consisted of two days' training on the CAPS-A tool followed by a third day, making independent ratings and transcriptions on ten new cases which had been previously recorded during routine audit data collection. This task was repeated on day 4, a minimum of one month later. Ratings were made using the CAPS-A record form with the CAPS-A definition table. An analysis was made of the speech and language therapists' CAPS-A ratings at occasion 1 and occasion 2 and the intra- and inter-rater reliability calculated. Trained therapists showed consistency in individual judgements on specific sections of the tool. Intraclass correlation coefficients were calculated for each section with good agreement on eight of 13 sections. There were only fair levels of agreement on anterior oral cleft speech characteristics, non-cleft errors/immaturities and voice. This was explained, at least in part, by their low prevalence which affects the calculation of the intraclass correlation coefficient statistic. Speech and language therapists benefited from training on the CAPS-A, focusing on specific aspects of speech using definitions of parameters and scalar points, in order to apply the tool systematically and reliably. Ratings are enhanced by ensuring a high degree of attention to the nature of the data, standardizing the speech sample, data acquisition, the listening process together with the use of high-quality recording and playback equipment. In addition, a method is proposed for maintaining listening skills following training as part of an individual's continuing education.

Speech outcome in unilateral complete cleft lip and palate patients: a descriptive study.

PubMed

Rullo, R; Di Maggio, D; Addabbo, F; Rullo, F; Festa, V M; Perillo, L

2014-09-01

In this study, resonance and articulation disorders were examined in a group of patients surgically treated for cleft lip and palate, considering family social background, and children's ability of self monitoring their speech output while speaking. Fifty children (32 males and 18 females) mean age 6.5 ± 1.6 years, affected by non-syndromic complete unilateral cleft of the lip and palate underwent the same surgical protocol. The speech level was evaluated using the Accordi's speech assessment protocol that focuses on intelligibility, nasality, nasal air escape, pharyngeal friction, and glottal stop. Pearson product-moment correlation analysis was used to detect significant associations between analysed parameters. A total of 16% (8 children) of the sample had severe to moderate degree of nasality and nasal air escape, presence of pharyngeal friction and glottal stop, which obviously compromise speech intelligibility. Ten children (10%) showed a barely acceptable phonological outcome: nasality and nasal air escape were mild to moderate, but the intelligibility remained poor. Thirty-two children (64%) had normal speech. Statistical analysis revealed a significant correlation between the severity of nasal resonance and nasal air escape (p ≤ 0.05). No statistical significant correlation was found between the final intelligibility and the patient social background, neither between the final intelligibility nor the age of the patients. The differences in speech outcome could be explained with a specific, subjective, and inborn ability, different for each child, in self-monitoring their speech output.

Vocal Parameters of Elderly Female Choir Singers

PubMed Central

Aquino, Fernanda Salvatico de; Ferreira, Léslie Piccolotto

2015-01-01

Introduction Due to increased life expectancy among the population, studying the vocal parameters of the elderly is key to promoting vocal health in old age. Objective This study aims to analyze the profile of the extension of speech of elderly female choristers, according to age group. Method The study counted on the participation of 25 elderly female choristers from the Choir of Messianic Church of São Paulo, with ages varying between 63 and 82 years, and an average of 71 years (standard deviation of 5.22). The elders were divided into two groups: G1 aged 63 to 71 years and G2 aged 72 to 82. We asked that each participant count from 20 to 30 in weak, medium, strong, and very strong intensities. Their speech was registered by the software Vocalgrama that allows the evaluation of the profile of speech range. We then submitted the parameters of frequency and intensity to descriptive analysis, both in minimum and maximum levels, and range of spoken voice. Results The average of minimum and maximum frequencies were respectively 134.82–349.96 Hz for G1 and 137.28–348.59 Hz for G2; the average for minimum and maximum intensities were respectively 40.28–95.50 dB for G1 and 40.63–94.35 dB for G2; the vocal range used in speech was 215.14 Hz for G1 and 211.30 Hz for G2. Conclusion The minimum and maximum frequencies, maximum intensity, and vocal range presented differences in favor of the younger elder group. PMID:26722341

Influence of signal processing strategy in auditory abilities.

PubMed

Melo, Tatiana Mendes de; Bevilacqua, Maria Cecília; Costa, Orozimbo Alves; Moret, Adriane Lima Mortari

2013-01-01

The signal processing strategy is a parameter that may influence the auditory performance of cochlear implant and is important to optimize this parameter to provide better speech perception, especially in difficult listening situations. To evaluate the individual's auditory performance using two different signal processing strategy. Prospective study with 11 prelingually deafened children with open-set speech recognition. A within-subjects design was used to compare performance with standard HiRes and HiRes 120 in three different moments. During test sessions, subject's performance was evaluated by warble-tone sound-field thresholds, speech perception evaluation, in quiet and in noise. In the silence, children S1, S4, S5, S7 showed better performance with the HiRes 120 strategy and children S2, S9, S11 showed better performance with the HiRes strategy. In the noise was also observed that some children performed better using the HiRes 120 strategy and other with HiRes. Not all children presented the same pattern of response to the different strategies used in this study, which reinforces the need to look at optimizing cochlear implant clinical programming.

JPEG XS, a new standard for visually lossless low-latency lightweight image compression

NASA Astrophysics Data System (ADS)

Descampe, Antonin; Keinert, Joachim; Richter, Thomas; Fößel, Siegfried; Rouvroy, Gaël.

2017-09-01

JPEG XS is an upcoming standard from the JPEG Committee (formally known as ISO/IEC SC29 WG1). It aims to provide an interoperable visually lossless low-latency lightweight codec for a wide range of applications including mezzanine compression in broadcast and Pro-AV markets. This requires optimal support of a wide range of implementation technologies such as FPGAs, CPUs and GPUs. Targeted use cases are professional video links, IP transport, Ethernet transport, real-time video storage, video memory buffers, and omnidirectional video capture and rendering. In addition to the evaluation of the visual transparency of the selected technologies, a detailed analysis of the hardware and software complexity as well as the latency has been done to make sure that the new codec meets the requirements of the above-mentioned use cases. In particular, the end-to-end latency has been constrained to a maximum of 32 lines. Concerning the hardware complexity, neither encoder nor decoder should require more than 50% of an FPGA similar to Xilinx Artix 7 or 25% of an FPGA similar to Altera Cyclon 5. This process resulted in a coding scheme made of an optional color transform, a wavelet transform, the entropy coding of the highest magnitude level of groups of coefficients, and the raw inclusion of the truncated wavelet coefficients. This paper presents the details and status of the standardization process, a technical description of the future standard, and the latest performance evaluation results.

Parallel efficient rate control methods for JPEG 2000

NASA Astrophysics Data System (ADS)

Martínez-del-Amor, Miguel Á.; Bruns, Volker; Sparenberg, Heiko

2017-09-01

Since the introduction of JPEG 2000, several rate control methods have been proposed. Among them, post-compression rate-distortion optimization (PCRD-Opt) is the most widely used, and the one recommended by the standard. The approach followed by this method is to first compress the entire image split in code blocks, and subsequently, optimally truncate the set of generated bit streams according to the maximum target bit rate constraint. The literature proposes various strategies on how to estimate ahead of time where a block will get truncated in order to stop the execution prematurely and save time. However, none of them have been defined bearing in mind a parallel implementation. Today, multi-core and many-core architectures are becoming popular for JPEG 2000 codecs implementations. Therefore, in this paper, we analyze how some techniques for efficient rate control can be deployed in GPUs. In order to do that, the design of our GPU-based codec is extended, allowing stopping the process at a given point. This extension also harnesses a higher level of parallelism on the GPU, leading to up to 40% of speedup with 4K test material on a Titan X. In a second step, three selected rate control methods are adapted and implemented in our parallel encoder. A comparison is then carried out, and used to select the best candidate to be deployed in a GPU encoder, which gave an extra 40% of speedup in those situations where it was really employed.

Robust video transmission with distributed source coded auxiliary channel.

PubMed

Wang, Jiajun; Majumdar, Abhik; Ramchandran, Kannan

2009-12-01

We propose a novel solution to the problem of robust, low-latency video transmission over lossy channels. Predictive video codecs, such as MPEG and H.26x, are very susceptible to prediction mismatch between encoder and decoder or "drift" when there are packet losses. These mismatches lead to a significant degradation in the decoded quality. To address this problem, we propose an auxiliary codec system that sends additional information alongside an MPEG or H.26x compressed video stream to correct for errors in decoded frames and mitigate drift. The proposed system is based on the principles of distributed source coding and uses the (possibly erroneous) MPEG/H.26x decoder reconstruction as side information at the auxiliary decoder. The distributed source coding framework depends upon knowing the statistical dependency (or correlation) between the source and the side information. We propose a recursive algorithm to analytically track the correlation between the original source frame and the erroneous MPEG/H.26x decoded frame. Finally, we propose a rate-distortion optimization scheme to allocate the rate used by the auxiliary encoder among the encoding blocks within a video frame. We implement the proposed system and present extensive simulation results that demonstrate significant gains in performance both visually and objectively (on the order of 2 dB in PSNR over forward error correction based solutions and 1.5 dB in PSNR over intrarefresh based solutions for typical scenarios) under tight latency constraints.

Acoustic Analysis of PD Speech

PubMed Central

Chenausky, Karen; MacAuslan, Joel; Goldhor, Richard

2011-01-01

According to the U.S. National Institutes of Health, approximately 500,000 Americans have Parkinson's disease (PD), with roughly another 50,000 receiving new diagnoses each year. 70%–90% of these people also have the hypokinetic dysarthria associated with PD. Deep brain stimulation (DBS) substantially relieves motor symptoms in advanced-stage patients for whom medication produces disabling dyskinesias. This study investigated speech changes as a result of DBS settings chosen to maximize motor performance. The speech of 10 PD patients and 12 normal controls was analyzed for syllable rate and variability, syllable length patterning, vowel fraction, voice-onset time variability, and spirantization. These were normalized by the controls' standard deviation to represent distance from normal and combined into a composite measure. Results show that DBS settings relieving motor symptoms can improve speech, making it up to three standard deviations closer to normal. However, the clinically motivated settings evaluated here show greater capacity to impair, rather than improve, speech. A feedback device developed from these findings could be useful to clinicians adjusting DBS parameters, as a means for ensuring they do not unwittingly choose DBS settings which impair patients' communication. PMID:21977333

Prediction of consonant recognition in quiet for listeners with normal and impaired hearing using an auditory model.

PubMed

Jürgens, Tim; Ewert, Stephan D; Kollmeier, Birger; Brand, Thomas

2014-03-01

Consonant recognition was assessed in normal-hearing (NH) and hearing-impaired (HI) listeners in quiet as a function of speech level using a nonsense logatome test. Average recognition scores were analyzed and compared to recognition scores of a speech recognition model. In contrast to commonly used spectral speech recognition models operating on long-term spectra, a "microscopic" model operating in the time domain was used. Variations of the model (accounting for hearing impairment) and different model parameters (reflecting cochlear compression) were tested. Using these model variations this study examined whether speech recognition performance in quiet is affected by changes in cochlear compression, namely, a linearization, which is often observed in HI listeners. Consonant recognition scores for HI listeners were poorer than for NH listeners. The model accurately predicted the speech reception thresholds of the NH and most HI listeners. A partial linearization of the cochlear compression in the auditory model, while keeping audibility constant, produced higher recognition scores and improved the prediction accuracy. However, including listener-specific information about the exact form of the cochlear compression did not improve the prediction further.

Sound System Engineering & Optimization: The effects of multiple arrivals on the intelligibility of reinforced speech

NASA Astrophysics Data System (ADS)

Ryan, Timothy James

The effects of multiple arrivals on the intelligibility of speech produced by live-sound reinforcement systems are examined. The intent is to determine if correlations exist between the manipulation of sound system optimization parameters and the subjective attribute speech intelligibility. Given the number, and wide range, of variables involved, this exploratory research project attempts to narrow the focus of further studies. Investigated variables are delay time between signals arriving from multiple elements of a loudspeaker array, array type and geometry and the two-way interactions of speech-to-noise ratio and array geometry with delay time. Intelligibility scores were obtained through subjective evaluation of binaural recordings, reproduced via headphone, using the Modified Rhyme Test. These word-score results are compared with objective measurements of Speech Transmission Index (STI). Results indicate that both variables, delay time and array geometry, have significant effects on intelligibility. Additionally, it is seen that all three of the possible two-way interactions have significant effects. Results further reveal that the STI measurement method overestimates the decrease in intelligibility due to short delay times between multiple arrivals.

Anxiolytic-like effect of oxytocin in the simulated public speaking test.

PubMed

de Oliveira, Danielle C G; Zuardi, Antonio W; Graeff, Frederico G; Queiroz, Regina H C; Crippa, José A S

2012-04-01

Oxytocin (OT) is known to be involved in anxiety, as well as cardiovascular and hormonal regulation. The objective of this study was to assess the acute effect of intranasally administered OT on subjective states, as well as cardiovascular and endocrine parameters, in healthy volunteers (n = 14) performing a simulated public speaking test. OT or placebo was administered intranasally 50 min before the test. Assessments were made across time during the experimental session: (1) baseline (-30 min); (2) pre-test (-15 min); (3) anticipation of the speech (50 min); (4) during the speech (1:03 h), post-test time 1 (1:26 h), and post-test time 2 (1:46 h). Subjective states were evaluated by self-assessment scales. Cortisol serum and plasma adrenocorticotropic hormone (ACTH) were measured. Additionally, heart rate, blood pressure, skin conductance, and the number of spontaneous fluctuations in skin conductance were measured. Compared with placebo, OT reduced the Visual Analogue Mood Scale (VAMS) anxiety index during the pre-test phase only, while increasing sedation at the pre-test, anticipation, and speech phases. OT also lowered the skin conductance level at the pre-test, anticipation, speech, and post-test 2 phases. Other parameters evaluated were not significantly affected by OT. The present results show that OT reduces anticipatory anxiety, but does not affect public speaking fear, suggesting that this hormone has anxiolytic properties.

Preliminary study of acoustic analysis for evaluating speech-aid oral prostheses: Characteristic dips in octave spectrum for comparison of nasality.

PubMed

Chang, Yen-Liang; Hung, Chao-Ho; Chen, Po-Yueh; Chen, Wei-Chang; Hung, Shih-Han

2015-10-01

Acoustic analysis is often used in speech evaluation but seldom for the evaluation of oral prostheses designed for reconstruction of surgical defect. This study aimed to introduce the application of acoustic analysis for patients with velopharyngeal insufficiency (VPI) due to oral surgery and rehabilitated with oral speech-aid prostheses. The pre- and postprosthetic rehabilitation acoustic features of sustained vowel sounds from two patients with VPI were analyzed and compared with the acoustic analysis software Praat. There were significant differences in the octave spectrum of sustained vowel speech sound between the pre- and postprosthetic rehabilitation. Acoustic measurements of sustained vowels for patients before and after prosthetic treatment showed no significant differences for all parameters of fundamental frequency, jitter, shimmer, noise-to-harmonics ratio, formant frequency, F1 bandwidth, and band energy difference. The decrease in objective nasality perceptions correlated very well with the decrease in dips of the spectra for the male patient with a higher speech bulb height. Acoustic analysis may be a potential technique for evaluating the functions of oral speech-aid prostheses, which eliminates dysfunctions due to the surgical defect and contributes to a high percentage of intelligible speech. Octave spectrum analysis may also be a valuable tool for detecting changes in nasality characteristics of the voice during prosthetic treatment of VPI. Copyright © 2014. Published by Elsevier B.V.

Voice parameters and videonasolaryngoscopy in children with vocal nodules: a longitudinal study, before and after voice therapy.

PubMed

Valadez, Victor; Ysunza, Antonio; Ocharan-Hernandez, Esther; Garrido-Bustamante, Norma; Sanchez-Valerio, Araceli; Pamplona, Ma C

2012-09-01

Vocal Nodules (VN) are a functional voice disorder associated with voice misuse and abuse in children. There are few reports addressing vocal parameters in children with VN, especially after a period of vocal rehabilitation. The purpose of this study is to describe measurements of vocal parameters including Fundamental Frequency (FF), Shimmer (S), and Jitter (J), videonasolaryngoscopy examination and clinical perceptual assessment, before and after voice therapy in children with VN. Voice therapy was provided using visual support through Speech-Viewer software. Twenty patients with VN were studied. An acoustical analysis of voice was performed and compared with data from subjects from a control group matched by age and gender. Also, clinical perceptual assessment of voice and videonasolaryngoscopy were performed to all patients with VN. After a period of voice therapy, provided with visual support using Speech Viewer-III (SV-III-IBM) software, new acoustical analyses, perceptual assessments and videonasolaryngoscopies were performed. Before the onset of voice therapy, there was a significant difference (p<0.05) in mean FF, S and J, between the patients with VN and subjects from the control group. After the voice therapy period, a significant improvement (p<0.05) was found in all acoustic voice parameters. Moreover, perceptual voice analysis demonstrated improvement in all cases. Finally, videonasolaryngoscopy demonstrated that vocal nodules were no longer discernible on the vocal folds in any of the cases. SV-III software seems to be a safe and reliable method for providing voice therapy in children with VN. Acoustic voice parameters, perceptual data and videonasolaryngoscopy were significantly improved after the speech therapy period was completed. Copyright © 2012 Elsevier Ireland Ltd. All rights reserved.

Speech coding

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

Ravishankar, C., Hughes Network Systems, Germantown, MD

Speech is the predominant means of communication between human beings and since the invention of the telephone by Alexander Graham Bell in 1876, speech services have remained to be the core service in almost all telecommunication systems. Original analog methods of telephony had the disadvantage of speech signal getting corrupted by noise, cross-talk and distortion Long haul transmissions which use repeaters to compensate for the loss in signal strength on transmission links also increase the associated noise and distortion. On the other hand digital transmission is relatively immune to noise, cross-talk and distortion primarily because of the capability to faithfullymore » regenerate digital signal at each repeater purely based on a binary decision. Hence end-to-end performance of the digital link essentially becomes independent of the length and operating frequency bands of the link Hence from a transmission point of view digital transmission has been the preferred approach due to its higher immunity to noise. The need to carry digital speech became extremely important from a service provision point of view as well. Modem requirements have introduced the need for robust, flexible and secure services that can carry a multitude of signal types (such as voice, data and video) without a fundamental change in infrastructure. Such a requirement could not have been easily met without the advent of digital transmission systems, thereby requiring speech to be coded digitally. The term Speech Coding is often referred to techniques that represent or code speech signals either directly as a waveform or as a set of parameters by analyzing the speech signal. In either case, the codes are transmitted to the distant end where speech is reconstructed or synthesized using the received set of codes. A more generic term that is applicable to these techniques that is often interchangeably used with speech coding is the term voice coding. This term is more generic in the sense that the coding techniques are equally applicable to any voice signal whether or not it carries any intelligible information, as the term speech implies. Other terms that are commonly used are speech compression and voice compression since the fundamental idea behind speech coding is to reduce (compress) the transmission rate (or equivalently the bandwidth) And/or reduce storage requirements In this document the terms speech and voice shall be used interchangeably.« less

Walking while talking: Young adults flexibly allocate resources between speech and gait.

PubMed

Raffegeau, Tiphanie E; Haddad, Jeffrey M; Huber, Jessica E; Rietdyk, Shirley

2018-05-26

Walking while talking is an ideal multitask behavior to assess how young healthy adults manage concurrent tasks as it is well-practiced, cognitively demanding, and has real consequences for impaired performance in either task. Since the association between cognitive tasks and gait appears stronger when the gait task is more challenging, gait challenge was systematically manipulated in this study. To understand how young adults accomplish the multitask behavior of walking while talking as the gait challenge was systematically manipulated. Sixteen young adults (21 ± 1.6 years, 9 males) performed three gait tasks with and without speech: unobstructed gait (easy), obstacle crossing (moderate), obstacle crossing and tray carrying (difficult). Participants also provided a speech sample while seated for a baseline indicator of speech. The speech task was to speak extemporaneously about a topic (e.g. first car). Gait speed and the duration of silent pauses during speaking were determined. Silent pauses reflect cognitive processes involved in speech production and language planning. When speaking and walking without obstacles, gait speed decreased (relative to walking without speaking) but silent pause duration did not change (relative to seated speech). These changes are consistent with the idea that, in the easy gait task, participants placed greater value on speech pauses than on gait speed, likely due to the negative social consequences of impaired speech. In the moderate and difficult gait tasks both parameters changed: gait speed decreased and silent pauses increased. Walking while talking is a cognitively demanding task for healthy young adults, despite being a well-practiced habitual activity. These findings are consistent with the integrated model of task prioritization from Yogev-Seligmann et al., [1]. Copyright © 2018 Elsevier B.V. All rights reserved.

[Acoustic conditions in open plan office - Application of technical measures in a typical room].

PubMed

Mikulski, Witold

2018-03-09

Noise in open plan offices should not exceed acceptable levels for the hearing protection. Its major negative effects on employees are nuisance and impediment in execution of work. Specific technical solutions should be introduced to provide proper acoustic conditions for work performance. Acoustic evaluation of a typical open plan office was presented in the article published in "Medycyna Pracy" 5/2016. None of the rooms meets all the criteria, therefore, in this article one of the rooms was chosen to apply different technical solutions to check the possibility of reaching proper acoustic conditions. Acoustic effectiveness of those solutions was verified by means of digital simulation. The model was checked by comparing the results of measurements and calculations before using simulation. The analyzis revealed that open plan offices supplemented with signals for masking speech signals can meet all the required criteria. It is relatively easy to reach proper reverberation time (i.e., sound absorption). It is more difficult to reach proper values of evaluation parameters determined from A-weighted sound pressure level (SPLA) of speech. The most difficult is to provide proper values of evaluation parameters determined from speech transmission index (STI). Finally, it is necessary (besides acoustic treatment) to use devices for speech masking. The study proved that it is technically possible to reach proper acoustic condition. Main causes of employees complaints in open plan office are inadequate acoustic work conditions. Therefore, it is necessary to apply specific technical solutions - not only sound absorbing suspended ceiling and high acoustic barriers, but also devices for speech masking. Med Pr 2018;69(2):153-165. This work is available in Open Access model and licensed under a CC BY-NC 3.0 PL license.

Some Behavioral and Neurobiological Constraints on Theories of Audiovisual Speech Integration: A Review and Suggestions for New Directions

PubMed Central

Altieri, Nicholas; Pisoni, David B.; Townsend, James T.

2012-01-01

Summerfield (1987) proposed several accounts of audiovisual speech perception, a field of research that has burgeoned in recent years. The proposed accounts included the integration of discrete phonetic features, vectors describing the values of independent acoustical and optical parameters, the filter function of the vocal tract, and articulatory dynamics of the vocal tract. The latter two accounts assume that the representations of audiovisual speech perception are based on abstract gestures, while the former two assume that the representations consist of symbolic or featural information obtained from visual and auditory modalities. Recent converging evidence from several different disciplines reveals that the general framework of Summerfield’s feature-based theories should be expanded. An updated framework building upon the feature-based theories is presented. We propose a processing model arguing that auditory and visual brain circuits provide facilitatory information when the inputs are correctly timed, and that auditory and visual speech representations do not necessarily undergo translation into a common code during information processing. Future research on multisensory processing in speech perception should investigate the connections between auditory and visual brain regions, and utilize dynamic modeling tools to further understand the timing and information processing mechanisms involved in audiovisual speech integration. PMID:21968081

Some behavioral and neurobiological constraints on theories of audiovisual speech integration: a review and suggestions for new directions.

PubMed

Altieri, Nicholas; Pisoni, David B; Townsend, James T

2011-01-01

Summerfield (1987) proposed several accounts of audiovisual speech perception, a field of research that has burgeoned in recent years. The proposed accounts included the integration of discrete phonetic features, vectors describing the values of independent acoustical and optical parameters, the filter function of the vocal tract, and articulatory dynamics of the vocal tract. The latter two accounts assume that the representations of audiovisual speech perception are based on abstract gestures, while the former two assume that the representations consist of symbolic or featural information obtained from visual and auditory modalities. Recent converging evidence from several different disciplines reveals that the general framework of Summerfield's feature-based theories should be expanded. An updated framework building upon the feature-based theories is presented. We propose a processing model arguing that auditory and visual brain circuits provide facilitatory information when the inputs are correctly timed, and that auditory and visual speech representations do not necessarily undergo translation into a common code during information processing. Future research on multisensory processing in speech perception should investigate the connections between auditory and visual brain regions, and utilize dynamic modeling tools to further understand the timing and information processing mechanisms involved in audiovisual speech integration.

Is automatic speech-to-text transcription ready for use in psychological experiments?

PubMed

Ziman, Kirsten; Heusser, Andrew C; Fitzpatrick, Paxton C; Field, Campbell E; Manning, Jeremy R

2018-04-23

Verbal responses are a convenient and naturalistic way for participants to provide data in psychological experiments (Salzinger, The Journal of General Psychology, 61(1),65-94:1959). However, audio recordings of verbal responses typically require additional processing, such as transcribing the recordings into text, as compared with other behavioral response modalities (e.g., typed responses, button presses, etc.). Further, the transcription process is often tedious and time-intensive, requiring human listeners to manually examine each moment of recorded speech. Here we evaluate the performance of a state-of-the-art speech recognition algorithm (Halpern et al., 2016) in transcribing audio data into text during a list-learning experiment. We compare transcripts made by human annotators to the computer-generated transcripts. Both sets of transcripts matched to a high degree and exhibited similar statistical properties, in terms of the participants' recall performance and recall dynamics that the transcripts captured. This proof-of-concept study suggests that speech-to-text engines could provide a cheap, reliable, and rapid means of automatically transcribing speech data in psychological experiments. Further, our findings open the door for verbal response experiments that scale to thousands of participants (e.g., administered online), as well as a new generation of experiments that decode speech on the fly and adapt experimental parameters based on participants' prior responses.

An EMA analysis of the effect of increasing word length on consonant production in apraxia of speech: a case study.

PubMed

Bartle, Carly J; Goozée, Justine V; Murdoch, Bruce E

2007-03-01

The effect of increasing word length on the articulatory dynamics (i.e. duration, distance, maximum acceleration, maximum deceleration, and maximum velocity) of consonant production in acquired apraxia of speech was investigated using electromagnetic articulography (EMA). Tongue-tip and tongue-back movement of one apraxic patient was recorded using the AG-200 EMA system during word-initial consonant productions in one, two, and three syllable words. Significantly deviant articulatory parameters were recorded for each of the target consonants during one, two, and three syllables words. Word length effects were most evident during the release phase of target consonant productions. The results are discussed with respect to theories of speech motor control as they relate to AOS.

Fostering a Culture of Engagement. (Military Review, September-October 2009)

DTIC Science & Technology

2009-10-01

the publication of any information that could even remotely be considered to aid the enemy.”8 A year later, the Sedition Act made criticism of the...Kenneth Payne, “The Media as an Instrument of War,” Parameters (Spring 2005), 81-93. 31. Transcript of speech given by GEN Martin E. Dempsey, Commanding...General TRADOC, to the U.S. Army War College, Carlisle Barracks, PA, 25 March 2009. <www.tradoc.army.mil/pao/ Speeches /Gen%20Dempsey%202008-09/AWC%20

Improved Objective Measurements for Speech Quality Testing

DTIC Science & Technology

1985-01-01

criterion with the possible exception of the channel vocoder, for which the spread in subjective responses was slightly less than des ired. 4 10 -a_- 14...n 5 19-.0 H ,e -- * ,- p--. -. a’ - - -:== .. LJ. . U .... :1±, .L±--± i- -.. ni m.,in h U ’ el l . ORII STCO 1RE PROIOP)LES p =.. •,,• mI.. 1 .S OI...parameter III specifies thce threshold between objectively interrupted and non-interrupted speech. In the foi- aula apecifying RATIO, mf is the index of the

A preliminary report of music-based training for adult cochlear implant users: Rationales and development.

PubMed

Gfeller, Kate; Guthe, Emily; Driscoll, Virginia; Brown, Carolyn J

2015-09-01

This paper provides a preliminary report of a music-based training program for adult cochlear implant (CI) recipients. Included in this report are descriptions of the rationale for music-based training, factors influencing program development, and the resulting program components. Prior studies describing experience-based plasticity in response to music training, auditory training for persons with hearing impairment, and music training for CI recipients were reviewed. These sources revealed rationales for using music to enhance speech, factors associated with successful auditory training, relevant aspects of electric hearing and music perception, and extant evidence regarding limitations and advantages associated with parameters for music training with CI users. This informed the development of a computer-based music training program designed specifically for adult CI users. Principles and parameters for perceptual training of music, such as stimulus choice, rehabilitation approach, and motivational concerns were developed in relation to the unique auditory characteristics of adults with electric hearing. An outline of the resulting program components and the outcome measures for evaluating program effectiveness are presented. Music training can enhance the perceptual accuracy of music, but is also hypothesized to enhance several features of speech with similar processing requirements as music (e.g., pitch and timbre). However, additional evaluation of specific training parameters and the impact of music-based training on speech perception of CI users is required.

A preliminary report of music-based training for adult cochlear implant users: rationales and development

PubMed Central

Gfeller, Kate; Guthe, Emily; Driscoll, Virginia; Brown, Carolyn J.

2015-01-01

Objective This paper provides a preliminary report of a music-based training program for adult cochlear implant (CI) recipients. Included in this report are descriptions of the rationale for music-based training, factors influencing program development, and the resulting program components. Methods Prior studies describing experience-based plasticity in response to music training, auditory training for persons with hearing impairment, and music training for cochlear implant recipients were reviewed. These sources revealed rationales for using music to enhance speech, factors associated with successful auditory training, relevant aspects of electric hearing and music perception, and extant evidence regarding limitations and advantages associated with parameters for music training with CI users. This information formed the development of a computer-based music training program designed specifically for adult CI users. Results Principles and parameters for perceptual training of music, such as stimulus choice, rehabilitation approach, and motivational concerns were developed in relation to the unique auditory characteristics of adults with electric hearing. An outline of the resulting program components and the outcome measures for evaluating program effectiveness are presented. Conclusions Music training can enhance the perceptual accuracy of music, but is also hypothesized to enhance several features of speech with similar processing requirements as music (e.g., pitch and timbre). However, additional evaluation of specific training parameters and the impact of music-based training on speech perception of CI users are required. PMID:26561884

Systematic review of compound action potentials as predictors for cochlear implant performance.

PubMed

van Eijl, Ruben H M; Buitenhuis, Patrick J; Stegeman, Inge; Klis, Sjaak F L; Grolman, Wilko

2017-02-01

The variability in speech perception between cochlear implant users is thought to result from the degeneration of the auditory nerve. Degeneration of the auditory nerve, histologically assessed, correlates with electrophysiologically acquired measures, such as electrically evoked compound action potentials (eCAPs) in experimental animals. To predict degeneration of the auditory nerve in humans, where histology is impossible, this paper reviews the correlation between speech perception and eCAP recordings in cochlear implant patients. PubMed and Embase. We performed a systematic search for articles containing the following major themes: cochlear implants, evoked potentials, and speech perception. Two investigators independently conducted title-abstract screening, full-text screening, and critical appraisal. Data were extracted from the remaining articles. Twenty-five of 1,429 identified articles described a correlation between speech perception and eCAP attributes. Due to study heterogeneity, a meta-analysis was not feasible, and studies were descriptively analyzed. Several studies investigating presence of the eCAP, recovery time constant, slope of the amplitude growth function, and spatial selectivity showed significant correlations with speech perception. In contrast, neural adaptation, eCAP threshold, and change with varying interphase gap did not significantly correlate with speech perception in any of the identified studies. Significant correlations between speech perception and parameters obtained through eCAP recordings have been documented in literature; however, reporting was ambiguous. There is insufficient evidence for eCAPs as a predictive factor for speech perception. More research is needed to further investigate this relation. Laryngoscope, 2016 127:476-487, 2017. © 2016 The American Laryngological, Rhinological and Otological Society, Inc.

Speech perception in noise with a harmonic complex excited vocoder.

PubMed

Churchill, Tyler H; Kan, Alan; Goupell, Matthew J; Ihlefeld, Antje; Litovsky, Ruth Y

2014-04-01

A cochlear implant (CI) presents band-pass-filtered acoustic envelope information by modulating current pulse train levels. Similarly, a vocoder presents envelope information by modulating an acoustic carrier. By studying how normal hearing (NH) listeners are able to understand degraded speech signals with a vocoder, the parameters that best simulate electric hearing and factors that might contribute to the NH-CI performance difference may be better understood. A vocoder with harmonic complex carriers (fundamental frequency, f0 = 100 Hz) was used to study the effect of carrier phase dispersion on speech envelopes and intelligibility. The starting phases of the harmonic components were randomly dispersed to varying degrees prior to carrier filtering and modulation. NH listeners were tested on recognition of a closed set of vocoded words in background noise. Two sets of synthesis filters simulated different amounts of current spread in CIs. Results showed that the speech vocoded with carriers whose starting phases were maximally dispersed was the most intelligible. Superior speech understanding may have been a result of the flattening of the dispersed-phase carrier's intrinsic temporal envelopes produced by the large number of interacting components in the high-frequency channels. Cross-correlogram analyses of auditory nerve model simulations confirmed that randomly dispersing the carrier's component starting phases resulted in better neural envelope representation. However, neural metrics extracted from these analyses were not found to accurately predict speech recognition scores for all vocoded speech conditions. It is possible that central speech understanding mechanisms are insensitive to the envelope-fine structure dichotomy exploited by vocoders.

Effects of Online Augmented Kinematic and Perceptual Feedback on Treatment of Speech Movements in Apraxia of Speech

PubMed Central

McNeil, M.R.; Katz, W.F.; Fossett, T.R.D.; Garst, D.M.; Szuminsky, N.J.; Carter, G.; Lim, K.Y.

2010-01-01

Apraxia of speech (AOS) is a motor speech disorder characterized by disturbed spatial and temporal parameters of movement. Research on motor learning suggests that augmented feedback may provide a beneficial effect for training movement. This study examined the effects of the presence and frequency of online augmented visual kinematic feedback (AVKF) and clinician-provided perceptual feedback on speech accuracy in 2 adults with acquired AOS. Within a single-subject multiple-baseline design, AVKF was provided using electromagnetic midsagittal articulography (EMA) in 2 feedback conditions (50 or 100%). Articulator placement was specified for speech motor targets (SMTs). Treated and baselined SMTs were in the initial or final position of single-syllable words, in varying consonant-vowel or vowel-consonant contexts. SMTs were selected based on each participant's pre-assessed erred productions. Productions were digitally recorded and online perceptual judgments of accuracy (including segment and intersegment distortions) were made. Inter- and intra-judge reliability for perceptual accuracy was high. Results measured by visual inspection and effect size revealed positive acquisition and generalization effects for both participants. Generalization occurred across vowel contexts and to untreated probes. Results of the frequency manipulation were confounded by presentation order. Maintenance of learned and generalized effects were demonstrated for 1 participant. These data provide support for the role of augmented feedback in treating speech movements that result in perceptually accurate speech production. Future investigations will explore the independent contributions of each feedback type (i.e. kinematic and perceptual) in producing efficient and effective training of SMTs in persons with AOS. PMID:20424468

A New Model of Sensorimotor Coupling in the Development of Speech

ERIC Educational Resources Information Center

Westermann, Gert; Miranda, Eduardo Reck

2004-01-01

We present a computational model that learns a coupling between motor parameters and their sensory consequences in vocal production during a babbling phase. Based on the coupling, preferred motor parameters and prototypically perceived sounds develop concurrently. Exposure to an ambient language modifies perception to coincide with the sounds from…

Stochastic Packet Loss Model to Evaluate QoE Impairments

NASA Astrophysics Data System (ADS)

Hohlfeld, Oliver

With provisioning of broadband access for mass market—even in wireless and mobile networks—multimedia content, especially real-time streaming of high-quality audio and video, is extensively viewed and exchanged over the Internet. Quality of Experience (QoE) aspects, describing the service quality perceived by the user, is a vital factor in ensuring customer satisfaction in today's communication networks. Frameworks for accessing quality degradations in streamed video currently are investigated as a complex multi-layered research topic, involving network traffic load, codec functions and measures of user perception of video quality.

Development of a Federal Standard for a Video Codec Operating at P X 64 Kbps

DTIC Science & Technology

1991-02-01

defined. Which blocks will be encoded, with what type of code, and with what accuracy is under control of the designer . While there is less freedom for the...decoder, post processing, such as filtering or interpolation, can be used to improve the appearance of the image and is under control of the designer ...the only one, or, in the case that more video sources are to be distinguished, it is that designated "video #1". VWA2 Equivalent to VIA, but designating

Articulatory speech synthesis and speech production modelling

NASA Astrophysics Data System (ADS)

Huang, Jun

This dissertation addresses the problem of speech synthesis and speech production modelling based on the fundamental principles of human speech production. Unlike the conventional source-filter model, which assumes the independence of the excitation and the acoustic filter, we treat the entire vocal apparatus as one system consisting of a fluid dynamic aspect and a mechanical part. We model the vocal tract by a three-dimensional moving geometry. We also model the sound propagation inside the vocal apparatus as a three-dimensional nonplane-wave propagation inside a viscous fluid described by Navier-Stokes equations. In our work, we first propose a combined minimum energy and minimum jerk criterion to estimate the dynamic vocal tract movements during speech production. Both theoretical error bound analysis and experimental results show that this method can achieve very close match at the target points and avoid the abrupt change in articulatory trajectory at the same time. Second, a mechanical vocal fold model is used to compute the excitation signal of the vocal tract. The advantage of this model is that it is closely coupled with the vocal tract system based on fundamental aerodynamics. As a result, we can obtain an excitation signal with much more detail than the conventional parametric vocal fold excitation model. Furthermore, strong evidence of source-tract interaction is observed. Finally, we propose a computational model of the fricative and stop types of sounds based on the physical principles of speech production. The advantage of this model is that it uses an exogenous process to model the additional nonsteady and nonlinear effects due to the flow mode, which are ignored by the conventional source- filter speech production model. A recursive algorithm is used to estimate the model parameters. Experimental results show that this model is able to synthesize good quality fricative and stop types of sounds. Based on our dissertation work, we carefully argue that the articulatory speech production model has the potential to flexibly synthesize natural-quality speech sounds and to provide a compact computational model for speech production that can be beneficial to a wide range of areas in speech signal processing.

Towards parameter-free classification of sound effects in movies

NASA Astrophysics Data System (ADS)

Chu, Selina; Narayanan, Shrikanth; Kuo, C.-C. J.

2005-08-01

The problem of identifying intense events via multimedia data mining in films is investigated in this work. Movies are mainly characterized by dialog, music, and sound effects. We begin our investigation with detecting interesting events through sound effects. Sound effects are neither speech nor music, but are closely associated with interesting events such as car chases and gun shots. In this work, we utilize low-level audio features including MFCC and energy to identify sound effects. It was shown in previous work that the Hidden Markov model (HMM) works well for speech/audio signals. However, this technique requires a careful choice in designing the model and choosing correct parameters. In this work, we introduce a framework that will avoid such necessity and works well with semi- and non-parametric learning algorithms.

Perceptual centres in speech - an acoustic analysis

NASA Astrophysics Data System (ADS)

Scott, Sophie Kerttu

Perceptual centres, or P-centres, represent the perceptual moments of occurrence of acoustic signals - the 'beat' of a sound. P-centres underlie the perception and production of rhythm in perceptually regular speech sequences. P-centres have been modelled both in speech and non speech (music) domains. The three aims of this thesis were toatest out current P-centre models to determine which best accounted for the experimental data bto identify a candidate parameter to map P-centres onto (a local approach) as opposed to the previous global models which rely upon the whole signal to determine the P-centre the final aim was to develop a model of P-centre location which could be applied to speech and non speech signals. The first aim was investigated by a series of experiments in which a) speech from different speakers was investigated to determine whether different models could account for variation between speakers b) whether rendering the amplitude time plot of a speech signal affects the P-centre of the signal c) whether increasing the amplitude at the offset of a speech signal alters P-centres in the production and perception of speech. The second aim was carried out by a) manipulating the rise time of different speech signals to determine whether the P-centre was affected, and whether the type of speech sound ramped affected the P-centre shift b) manipulating the rise time and decay time of a synthetic vowel to determine whether the onset alteration was had more affect on P-centre than the offset manipulation c) and whether the duration of a vowel affected the P-centre, if other attributes (amplitude, spectral contents) were held constant. The third aim - modelling P-centres - was based on these results. The Frequency dependent Amplitude Increase Model of P-centre location (FAIM) was developed using a modelling protocol, the APU GammaTone Filterbank and the speech from different speakers. The P-centres of the stimuli corpus were highly predicted by attributes of the increase in amplitude within one output channel of the filterbank. When this was used to make predictions of the P-centres for all the stimuli used in the thesis, 85[percent] of the observed variance was accounted for. The FAIM approach combines aspects of previous, speech and non speech models (Gordon 1987, Marcus 1981, Vos and Rasch 1981). P-centre were thus modelled in a non speech specific, local manner.

Self-organizing map classifier for stressed speech recognition

NASA Astrophysics Data System (ADS)

Partila, Pavol; Tovarek, Jaromir; Voznak, Miroslav

2016-05-01

This paper presents a method for detecting speech under stress using Self-Organizing Maps. Most people who are exposed to stressful situations can not adequately respond to stimuli. Army, police, and fire department occupy the largest part of the environment that are typical of an increased number of stressful situations. The role of men in action is controlled by the control center. Control commands should be adapted to the psychological state of a man in action. It is known that the psychological changes of the human body are also reflected physiologically, which consequently means the stress effected speech. Therefore, it is clear that the speech stress recognizing system is required in the security forces. One of the possible classifiers, which are popular for its flexibility, is a self-organizing map. It is one type of the artificial neural networks. Flexibility means independence classifier on the character of the input data. This feature is suitable for speech processing. Human Stress can be seen as a kind of emotional state. Mel-frequency cepstral coefficients, LPC coefficients, and prosody features were selected for input data. These coefficients were selected for their sensitivity to emotional changes. The calculation of the parameters was performed on speech recordings, which can be divided into two classes, namely the stress state recordings and normal state recordings. The benefit of the experiment is a method using SOM classifier for stress speech detection. Results showed the advantage of this method, which is input data flexibility.

Phonological Feature Repetition Suppression in the Left Inferior Frontal Gyrus.

PubMed

Okada, Kayoko; Matchin, William; Hickok, Gregory

2018-06-07

Models of speech production posit a role for the motor system, predominantly the posterior inferior frontal gyrus, in encoding complex phonological representations for speech production, at the phonemic, syllable, and word levels [Roelofs, A. A dorsal-pathway account of aphasic language production: The WEAVER++/ARC model. Cortex, 59(Suppl. C), 33-48, 2014; Hickok, G. Computational neuroanatomy of speech production. Nature Reviews Neuroscience, 13, 135-145, 2012; Guenther, F. H. Cortical interactions underlying the production of speech sounds. Journal of Communication Disorders, 39, 350-365, 2006]. However, phonological theory posits subphonemic units of representation, namely phonological features [Chomsky, N., & Halle, M. The sound pattern of English, 1968; Jakobson, R., Fant, G., & Halle, M. Preliminaries to speech analysis. The distinctive features and their correlates. Cambridge, MA: MIT Press, 1951], that specify independent articulatory parameters of speech sounds, such as place and manner of articulation. Therefore, motor brain systems may also incorporate phonological features into speech production planning units. Here, we add support for such a role with an fMRI experiment of word sequence production using a phonemic similarity manipulation. We adapted and modified the experimental paradigm of Oppenheim and Dell [Oppenheim, G. M., & Dell, G. S. Inner speech slips exhibit lexical bias, but not the phonemic similarity effect. Cognition, 106, 528-537, 2008; Oppenheim, G. M., & Dell, G. S. Motor movement matters: The flexible abstractness of inner speech. Memory & Cognition, 38, 1147-1160, 2010]. Participants silently articulated words cued by sequential visual presentation that varied in degree of phonological feature overlap in consonant onset position: high overlap (two shared phonological features; e.g., /r/ and /l/) or low overlap (one shared phonological feature, e.g., /r/ and /b/). We found a significant repetition suppression effect in the left posterior inferior frontal gyrus, with increased activation for phonologically dissimilar words compared with similar words. These results suggest that phonemes, particularly phonological features, are part of the planning units of the motor speech system.

Transcranial electric stimulation for the investigation of speech perception and comprehension

PubMed Central

Zoefel, Benedikt; Davis, Matthew H.

2017-01-01

ABSTRACT Transcranial electric stimulation (tES), comprising transcranial direct current stimulation (tDCS) and transcranial alternating current stimulation (tACS), involves applying weak electrical current to the scalp, which can be used to modulate membrane potentials and thereby modify neural activity. Critically, behavioural or perceptual consequences of this modulation provide evidence for a causal role of neural activity in the stimulated brain region for the observed outcome. We present tES as a tool for the investigation of which neural responses are necessary for successful speech perception and comprehension. We summarise existing studies, along with challenges that need to be overcome, potential solutions, and future directions. We conclude that, although standardised stimulation parameters still need to be established, tES is a promising tool for revealing the neural basis of speech processing. Future research can use this method to explore the causal role of brain regions and neural processes for the perception and comprehension of speech. PMID:28670598

Respiratory Constraints in Verbal and Non-verbal Communication.

PubMed

Włodarczak, Marcin; Heldner, Mattias

2017-01-01

In the present paper we address the old question of respiratory planning in speech production. We recast the problem in terms of speakers' communicative goals and propose that speakers try to minimize respiratory effort in line with the H&H theory. We analyze respiratory cycles coinciding with no speech (i.e., silence), short verbal feedback expressions (SFE's) as well as longer vocalizations in terms of parameters of the respiratory cycle and find little evidence for respiratory planning in feedback production. We also investigate timing of speech and SFEs in the exhalation and contrast it with nods. We find that while speech is strongly tied to the exhalation onset, SFEs are distributed much more uniformly throughout the exhalation and are often produced on residual air. Given that nods, which do not have any respiratory constraints, tend to be more frequent toward the end of an exhalation, we propose a mechanism whereby respiratory patterns are determined by the trade-off between speakers' communicative goals and respiratory constraints.

Language specificity in the perception of voiceless sibilant fricatives in Japanese and English: Implications for cross-language differences in speech-sound development

PubMed Central

Li, Fangfang; Munson, Benjamin; Edwards, Jan; Yoneyama, Kiyoko; Hall, Kathleen

2011-01-01

Both English and Japanese have two voiceless sibilant fricatives, an anterior fricative ∕s∕ contrasting with a more posterior fricative ∕∫∕. When children acquire sibilant fricatives, English children typically substitute [s] for ∕∫∕, whereas Japanese children typically substitute [∫] for ∕∫∕. This study examined English- and Japanese-speaking adults’ perception of children’s productions of voiceless sibilant fricatives to investigate whether the apparent asymmetry in the acquisition of voiceless sibilant fricatives reported previously in the two languages was due in part to how adults perceive children’s speech. The results of this study show that adult speakers of English and Japanese weighed acoustic parameters differently when identifying fricatives produced by children and that these differences explain, in part, the apparent cross-language asymmetry in fricative acquisition. This study shows that generalizations about universal and language-specific patterns in speech-sound development cannot be determined without considering all sources of variation including speech perception. PMID:21361456

A Novel Application System of Assessing the Pronunciation Differences Between Chinese Children and Adults.

PubMed

Zhang, Xiaoyang; Xue, Lei; Zhang, Zhi; Zhang, Yiwen

2016-01-01

Health problems about children have been attracting much attention of parents and even the whole society all the time, among which, child-language development is a hot research topic. The experts and scholars have studied and found that the guardians taking appropriate intervention in children at the early stage can promote children's language and cognitive ability development effectively, and carry out analysis of quantity. The intervention of Artificial Intelligence Technology has effect on the autistic spectrum disorders of children obviously. This paper presents a speech signal analysis system for children, with preprocessing of the speaker speech signal, subsequent calculation of the number in the speech of guardians and children, and some other characteristic parameters or indicators (e.g cognizable syllable number, the continuity of the language). With these quantitative analysis tool and parameters, we can evaluate and analyze the quality of children's language and cognitive ability objectively and quantitatively to provide the basis for decision-making criteria for parents. Thereby, they can adopt appropriate measures for children to promote the development of children's language and cognitive status. In this paper, according to the existing study of children's language development, we put forward several indicators in the process of automatic measurement for language development which influence the formation of children's language. From the experimental results we can see that after the pretreatment (including signal enhancement, speech activity detection), both divergence algorithm calculation results and the later words count are quite satisfactory compared with the actual situation.

Detection of cardiac activity changes from human speech

NASA Astrophysics Data System (ADS)

Tovarek, Jaromir; Partila, Pavol; Voznak, Miroslav; Mikulec, Martin; Mehic, Miralem

2015-05-01

Impact of changes in blood pressure and pulse from human speech is disclosed in this article. The symptoms of increased physical activity are pulse, systolic and diastolic pressure. There are many methods of measuring and indicating these parameters. The measurements must be carried out using devices which are not used in everyday life. In most cases, the measurement of blood pressure and pulse following health problems or other adverse feelings. Nowadays, research teams are trying to design and implement modern methods in ordinary human activities. The main objective of the proposal is to reduce the delay between detecting the adverse pressure and to the mentioned warning signs and feelings. Common and frequent activity of man is speaking, while it is known that the function of the vocal tract can be affected by the change in heart activity. Therefore, it can be a useful parameter for detecting physiological changes. A method for detecting human physiological changes by speech processing and artificial neural network classification is described in this article. The pulse and blood pressure changes was induced by physical exercises in this experiment. The set of measured subjects was formed by ten healthy volunteers of both sexes. None of the subjects was a professional athlete. The process of the experiment was divided into phases before, during and after physical training. Pulse, systolic, diastolic pressure was measured and voice activity was recorded after each of them. The results of this experiment describe a method for detecting increased cardiac activity from human speech using artificial neural network.

Effect of the loss of auditory feedback on segmental parameters of vowels of postlingually deafened speakers.

PubMed

Schenk, Barbara S; Baumgartner, Wolf Dieter; Hamzavi, Jafar Sasan

2003-12-01

The most obvious and best documented changes in speech of postlingually deafened speakers are the rate, fundamental frequency, and volume (energy). These changes are due to the lack of auditory feedback. But auditory feedback affects not only the suprasegmental parameters of speech. The aim of this study was to determine the change at the segmental level of speech in terms of vowel formants. Twenty-three postlingually deafened and 18 normally hearing speakers were recorded reading a German text. The frequencies of the first and second formants and the vowel spaces of selected vowels in word-in-context condition were compared. All first formant frequencies (F1) of the postlingually deafened speakers were significantly different from those of the normally hearing people. The values of F1 were higher for the vowels /e/ (418+/-61 Hz compared with 359+/-52 Hz, P=0.006) and /o/ (459+/-58 compared with 390+/-45 Hz, P=0.0003) and lower for /a/ (765+/-115 Hz compared with 851+/-146 Hz, P=0.038). The second formant frequency (F2) only showed a significant increase for the vowel/e/(2016+/-347 Hz compared with 2279+/-250 Hz, P=0.012). The postlingually deafened people were divided into two subgroups according to duration of deafness (shorter/longer than 10 years of deafness). There was no significant difference in formant changes between the two groups. Our report demonstrated an effect of auditory feedback also on segmental features of speech of postlingually deafened people.

Perceptual Learning and Auditory Training in Cochlear Implant Recipients

PubMed Central

Fu, Qian-Jie; Galvin, John J.

2007-01-01

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

The evaluation of bulbar dysfunction in amyotrophic lateral sclerosis: survey of clinical practice patterns in the United States.

PubMed

Plowman, Emily K; Tabor, Lauren C; Wymer, James; Pattee, Gary

2017-08-01

Speech and swallowing impairments are highly prevalent in individuals with amyotrophic lateral sclerosis (ALS) and contribute to reduced quality of life, malnutrition, aspiration, pneumonia and death. Established practice parameters for bulbar dysfunction in ALS do not currently exist. The aim of this study was to identify current practice patterns for the evaluation of speech and swallowing function within participating Northeast ALS clinics in the United States. A 15-item survey was emailed to all registered NEALS centres. Thirty-eight sites completed the survey. The majority (92%) offered Speech-Language Pathology, augmentative and alternative communication (71%), and dietician (92%) health care services. The ALS Functional Rating Scale-Revised and body weight represented the only parameters routinely collected in greater then 90% of responding sites. Referral for modified barium swallow study was routinely utilised in only 27% of sites and the use of percutaneous gastrostomy tubes in ALS patient care was found to vary considerably. This survey reveals significant variability and inconsistency in the management of bulbar dysfunction in ALS across NEALS sites. We conclude that a great need exists for the development of bulbar practice guidelines in ALS clinical care to accurately detect and monitor bulbar dysfunction.

Multimedia Classifier

NASA Astrophysics Data System (ADS)

Costache, G. N.; Gavat, I.

2004-09-01

Along with the aggressive growing of the amount of digital data available (text, audio samples, digital photos and digital movies joined all in the multimedia domain) the need for classification, recognition and retrieval of this kind of data became very important. In this paper will be presented a system structure to handle multimedia data based on a recognition perspective. The main processing steps realized for the interesting multimedia objects are: first, the parameterization, by analysis, in order to obtain a description based on features, forming the parameter vector; second, a classification, generally with a hierarchical structure to make the necessary decisions. For audio signals, both speech and music, the derived perceptual features are the melcepstral (MFCC) and the perceptual linear predictive (PLP) coefficients. For images, the derived features are the geometric parameters of the speaker mouth. The hierarchical classifier consists generally in a clustering stage, based on the Kohonnen Self-Organizing Maps (SOM) and a final stage, based on a powerful classification algorithm called Support Vector Machines (SVM). The system, in specific variants, is applied with good results in two tasks: the first, is a bimodal speech recognition which uses features obtained from speech signal fused to features obtained from speaker's image and the second is a music retrieval from large music database.

Approximated mutual information training for speech recognition using myoelectric signals.

PubMed

Guo, Hua J; Chan, A D C

2006-01-01

A new training algorithm called the approximated maximum mutual information (AMMI) is proposed to improve the accuracy of myoelectric speech recognition using hidden Markov models (HMMs). Previous studies have demonstrated that automatic speech recognition can be performed using myoelectric signals from articulatory muscles of the face. Classification of facial myoelectric signals can be performed using HMMs that are trained using the maximum likelihood (ML) algorithm; however, this algorithm maximizes the likelihood of the observations in the training sequence, which is not directly associated with optimal classification accuracy. The AMMI training algorithm attempts to maximize the mutual information, thereby training the HMMs to optimize their parameters for discrimination. Our results show that AMMI training consistently reduces the error rates compared to these by the ML training, increasing the accuracy by approximately 3% on average.

Voice stress analysis

NASA Technical Reports Server (NTRS)

Brenner, Malcolm; Shipp, Thomas

1988-01-01

In a study of the validity of eight candidate voice measures (fundamental frequency, amplitude, speech rate, frequency jitter, amplitude shimmer, Psychological Stress Evaluator scores, energy distribution, and the derived measure of the above measures) for determining psychological stress, 17 males age 21 to 35 were subjected to a tracking task on a microcomputer CRT while parameters of vocal production as well as heart rate were measured. Findings confirm those of earlier studies that increases in fundamental frequency, amplitude, and speech rate are found in speakers involved in extreme levels of stress. In addition, it was found that the same changes appear to occur in a regular fashion within a more subtle level of stress that may be characteristic, for example, of routine flying situations. None of the individual speech measures performed as robustly as did heart rate.

Does computer-synthesized speech manifest personality? Experimental tests of recognition, similarity-attraction, and consistency-attraction.

PubMed

Nass, C; Lee, K M

2001-09-01

Would people exhibit similarity-attraction and consistency-attraction toward unambiguously computer-generated speech even when personality is clearly not relevant? In Experiment 1, participants (extrovert or introvert) heard a synthesized voice (extrovert or introvert) on a book-buying Web site. Participants accurately recognized personality cues in text to speech and showed similarity-attraction in their evaluation of the computer voice, the book reviews, and the reviewer. Experiment 2, in a Web auction context, added personality of the text to the previous design. The results replicated Experiment 1 and demonstrated consistency (voice and text personality)-attraction. To maximize liking and trust, designers should set parameters, for example, words per minute or frequency range, that create a personality that is consistent with the user and the content being presented.

Effects of Increasing Sound Pressure Level on Lip and Jaw Movement Parameters and Consistency in Young Adults

ERIC Educational Resources Information Center

Huber, Jessica E.; Chandrasekaran, Bharath

2006-01-01

Purpose: Examination of movement parameters and consistency has been used to infer underlying neural control of movement. However, there has been no systematic investigation of whether the way individuals are asked (or cued) to increase loudness alters articulation. This study examined whether different cues to elicit louder speech induce…

JPEG XS-based frame buffer compression inside HEVC for power-aware video compression

NASA Astrophysics Data System (ADS)

Willème, Alexandre; Descampe, Antonin; Rouvroy, Gaël.; Pellegrin, Pascal; Macq, Benoit

2017-09-01

With the emergence of Ultra-High Definition video, reference frame buffers (FBs) inside HEVC-like encoders and decoders have to sustain huge bandwidth. The power consumed by these external memory accesses accounts for a significant share of the codec's total consumption. This paper describes a solution to significantly decrease the FB's bandwidth, making HEVC encoder more suitable for use in power-aware applications. The proposed prototype consists in integrating an embedded lightweight, low-latency and visually lossless codec at the FB interface inside HEVC in order to store each reference frame as several compressed bitstreams. As opposed to previous works, our solution compresses large picture areas (ranging from a CTU to a frame stripe) independently in order to better exploit the spatial redundancy found in the reference frame. This work investigates two data reuse schemes namely Level-C and Level-D. Our approach is made possible thanks to simplified motion estimation mechanisms further reducing the FB's bandwidth and inducing very low quality degradation. In this work, we integrated JPEG XS, the upcoming standard for lightweight low-latency video compression, inside HEVC. In practice, the proposed implementation is based on HM 16.8 and on XSM 1.1.2 (JPEG XS Test Model). Through this paper, the architecture of our HEVC with JPEG XS-based frame buffer compression is described. Then its performance is compared to HM encoder. Compared to previous works, our prototype provides significant external memory bandwidth reduction. Depending on the reuse scheme, one can expect bandwidth and FB size reduction ranging from 50% to 83.3% without significant quality degradation.

Adaptive Distributed Video Coding with Correlation Estimation using Expectation Propagation

PubMed Central

Cui, Lijuan; Wang, Shuang; Jiang, Xiaoqian; Cheng, Samuel

2013-01-01

Distributed video coding (DVC) is rapidly increasing in popularity by the way of shifting the complexity from encoder to decoder, whereas no compression performance degrades, at least in theory. In contrast with conventional video codecs, the inter-frame correlation in DVC is explored at decoder based on the received syndromes of Wyner-Ziv (WZ) frame and side information (SI) frame generated from other frames available only at decoder. However, the ultimate decoding performances of DVC are based on the assumption that the perfect knowledge of correlation statistic between WZ and SI frames should be available at decoder. Therefore, the ability of obtaining a good statistical correlation estimate is becoming increasingly important in practical DVC implementations. Generally, the existing correlation estimation methods in DVC can be classified into two main types: pre-estimation where estimation starts before decoding and on-the-fly (OTF) estimation where estimation can be refined iteratively during decoding. As potential changes between frames might be unpredictable or dynamical, OTF estimation methods usually outperforms pre-estimation techniques with the cost of increased decoding complexity (e.g., sampling methods). In this paper, we propose a low complexity adaptive DVC scheme using expectation propagation (EP), where correlation estimation is performed OTF as it is carried out jointly with decoding of the factor graph-based DVC code. Among different approximate inference methods, EP generally offers better tradeoff between accuracy and complexity. Experimental results show that our proposed scheme outperforms the benchmark state-of-the-art DISCOVER codec and other cases without correlation tracking, and achieves comparable decoding performance but with significantly low complexity comparing with sampling method. PMID:23750314

Single-layer HDR video coding with SDR backward compatibility

NASA Astrophysics Data System (ADS)

Lasserre, S.; François, E.; Le Léannec, F.; Touzé, D.

2016-09-01

The migration from High Definition (HD) TV to Ultra High Definition (UHD) is already underway. In addition to an increase of picture spatial resolution, UHD will bring more color and higher contrast by introducing Wide Color Gamut (WCG) and High Dynamic Range (HDR) video. As both Standard Dynamic Range (SDR) and HDR devices will coexist in the ecosystem, the transition from Standard Dynamic Range (SDR) to HDR will require distribution solutions supporting some level of backward compatibility. This paper presents a new HDR content distribution scheme, named SL-HDR1, using a single layer codec design and providing SDR compatibility. The solution is based on a pre-encoding HDR-to-SDR conversion, generating a backward compatible SDR video, with side dynamic metadata. The resulting SDR video is then compressed, distributed and decoded using standard-compliant decoders (e.g. HEVC Main 10 compliant). The decoded SDR video can be directly rendered on SDR displays without adaptation. Dynamic metadata of limited size are generated by the pre-processing and used to reconstruct the HDR signal from the decoded SDR video, using a post-processing that is the functional inverse of the pre-processing. Both HDR quality and artistic intent are preserved. Pre- and post-processing are applied independently per picture, do not involve any inter-pixel dependency, and are codec agnostic. Compression performance, and SDR quality are shown to be solidly improved compared to the non-backward and backward-compatible approaches, respectively using the Perceptual Quantization (PQ) and Hybrid Log Gamma (HLG) Opto-Electronic Transfer Functions (OETF).

Adaptive distributed video coding with correlation estimation using expectation propagation

NASA Astrophysics Data System (ADS)

Cui, Lijuan; Wang, Shuang; Jiang, Xiaoqian; Cheng, Samuel

2012-10-01

Distributed video coding (DVC) is rapidly increasing in popularity by the way of shifting the complexity from encoder to decoder, whereas no compression performance degrades, at least in theory. In contrast with conventional video codecs, the inter-frame correlation in DVC is explored at decoder based on the received syndromes of Wyner-Ziv (WZ) frame and side information (SI) frame generated from other frames available only at decoder. However, the ultimate decoding performances of DVC are based on the assumption that the perfect knowledge of correlation statistic between WZ and SI frames should be available at decoder. Therefore, the ability of obtaining a good statistical correlation estimate is becoming increasingly important in practical DVC implementations. Generally, the existing correlation estimation methods in DVC can be classified into two main types: pre-estimation where estimation starts before decoding and on-the-fly (OTF) estimation where estimation can be refined iteratively during decoding. As potential changes between frames might be unpredictable or dynamical, OTF estimation methods usually outperforms pre-estimation techniques with the cost of increased decoding complexity (e.g., sampling methods). In this paper, we propose a low complexity adaptive DVC scheme using expectation propagation (EP), where correlation estimation is performed OTF as it is carried out jointly with decoding of the factor graph-based DVC code. Among different approximate inference methods, EP generally offers better tradeoff between accuracy and complexity. Experimental results show that our proposed scheme outperforms the benchmark state-of-the-art DISCOVER codec and other cases without correlation tracking, and achieves comparable decoding performance but with significantly low complexity comparing with sampling method.

Adaptive Distributed Video Coding with Correlation Estimation using Expectation Propagation.

PubMed

Cui, Lijuan; Wang, Shuang; Jiang, Xiaoqian; Cheng, Samuel

2012-10-15

Distributed video coding (DVC) is rapidly increasing in popularity by the way of shifting the complexity from encoder to decoder, whereas no compression performance degrades, at least in theory. In contrast with conventional video codecs, the inter-frame correlation in DVC is explored at decoder based on the received syndromes of Wyner-Ziv (WZ) frame and side information (SI) frame generated from other frames available only at decoder. However, the ultimate decoding performances of DVC are based on the assumption that the perfect knowledge of correlation statistic between WZ and SI frames should be available at decoder. Therefore, the ability of obtaining a good statistical correlation estimate is becoming increasingly important in practical DVC implementations. Generally, the existing correlation estimation methods in DVC can be classified into two main types: pre-estimation where estimation starts before decoding and on-the-fly (OTF) estimation where estimation can be refined iteratively during decoding. As potential changes between frames might be unpredictable or dynamical, OTF estimation methods usually outperforms pre-estimation techniques with the cost of increased decoding complexity (e.g., sampling methods). In this paper, we propose a low complexity adaptive DVC scheme using expectation propagation (EP), where correlation estimation is performed OTF as it is carried out jointly with decoding of the factor graph-based DVC code. Among different approximate inference methods, EP generally offers better tradeoff between accuracy and complexity. Experimental results show that our proposed scheme outperforms the benchmark state-of-the-art DISCOVER codec and other cases without correlation tracking, and achieves comparable decoding performance but with significantly low complexity comparing with sampling method.

The Americleft Speech Project: A Training and Reliability Study

PubMed Central

Chapman, Kathy L.; Baylis, Adriane; Trost-Cardamone, Judith; Cordero, Kelly Nett; Dixon, Angela; Dobbelsteyn, Cindy; Thurmes, Anna; Wilson, Kristina; Harding-Bell, Anne; Sweeney, Triona; Stoddard, Gregory; Sell, Debbie

2017-01-01

Objective To describe the results of two reliability studies and to assess the effect of training on interrater reliability scores. Design The first study (1) examined interrater and intrarater reliability scores (weighted and unweighted kappas) and (2) compared interrater reliability scores before and after training on the use of the Cleft Audit Protocol for Speech–Augmented (CAPS-A) with British English-speaking children. The second study examined interrater and intrarater reliability on a modified version of the CAPS-A (CAPS-A Americleft Modification) with American and Canadian English-speaking children. Finally, comparisons were made between the interrater and intrarater reliability scores obtained for Study 1 and Study 2. Participants The participants were speech-language pathologists from the Americleft Speech Project. Results In Study 1, interrater reliability scores improved for 6 of the 13 parameters following training on the CAPS-A protocol. Comparison of the reliability results for the two studies indicated lower scores for Study 2 compared with Study 1. However, this appeared to be an artifact of the kappa statistic that occurred due to insufficient variability in the reliability samples for Study 2. When percent agreement scores were also calculated, the ratings appeared similar across Study 1 and Study 2. Conclusion The findings of this study suggested that improvements in interrater reliability could be obtained following a program of systematic training. However, improvements were not uniform across all parameters. Acceptable levels of reliability were achieved for those parameters most important for evaluation of velopharyngeal function. PMID:25531738

Exploring the anatomical encoding of voice with a mathematical model of the vocal system.

PubMed

Assaneo, M Florencia; Sitt, Jacobo; Varoquaux, Gael; Sigman, Mariano; Cohen, Laurent; Trevisan, Marcos A

2016-11-01

The faculty of language depends on the interplay between the production and perception of speech sounds. A relevant open question is whether the dimensions that organize voice perception in the brain are acoustical or depend on properties of the vocal system that produced it. One of the main empirical difficulties in answering this question is to generate sounds that vary along a continuum according to the anatomical properties the vocal apparatus that produced them. Here we use a mathematical model that offers the unique possibility of synthesizing vocal sounds by controlling a small set of anatomically based parameters. In a first stage the quality of the synthetic voice was evaluated. Using specific time traces for sub-glottal pressure and tension of the vocal folds, the synthetic voices generated perceptual responses, which are indistinguishable from those of real speech. The synthesizer was then used to investigate how the auditory cortex responds to the perception of voice depending on the anatomy of the vocal apparatus. Our fMRI results show that sounds are perceived as human vocalizations when produced by a vocal system that follows a simple relationship between the size of the vocal folds and the vocal tract. We found that these anatomical parameters encode the perceptual vocal identity (male, female, child) and show that the brain areas that respond to human speech also encode vocal identity. On the basis of these results, we propose that this low-dimensional model of the vocal system is capable of generating realistic voices and represents a novel tool to explore the voice perception with a precise control of the anatomical variables that generate speech. Furthermore, the model provides an explanation of how auditory cortices encode voices in terms of the anatomical parameters of the vocal system. Copyright © 2016 Elsevier Inc. All rights reserved.

A physiologically-inspired model reproducing the speech intelligibility benefit in cochlear implant listeners with residual acoustic hearing.

PubMed

Zamaninezhad, Ladan; Hohmann, Volker; Büchner, Andreas; Schädler, Marc René; Jürgens, Tim

2017-02-01

This study introduces a speech intelligibility model for cochlear implant users with ipsilateral preserved acoustic hearing that aims at simulating the observed speech-in-noise intelligibility benefit when receiving simultaneous electric and acoustic stimulation (EA-benefit). The model simulates the auditory nerve spiking in response to electric and/or acoustic stimulation. The temporally and spatially integrated spiking patterns were used as the final internal representation of noisy speech. Speech reception thresholds (SRTs) in stationary noise were predicted for a sentence test using an automatic speech recognition framework. The model was employed to systematically investigate the effect of three physiologically relevant model factors on simulated SRTs: (1) the spatial spread of the electric field which co-varies with the number of electrically stimulated auditory nerves, (2) the "internal" noise simulating the deprivation of auditory system, and (3) the upper bound frequency limit of acoustic hearing. The model results show that the simulated SRTs increase monotonically with increasing spatial spread for fixed internal noise, and also increase with increasing the internal noise strength for a fixed spatial spread. The predicted EA-benefit does not follow such a systematic trend and depends on the specific combination of the model parameters. Beyond 300 Hz, the upper bound limit for preserved acoustic hearing is less influential on speech intelligibility of EA-listeners in stationary noise. The proposed model-predicted EA-benefits are within the range of EA-benefits shown by 18 out of 21 actual cochlear implant listeners with preserved acoustic hearing. Copyright © 2016 Elsevier B.V. All rights reserved.

Gender disparity in subcortical encoding of binaurally presented speech stimuli: an auditory evoked potentials study.

PubMed

Ahadi, Mohsen; Pourbakht, Akram; Jafari, Amir Homayoun; Shirjian, Zahra; Jafarpisheh, Amir Salar

2014-06-01

To investigate the influence of gender on subcortical representation of speech acoustic parameters where simultaneously presented to both ears. Two-channel speech-evoked auditory brainstem responses were obtained in 25 female and 23 male normal hearing young adults by using binaural presentation of the 40 ms synthetic consonant-vowel/da/, and the encoding of the fast and slow elements of speech stimuli at subcortical level were compared in the temporal and spectral domains between the sexes using independent sample, two tailed t-test. Highly detectable responses were established in both groups. Analysis in the time domain revealed earlier and larger Fast-onset-responses in females but there was no gender related difference in sustained segment and offset of the response. Interpeak intervals between Frequency Following Response peaks were also invariant to sex. Based on shorter onset responses in females, composite onset measures were also sex dependent. Analysis in the spectral domain showed more robust and better representation of fundamental frequency as well as the first formant and high frequency components of first formant in females than in males. Anatomical, biological and biochemical distinctions between females and males could alter the neural encoding of the acoustic cues of speech stimuli at subcortical level. Females have an advantage in binaural processing of the slow and fast elements of speech. This could be a physiological evidence for better identification of speaker and emotional tone of voice, as well as better perceiving the phonetic information of speech in women. Copyright © 2013 Elsevier Ireland Ltd. All rights reserved.

Classification of vocal aging using parameters extracted from the glottal signal.

PubMed

Forero Mendoza, Leonardo A; Cataldo, Edson; Vellasco, Marley M B R; Silva, Marco A; Apolinário, José A

2014-09-01

This article proposes and evaluates a method to classify vocal aging using artificial neural network (ANN) and support vector machine (SVM), using the parameters extracted from the speech signal as inputs. For each recorded speech, from a corpus of male and female speakers of different ages, the corresponding glottal signal is obtained using an inverse filtering algorithm. The Mel Frequency Cepstrum Coefficients (MFCC) also extracted from the voice signal and the features extracted from the glottal signal are supplied to an ANN and an SVM with a previous selection. The selection is performed by a wrapper approach of the most relevant parameters. Three groups are considered for the aging-voice classification: young (aged 15-30 years), adult (aged 31-60 years), and senior (aged 61-90 years). The results are compared using different possibilities: with only the parameters extracted from the glottal signal, with only the MFCC, and with a combination of both. The results demonstrate that the best classification rate is obtained using the glottal signal features, which is a novel result and the main contribution of this article. Copyright © 2014 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

Development of a mobile satellite communication unit

NASA Technical Reports Server (NTRS)

Suzuki, Ryutaro; Ikegami, Tetsushi; Hamamoto, Naokazu; Taguchi, Tetsu; Endo, Nobuhiro; Yamamoto, Osamu; Ichiyoshi, Osamu

1988-01-01

A compact 210(W) x 280(H) x 330(D) mm mobile terminal capable of transmitting voice and data through L-band mobile satellites is described. The Voice Codec can convert an analog voice to or from digital codes at rates of 9.6, 8 and 4.8 kb/s by an MPC algorithm. The terminal functions with a single 12 V power supplied vehicle battery. The equipment can operate at any L-band frequency allocated for mobile uses in a full duplex mode and will soon be put into a field test via Japans's ETS-V satellite.

A Novel Application System of Assessing the Pronunciation Differences Between Chinese Children and Adults

PubMed Central

Zhang, Xiaoyang; Xue, Lei; Zhang, Zhi; Zhang, Yiwen

2016-01-01

Background: Health problems about children have been attracting much attention of parents and even the whole society all the time, among which, child-language development is a hot research topic. The experts and scholars have studied and found that the guardians taking appropriate intervention in children at the early stage can promote children’s language and cognitive ability development effectively, and carry out analysis of quantity. The intervention of Artificial Intelligence Technology has effect on the autistic spectrum disorders of children obviously. Objective and Methods: This paper presents a speech signal analysis system for children, with preprocessing of the speaker speech signal, subsequent calculation of the number in the speech of guardians and children, and some other characteristic parameters or indicators (e.g cognizable syllable number, the continuity of the language). Results: With these quantitative analysis tool and parameters, we can evaluate and analyze the quality of children’s language and cognitive ability objectively and quantitatively to provide the basis for decision-making criteria for parents. Thereby, they can adopt appropriate measures for children to promote the development of children's language and cognitive status. Conclusion: In this paper, according to the existing study of children’s language development, we put forward several indicators in the process of automatic measurement for language development which influence the formation of children’s language. From the experimental results we can see that after the pretreatment (including signal enhancement, speech activity detection), both divergence algorithm calculation results and the later words count are quite satisfactory compared with the actual situation. PMID:27583037

A constraint-based evolutionary learning approach to the expectation maximization for optimal estimation of the hidden Markov model for speech signal modeling.

PubMed

Huda, Shamsul; Yearwood, John; Togneri, Roberto

2009-02-01

This paper attempts to overcome the tendency of the expectation-maximization (EM) algorithm to locate a local rather than global maximum when applied to estimate the hidden Markov model (HMM) parameters in speech signal modeling. We propose a hybrid algorithm for estimation of the HMM in automatic speech recognition (ASR) using a constraint-based evolutionary algorithm (EA) and EM, the CEL-EM. The novelty of our hybrid algorithm (CEL-EM) is that it is applicable for estimation of the constraint-based models with many constraints and large numbers of parameters (which use EM) like HMM. Two constraint-based versions of the CEL-EM with different fusion strategies have been proposed using a constraint-based EA and the EM for better estimation of HMM in ASR. The first one uses a traditional constraint-handling mechanism of EA. The other version transforms a constrained optimization problem into an unconstrained problem using Lagrange multipliers. Fusion strategies for the CEL-EM use a staged-fusion approach where EM has been plugged with the EA periodically after the execution of EA for a specific period of time to maintain the global sampling capabilities of EA in the hybrid algorithm. A variable initialization approach (VIA) has been proposed using a variable segmentation to provide a better initialization for EA in the CEL-EM. Experimental results on the TIMIT speech corpus show that CEL-EM obtains higher recognition accuracies than the traditional EM algorithm as well as a top-standard EM (VIA-EM, constructed by applying the VIA to EM).

Early Recovery of Aphasia through Thrombolysis: The Significance of Spontaneous Speech.

PubMed

Furlanis, Giovanni; Ridolfi, Mariana; Polverino, Paola; Menichelli, Alina; Caruso, Paola; Naccarato, Marcello; Sartori, Arianna; Torelli, Lucio; Pesavento, Valentina; Manganotti, Paolo

2018-07-01

Aphasia is one of the most devastating stroke-related consequences for social interaction and daily activities. Aphasia recovery in acute stroke depends on the degree of reperfusion after thrombolysis or thrombectomy. As aphasia assessment tests are often time-consuming for patients with acute stroke, physicians have been developing rapid and simple tests. The aim of our study is to evaluate the improvement of language functions in the earliest stage in patients treated with thrombolysis and in nontreated patients using our rapid screening test. Our study is a single-center prospective observational study conducted at the Stroke Unit of the University Medical Hospital of Trieste (January-December 2016). Patients treated with thrombolysis and nontreated patients underwent 3 aphasia assessments through our rapid screening test (at baseline, 24 hours, and 72 hours). The screening test assesses spontaneous speech, oral comprehension of words, reading aloud and comprehension of written words, oral comprehension of sentences, naming, repetition of words and a sentence, and writing words. The study included 40 patients: 18 patients treated with thrombolysis and 22 nontreated patients. Both groups improved over time. Among all language parameters, spontaneous speech was statistically significant between 24 and 72 hours (P value = .012), and between baseline and 72 hours (P value = .017). Our study demonstrates that patients treated with thrombolysis experience greater improvement in language than the nontreated patients. The difference between the 2 groups is increasingly evident over time. Moreover, spontaneous speech is the parameter marked by the greatest improvement. Copyright © 2018 National Stroke Association. Published by Elsevier Inc. All rights reserved.

[Acoustic voice analysis using the Praat program: comparative study with the Dr. Speech program].

PubMed

Núñez Batalla, Faustino; González Márquez, Rocío; Peláez González, M Belén; González Laborda, Irene; Fernández Fernández, María; Morato Galán, Marta

2014-01-01

The European Laryngological Society (ELS) basic protocol for functional assessment of voice pathology includes 5 different approaches: perception, videostroboscopy, acoustics, aerodynamics and subjective rating by the patient. In this study we focused on acoustic voice analysis. The purpose of the present study was to correlate the results obtained by the commercial software Dr. Speech and the free software Praat in 2 fields: 1. Narrow-band spectrogram (the presence of noise according to Yanagihara, and the presence of subharmonics) (semi-quantitative). 2. Voice acoustic parameters (jitter, shimmer, harmonics-to-noise ratio, fundamental frequency) (quantitative). We studied a total of 99 voice samples from individuals with Reinke's oedema diagnosed using videostroboscopy. One independent observer used Dr. Speech 3.0 and a second one used the Praat program (Phonetic Sciences, University of Amsterdam). The spectrographic analysis consisted of obtaining a narrow-band spectrogram from the previous digitalised voice samples by the 2 independent observers. They then determined the presence of noise in the spectrogram, using the Yanagihara grades, as well as the presence of subharmonics. As a final result, the acoustic parameters of jitter, shimmer, harmonics-to-noise ratio and fundamental frequency were obtained from the 2 acoustic analysis programs. The results indicated that the sound spectrogram and the numerical values obtained for shimmer and jitter were similar for both computer programs, even though types 1, 2 and 3 voice samples were analysed. The Praat and Dr. Speech programs provide similar results in the acoustic analysis of pathological voices. Copyright © 2013 Elsevier España, S.L. All rights reserved.

Classifier Subset Selection for the Stacked Generalization Method Applied to Emotion Recognition in Speech

PubMed Central

Álvarez, Aitor; Sierra, Basilio; Arruti, Andoni; López-Gil, Juan-Miguel; Garay-Vitoria, Nestor

2015-01-01

In this paper, a new supervised classification paradigm, called classifier subset selection for stacked generalization (CSS stacking), is presented to deal with speech emotion recognition. The new approach consists of an improvement of a bi-level multi-classifier system known as stacking generalization by means of an integration of an estimation of distribution algorithm (EDA) in the first layer to select the optimal subset from the standard base classifiers. The good performance of the proposed new paradigm was demonstrated over different configurations and datasets. First, several CSS stacking classifiers were constructed on the RekEmozio dataset, using some specific standard base classifiers and a total of 123 spectral, quality and prosodic features computed using in-house feature extraction algorithms. These initial CSS stacking classifiers were compared to other multi-classifier systems and the employed standard classifiers built on the same set of speech features. Then, new CSS stacking classifiers were built on RekEmozio using a different set of both acoustic parameters (extended version of the Geneva Minimalistic Acoustic Parameter Set (eGeMAPS)) and standard classifiers and employing the best meta-classifier of the initial experiments. The performance of these two CSS stacking classifiers was evaluated and compared. Finally, the new paradigm was tested on the well-known Berlin Emotional Speech database. We compared the performance of single, standard stacking and CSS stacking systems using the same parametrization of the second phase. All of the classifications were performed at the categorical level, including the six primary emotions plus the neutral one. PMID:26712757

Side-information-dependent correlation channel estimation in hash-based distributed video coding.

PubMed

Deligiannis, Nikos; Barbarien, Joeri; Jacobs, Marc; Munteanu, Adrian; Skodras, Athanassios; Schelkens, Peter

2012-04-01

In the context of low-cost video encoding, distributed video coding (DVC) has recently emerged as a potential candidate for uplink-oriented applications. This paper builds on a concept of correlation channel (CC) modeling, which expresses the correlation noise as being statistically dependent on the side information (SI). Compared with classical side-information-independent (SII) noise modeling adopted in current DVC solutions, it is theoretically proven that side-information-dependent (SID) modeling improves the Wyner-Ziv coding performance. Anchored in this finding, this paper proposes a novel algorithm for online estimation of the SID CC parameters based on already decoded information. The proposed algorithm enables bit-plane-by-bit-plane successive refinement of the channel estimation leading to progressively improved accuracy. Additionally, the proposed algorithm is included in a novel DVC architecture that employs a competitive hash-based motion estimation technique to generate high-quality SI at the decoder. Experimental results corroborate our theoretical gains and validate the accuracy of the channel estimation algorithm. The performance assessment of the proposed architecture shows remarkable and consistent coding gains over a germane group of state-of-the-art distributed and standard video codecs, even under strenuous conditions, i.e., large groups of pictures and highly irregular motion content.

Speech-on-speech masking in a front-back dimension and analysis of binaural parameters in rooms using MLS methods

NASA Astrophysics Data System (ADS)

Aaronson, Neil L.

This dissertation deals with questions important to the problem of human sound source localization in rooms, starting with perceptual studies and moving on to physical measurements made in rooms. In Chapter 1, a perceptual study is performed relevant to a specific phenomenon the effect of speech reflections occurring in the front-back dimension and the ability of humans to segregate that from unreflected speech. Distracters were presented from the same source as the target speech, a loudspeaker directly in front of the listener, and also from a loudspeaker directly behind the listener, delayed relative to the front loudspeaker. Steps were taken to minimize the contributions of binaural difference cues. For all delays within +/-32 ms, a release from informational masking of about 2 dB occurred. This suggested that human listeners are able to segregate speech sources based on spatial cues, even with minimal binaural cues. In moving on to physical measurements in rooms, a method was sought for simultaneous measurement of room characteristics such as impulse response (IR) and reverberation time (RT60), and binaural parameters such as interaural time difference (ITD), interaural level difference (ILD), and the interaural cross-correlation function and coherence. Chapter 2 involves investigations into the usefulness of maximum length sequences (MLS) for these purposes. Comparisons to random telegraph noise (RTN) show that MLS performs better in the measurement of stationary and room transfer functions, IR, and RT60 by an order of magnitude in RMS percent error, even after Wiener filtering and exponential time-domain filtering have improved the accuracy of RTN measurements. Measurements were taken in real rooms in an effort to understand how the reverberant characteristics of rooms affect binaural parameters important to sound source localization. Chapter 3 deals with interaural coherence, a parameter important for localization and perception of auditory source width. MLS were used to measure waveform and envelope coherences in two rooms for various source distances and 0° azimuth through a head-and-torso simulator (KEMAR). A relationship is sought that relates these two types of coherence, since envelope coherence, while an important quantity, is generally less accessible than waveform coherence. A power law relationship is shown to exist between the two that works well within and across bands, for any source distance, and is robust to reverberant conditions of the room. Measurements of ITD, ILD, and coherence in rooms give insight into the way rooms affect these parameters, and in turn, the ability of listeners to localize sounds in rooms. Such measurements, along with room properties, are made and analyzed using MLS methods in Chapter 4. It was found that the pinnae cause incoherence for sound sources incident between 30° and 90°. In human listeners, this does not seem to adversely affect performance in lateralization experiments. The cause of poor coherence in rooms was studied as part of Chapter 4 as well. It was found that rooms affect coherence by introducing variance into the ITD spectra within the bands in which it is measured. A mathematical model to predict the interaural coherence within a band given the standard deviation of the ITD spectrum and the center frequency of the band gives an exponential relationship. This is found to work well in predicting measured coherence given ITD spectrum variance. The pinnae seem to affect the ITD spectrum in a similar way at incident sound angles for which coherence is poor in an anechoic environment.

Effects of exposure to sulfur mustard on speech aerodynamics.

PubMed

Heydari, Fatemeh; Ghanei, Mostafa

2011-01-01

Sulfur mustard is an alkylating agent with highly cytotoxic properties even at low exposure. It was used widely against both military and civilian population by Iraqi forces in the Iraq-Iran war (1983-1988). Although various aspects of mustard gas effects on patients with chemical injury have been relatively well characterized, its effects on speech are still evolving. We evaluated aerodynamics of speech in male patients following sulfur mustard inhalation. In a case-control study patients with chemical injuries (n=19) along with age and sex-matched healthy control group (n=20) were selected. Aerodynamic analyses were performed by using the Glasgow Airflow Measurement System (known as ST1 dysphonia). Results indicated that except mean flow rate, there were statistically significant differences in vital capacity, phonation time, phonation volume, vocal velocity index, total expired volume and phonation quotient of patients between experimental and control groups (P<0.05). This study demonstrated mustard gas can impair different parameters of speech aerodynamics. As a result of this activity, the reader will be able to describe: (1) the evaluation of air flow in relation to speech system dysfunction and efficiency; (2) the effect of sulfur mustard known as mustard gas on respiratory physiology. Copyright © 2011 Elsevier Inc. All rights reserved.

Electrophysiological evidence of functional integration between the language and motor systems in the brain: a study of the speech Bereitschaftspotential.

PubMed

McArdle, J J; Mari, Z; Pursley, R H; Schulz, G M; Braun, A R

2009-02-01

We investigated whether the Bereitschaftspotential (BP), an event related potential believed to reflect motor planning, would be modulated by language-related parameters prior to speech. We anticipated that articulatory complexity would produce effects on the BP distribution similar to those demonstrated for complex limb movements. We also hypothesized that lexical semantic operations would independently impact the BP. Eighteen participants performed 3 speech tasks designed to differentiate lexical semantic and articulatory contributions to the BP. EEG epochs were time-locked to the earliest source of speech movement per trial. Lip movements were assessed using EMG recordings. Doppler imaging was used to determine the onset of tongue movement during speech, providing a means of identification and elimination of potential artifact. Compared to simple repetition, complex articulations produced an anterior shift in the maximum midline BP. Tasks requiring lexical search and selection augmented these effects and independently elicited a left lateralized asymmetry in the frontal distribution. The findings indicate that the BP is significantly modulated by linguistic processing, suggesting that the premotor system might play a role in lexical access. These novel findings support the notion that the motor systems may play a significant role in the formulation of language.

Perceived gender in clear and conversational speech

NASA Astrophysics Data System (ADS)

Booz, Jaime A.

Although many studies have examined acoustic and sociolinguistic differences between male and female speech, the relationship between talker speaking style and perceived gender has not yet been explored. The present study attempts to determine whether clear speech, a style adopted by talkers who perceive some barrier to effective communication, shifts perceptions of femininity for male and female talkers. Much of our understanding of gender perception in voice and speech is based on sustained vowels or single words, eliminating temporal, prosodic, and articulatory cues available in more naturalistic, connected speech. Thus, clear and conversational sentence stimuli, selected from the 41 talkers of the Ferguson Clear Speech Database (Ferguson, 2004) were presented to 17 normal-hearing listeners, aged 18 to 30. They rated the talkers' gender using a visual analog scale with "masculine" and "feminine" endpoints. This response method was chosen to account for within-category shifts of gender perception by allowing nonbinary responses. Mixed-effects regression analysis of listener responses revealed a small but significant effect of speaking style, and this effect was larger for male talkers than female talkers. Because of the high degree of talker variability observed for talker gender, acoustic analyses of these sentences were undertaken to determine the relationship between acoustic changes in clear and conversational speech and perceived femininity. Results of these analyses showed that mean fundamental frequency (fo) and f o standard deviation were significantly correlated to perceived gender for both male and female talkers, and vowel space was significantly correlated only for male talkers. Speaking rate and breathiness measures (CPPS) were not significantly related for either group. Outcomes of this study indicate that adopting a clear speaking style is correlated with increases in perceived femininity. Although the increase was small, some changes associated with making adjustments to improve speech clarity have a larger impact on perceived femininity than others. Using a clear speech strategy alone may not be sufficient for a male speaker to be perceived as female, but could be used as one of many tools to help speakers achieve more "feminine" speech, in conjunction with more specific strategies targeting the acoustic parameters outlined in this study.

Evaluation of the mobile phone electromagnetic radiation on serum iron parameters in rats.

PubMed

Çetkin, Murat; Demirel, Can; Kızılkan, Neşe; Aksoy, Nur; Erbağcı, Hülya

2017-03-01

Electromagnetic fields (EMF) created by mobile phones during communication have harmful effects on different organs. It was aimed to investigate the effects of an EMF created by a mobile phone on serum iron level, ferritin, unsaturated iron binding capacity and total iron binding capacity within a rat experiment model. A total of 32 male Wistar albino rats were randomly divided into the control, sham, mobile phone speech (2h/day) and stand by (12 h/day) groups. The speech and stand by groups were subjected to the EMF for a total of 10 weeks. No statistically significant difference was observed between the serum iron and ferritin values of the rats in the speech and stand by groups than the control and sham groups (p>0.05). The unsaturated iron binding capacity and total iron capacity values of the rats in the speech and stand by groups were significantly lower in comparison to the control group (p<0.01). It was found that exposure to EMF created by mobile phones affected unsaturated iron binding capacity and total iron binding capacity negatively.

Automatic conversational scene analysis in children with Asperger syndrome/high-functioning autism and typically developing peers.

PubMed

Tavano, Alessandro; Pesarin, Anna; Murino, Vittorio; Cristani, Marco

2014-01-01

Individuals with Asperger syndrome/High Functioning Autism fail to spontaneously attribute mental states to the self and others, a life-long phenotypic characteristic known as mindblindness. We hypothesized that mindblindness would affect the dynamics of conversational interaction. Using generative models, in particular Gaussian mixture models and observed influence models, conversations were coded as interacting Markov processes, operating on novel speech/silence patterns, termed Steady Conversational Periods (SCPs). SCPs assume that whenever an agent's process changes state (e.g., from silence to speech), it causes a general transition of the entire conversational process, forcing inter-actant synchronization. SCPs fed into observed influence models, which captured the conversational dynamics of children and adolescents with Asperger syndrome/High Functioning Autism, and age-matched typically developing participants. Analyzing the parameters of the models by means of discriminative classifiers, the dialogs of patients were successfully distinguished from those of control participants. We conclude that meaning-free speech/silence sequences, reflecting inter-actant synchronization, at least partially encode typical and atypical conversational dynamics. This suggests a direct influence of theory of mind abilities onto basic speech initiative behavior.

Opinion formation of free speech on the directed social network

NASA Astrophysics Data System (ADS)

Su, Jiongming; Ma, Hongxu; Liu, Baohong; Li, Qi

2014-12-01

A dynamical model with continuous opinion is proposed to study how the speech order and the topology of directed social network affect the opinion formation of free speech. In the model, agents express their opinions one by one with random order (RO) or probability order (PO), other agents paying attentions to the speaking agent, receive provider's opinion, update their opinions and then express their new opinions in their turns. It is proved that with the same agent j repeats its opinion more, other agents who pay their attentions to j and include j's opinion in their confidence level at initial time, will continue approaching j's opinion. Simulation results reveal that on directed scale-free network: (1) the model for PO forms fewer opinion clusters, larger maximum cluster (MC), smaller standard deviation (SD), and needs less waiting time to reach a middle level of consensus than RO; (2) as the parameter of scale-free degree distribution decreases or the confidence level increases, the results often get better for both speech orders; (3) the differences between PO and RO get smaller as the size of network decreases.

A comparison of recordings of sentences and spontaneous speech: perceptual and acoustic measures in preschool children's voices.

PubMed

McAllister, Anita; Brandt, Signe Kofoed

2012-09-01

A well-controlled recording in a studio is fundamental in most voice rehabilitation. However, this laboratory like recording method has been questioned because voice use in a natural environment may be quite different. In children's natural environment, high background noise levels are common and are an important factor contributing to voice problems. The primary noise source in day-care centers is the children themselves. The aim of the present study was to compare perceptual evaluations of voice quality and acoustic measures from a controlled recording with recordings of spontaneous speech in children's natural environment in a day-care setting. Eleven 5-year-old children were recorded three times during a day at the day care. The controlled speech material consisted of repeated sentences. Matching sentences were selected from the spontaneous speech. All sentences were repeated three times. Recordings were randomized and analyzed acoustically and perceptually. Statistic analyses showed that fundamental frequency was significantly higher in spontaneous speech (P<0.01) as was hyperfunction (P<0.001). The only characteristic the controlled sentences shared with spontaneous speech was degree of hoarseness (Spearman's rho=0.564). When data for boys and girls were analyzed separately, a correlation was found for the parameter breathiness (rho=0.551) for boys, and for girls the correlation for hoarseness remained (rho=0.752). Regarding acoustic data, none of the measures correlated across recording conditions for the whole group. Copyright © 2012 The Voice Foundation. Published by Mosby, Inc. All rights reserved.

Standardization of pitch-range settings in voice acoustic analysis.

PubMed

Vogel, Adam P; Maruff, Paul; Snyder, Peter J; Mundt, James C

2009-05-01

Voice acoustic analysis is typically a labor-intensive, time-consuming process that requires the application of idiosyncratic parameters tailored to individual aspects of the speech signal. Such processes limit the efficiency and utility of voice analysis in clinical practice as well as in applied research and development. In the present study, we analyzed 1,120 voice files, using standard techniques (case-by-case hand analysis), taking roughly 10 work weeks of personnel time to complete. The results were compared with the analytic output of several automated analysis scripts that made use of preset pitch-range parameters. After pitch windows were selected to appropriately account for sex differences, the automated analysis scripts reduced processing time of the 1,120 speech samples to less than 2.5 h and produced results comparable to those obtained with hand analysis. However, caution should be exercised when applying the suggested preset values to pathological voice populations.

Synthesis of Polysyllabic Sequences of Thai Tones Using a Generative Model of Fundamental Frequency Contours

NASA Astrophysics Data System (ADS)

Seresangtakul, Pusadee; Takara, Tomio

In this paper, the distinctive tones of Thai in running speech are studied. We present rules to synthesize F0 contours of Thai tones in running speech by using the generative model of F0 contours. Along with our method, the pitch contours of Thai polysyllabic words, both disyllabic and trisyllabic words, were analyzed. The coarticulation effect of Thai tones in running speech were found. Based on the analysis of the polysyllabic words using this model, rules are derived and applied to synthesize Thai polysyllabic tone sequences. We performed listening tests to evaluate intelligibility of the rules for Thai tones generation. The average intelligibility scores became 98.8%, and 96.6% for disyllabic and trisyllabic words, respectively. From these result, the rule of the tones' generation was shown to be effective. Furthermore, we constructed the connecting rules to synthesize suprasegmental F0 contours using the trisyllable training rules' parameters. The parameters of the first, the third, and the second syllables were selected and assigned to the initial, the ending, and the remaining syllables in a sentence, respectively. Even such a simple rule, the synthesized phrases/senetences were completely identified in listening tests. The MOSs (Mean Opinion Score) was 3.50 while the original and analysis/synthesis samples were 4.82 and 3.59, respectively.

Determining the importance of fundamental hearing aid attributes.

PubMed

Meister, Hartmut; Lausberg, Isabel; Kiessling, Juergen; Walger, Martin; von Wedel, Hasso

2002-07-01

To determine the importance of fundamental hearing aid attributes and to elicit measures of satisfaction and dissatisfaction. A prospective study based on a survey using a decompositional approach of preference measurement (conjoint analysis). Ear, nose, and throat university hospitals in Cologne and Giessen; various branches of hearing aid dispensers. A random sample of 175 experienced hearing aid users aged 20 to 91 years (mean age, 61 yr) recruited at two different sites. Relative importance of different hearing aid attributes, satisfaction and dissatisfaction with hearing aid attributes. Of the six fundamental hearing aid attributes assessed by the hearing aid users, the two features concerning speech perception attained the highest relative importance (25% speech in quiet, 27% speech in noise). The remaining four attributes (sound quality, handling, feedback, localization) had significantly lower values in a narrow range of 10 to 12%. Comparison of different subgroups of hearing aid wearers based on sociodemographic and user-specific data revealed a large interindividual scatter of the preferences for the attributes. A similar examination with 25 clinicians revealed overestimation of the importance of the attributes commonly associated with problems. Moreover, examination of satisfaction showed that speech in noise was the most frequent source of dissatisfaction (30% of all statements), whereas the subjects were satisfied with speech in quiet. The results emphasize the high importance of attributes related to speech perception. Speech discrimination in noise was the most important but also the most frequent source of negative statements. This attribute will be the outstanding parameter of future developments. Appropriate handling becomes an important factor for elderly subjects. However, because of the large interindividual scatter of data, the preferences of different hearing aid users were hardly predictable, giving evidence of multifactorial influences.

Comparison of the Spectral-Temporally Modulated Ripple Test With the Arizona Biomedical Institute Sentence Test in Cochlear Implant Users.

PubMed

Lawler, Marshall; Yu, Jeffrey; Aronoff, Justin M

Although speech perception is the gold standard for measuring cochlear implant (CI) users' performance, speech perception tests often require extensive adaptation to obtain accurate results, particularly after large changes in maps. Spectral ripple tests, which measure spectral resolution, are an alternate measure that has been shown to correlate with speech perception. A modified spectral ripple test, the spectral-temporally modulated ripple test (SMRT) has recently been developed, and the objective of this study was to compare speech perception and performance on the SMRT for a heterogeneous population of unilateral CI users, bilateral CI users, and bimodal users. Twenty-five CI users (eight using unilateral CIs, nine using bilateral CIs, and eight using a CI and a hearing aid) were tested on the Arizona Biomedical Institute Sentence Test (AzBio) with a +8 dB signal to noise ratio, and on the SMRT. All participants were tested with their clinical programs. There was a significant correlation between SMRT and AzBio performance. After a practice block, an improvement of one ripple per octave for SMRT corresponded to an improvement of 12.1% for AzBio. Additionally, there was no significant difference in slope or intercept between any of the CI populations. The results indicate that performance on the SMRT correlates with speech recognition in noise when measured across unilateral, bilateral, and bimodal CI populations. These results suggest that SMRT scores are strongly associated with speech recognition in noise ability in experienced CI users. Further studies should focus on increasing both the size and diversity of the tested participants, and on determining whether the SMRT technique can be used for early predictions of long-term speech scores, or for evaluating differences among different stimulation strategies or parameter settings.

A large-scale video codec comparison of x264, x265 and libvpx for practical VOD applications

NASA Astrophysics Data System (ADS)

De Cock, Jan; Mavlankar, Aditya; Moorthy, Anush; Aaron, Anne

2016-09-01

Over the last years, we have seen exciting improvements in video compression technology, due to the introduction of HEVC and royalty-free coding specifications such as VP9. The potential compression gains of HEVC over H.264/AVC have been demonstrated in different studies, and are usually based on the HM reference software. For VP9, substantial gains over H.264/AVC have been reported in some publications, whereas others reported less optimistic results. Differences in configurations between these publications make it more difficult to assess the true potential of VP9. Practical open-source encoder implementations such as x265 and libvpx (VP9) have matured, and are now showing high compression gains over x264. In this paper, we demonstrate the potential of these encoder imple- mentations, with settings optimized for non-real-time random access, as used in a video-on-demand encoding pipeline. We report results from a large-scale video codec comparison test, which includes x264, x265 and libvpx. A test set consisting of a variety of titles with varying spatio-temporal characteristics from our catalog is used, resulting in tens of millions of encoded frames, hence larger than test sets previously used in the literature. Re- sults are reported in terms of PSNR, SSIM, MS-SSIM, VIF and the recently introduced VMAF quality metric. BD-rate calculations show that using x265 and libvpx vs. x264 can lead to significant bitrate savings for the same quality. x265 outperforms libvpx in most cases, but the performance gap narrows (or even reverses) at the higher resolutions.

Addition of Kinesio Taping of the orbicularis oris muscles to speech therapy rapidly improves drooling in children with neurological disorders.

PubMed

Mikami, Denise Lica Yoshimura; Furia, Cristina Lemos Barbosa; Welker, Alexis Fonseca

2017-09-21

To evaluate the effects of Kinesio Taping (KT) of the orbicularis oris muscles as an adjunct to standard therapy for drooling. Fifteen children with neurological disorders and drooling received speech therapy and twice-weekly KT of the orbicularis muscles over a 30-day period. Drooling was assessed by six parameters: impact on the life of the child and caregiver; severity of drooling; frequency of drooling; drooling volume (estimated by number of bibs used); salivary leak; and interlabial gap. Seven markers of oral motor skills were also assessed. KT of the orbicularis oris region reduced the interlabial gap. All oral motor skills and almost all markers of drooling improved after 15 days of treatment. In this sample of children with neurological disorders, adding KT of the orbicularis oris muscles to speech therapy caused rapid improvement in oral motor skills and drooling.

Gender typicality in children's speech: A comparison of boys with and without gender identity disorder.

PubMed

Munson, Benjamin; Crocker, Laura; Pierrehumbert, Janet B; Owen-Anderson, Allison; Zucker, Kenneth J

2015-04-01

This study examined whether boys with gender identity disorder (GID) produced less prototypically male speech than control boys without GID, a possibility that has been suggested by clinical observations. Two groups of listeners participated in tasks where they rated the gender typicality of single words (group 1) or sentences (group 2) produced by 15 5-13 year old boys with GID and 15 age-matched boys without GID. Detailed acoustic analyses of the stimuli were also conducted. Boys with GID were rated as less boy-like than boys without GID. In the experiment using sentence stimuli, these group differences were larger than in the experiment using single-word stimuli. Listeners' ratings were predicted by a variety of acoustic parameters, including ones that differ between the two groups and ones that are stereotypically associated with adult men's and women's speech. Future research should examine how these variants are acquired.

A novel speech watermarking algorithm by line spectrum pair modification

NASA Astrophysics Data System (ADS)

Zhang, Qian; Yang, Senbin; Chen, Guang; Zhou, Jun

2011-10-01

To explore digital watermarking specifically suitable for the speech domain, this paper experimentally investigates the properties of line spectrum pair (LSP) parameters firstly. The results show that the differences between contiguous LSPs are robust against common signal processing operations and small modifications of LSPs are imperceptible to the human auditory system (HAS). According to these conclusions, three contiguous LSPs of a speech frame are selected to embed a watermark bit. The middle LSP is slightly altered to modify the differences of these LSPs when embedding watermark. Correspondingly, the watermark is extracted by comparing these differences. The proposed algorithm's transparency is adjustable to meet the needs of different applications. The algorithm has good robustness against additive noise, quantization, amplitude scale and MP3 compression attacks, for the bit error rate (BER) is less than 5%. In addition, the algorithm allows a relatively low capacity, which approximates to 50 bps.

[Phoneme analysis and phoneme discrimination of juvenile speech therapy school students].

PubMed

Franz, S; Rosanowski, F; Eysholdt, U; Hoppe, U

2011-05-01

Phoneme analysis and phoneme discrimination, important factors in acquiring spoken and written language, have been evaluated in juvenile speech therapy school students. The results have been correlated with the results of a school achievement test. The following questions were of interest: Do students in the lower verbal skill segment show pathological phoneme analysis and phoneme discrimination skills? Do the results of the school achievement test differ from the results by students visiting German "Hauptschule"? How does phoneme analysis and phoneme discrimination performance correlate to other tested parameters? 74 students of a speech therapy school ranging from 7 (th) to 9 (th) grade were examined (ages 12;10-17;04) with the Heidelberg Phoneme Discrimination Test H-LAD and the school achievement test "Prüfsystem für Schul- und Bildungsberatung PSB-R 6-13". Compared to 4 (th) graders the juvenile speech therapy school students showed worse results in the H-LAD test with good differentiation in the lower measuring range. In the PSB-R 6-13 test the examined students did worse compared to students visiting German "Hauptschule" for all grades except 9 (th) grade. Comparing H-LAD and PSB-R 6-13 shows a significant correlation for the sub-tests covering language competence and intelligence but not for the concentration tests. Pathological phoneme analysis and phoneme discrimination skills suggest elevated need for counseling, but this needs to corroborated through additional linguistic parameters and measuring non-verbal intelligence. Further trails are needed in order to clarify whether the results can lead to sophisticated therapy algorithms for educational purposes. © Georg Thieme Verlag KG Stuttgart · New York.

Using genetic algorithms with subjective input from human subjects: implications for fitting hearing aids and cochlear implants.

PubMed

Başkent, Deniz; Eiler, Cheryl L; Edwards, Brent

2007-06-01

To present a comprehensive analysis of the feasibility of genetic algorithms (GA) for finding the best fit of hearing aids or cochlear implants for individual users in clinical or research settings, where the algorithm is solely driven by subjective human input. Due to varying pathology, the best settings of an auditory device differ for each user. It is also likely that listening preferences vary at the same time. The settings of a device customized for a particular user can only be evaluated by the user. When optimization algorithms are used for fitting purposes, this situation poses a difficulty for a systematic and quantitative evaluation of the suitability of the fitting parameters produced by the algorithm. In the present study, an artificial listening environment was generated by distorting speech using a noiseband vocoder. The settings produced by the GA for this listening problem could objectively be evaluated by measuring speech recognition and comparing the performance to the best vocoder condition where speech was least distorted. Nine normal-hearing subjects participated in the study. The parameters to be optimized were the number of vocoder channels, the shift between the input frequency range and the synthesis frequency range, and the compression-expansion of the input frequency range over the synthesis frequency range. The subjects listened to pairs of sentences processed with the vocoder, and entered a preference for the sentence with better intelligibility. The GA modified the solutions iteratively according to the subject preferences. The program converged when the user ranked the same set of parameters as the best in three consecutive steps. The results produced by the GA were analyzed for quality by measuring speech intelligibility, for test-retest reliability by running the GA three times with each subject, and for convergence properties. Speech recognition scores averaged across subjects were similar for the best vocoder solution and for the solutions produced by the GA. The average number of iterations was 8 and the average convergence time was 25.5 minutes. The settings produced by different GA runs for the same subject were slightly different; however, speech recognition scores measured with these settings were similar. Individual data from subjects showed that in each run, a small number of GA solutions produced poorer speech intelligibility than for the best setting. This was probably a result of the combination of the inherent randomness of the GA, the convergence criterion used in the present study, and possible errors that the users might have made during the paired comparisons. On the other hand, the effect of these errors was probably small compared to the other two factors, as a comparison between subjective preferences and objective measures showed that for many subjects the two were in good agreement. The results showed that the GA was able to produce good solutions by using listener preferences in a relatively short time. For practical applications, the program can be made more robust by running the GA twice or by not using an automatic stopping criterion, and it can be made faster by optimizing the number of the paired comparisons completed in each iteration.

Sensory-Motor Networks Involved in Speech Production and Motor Control: An fMRI Study

PubMed Central

Behroozmand, Roozbeh; Shebek, Rachel; Hansen, Daniel R.; Oya, Hiroyuki; Robin, Donald A.; Howard, Matthew A.; Greenlee, Jeremy D.W.

2015-01-01

Speaking is one of the most complex motor behaviors developed to facilitate human communication. The underlying neural mechanisms of speech involve sensory-motor interactions that incorporate feedback information for online monitoring and control of produced speech sounds. In the present study, we adopted an auditory feedback pitch perturbation paradigm and combined it with functional magnetic resonance imaging (fMRI) recordings in order to identify brain areas involved in speech production and motor control. Subjects underwent fMRI scanning while they produced a steady vowel sound /a/ (speaking) or listened to the playback of their own vowel production (playback). During each condition, the auditory feedback from vowel production was either normal (no perturbation) or perturbed by an upward (+600 cents) pitch shift stimulus randomly. Analysis of BOLD responses during speaking (with and without shift) vs. rest revealed activation of a complex network including bilateral superior temporal gyrus (STG), Heschl's gyrus, precentral gyrus, supplementary motor area (SMA), Rolandic operculum, postcentral gyrus and right inferior frontal gyrus (IFG). Performance correlation analysis showed that the subjects produced compensatory vocal responses that significantly correlated with BOLD response increases in bilateral STG and left precentral gyrus. However, during playback, the activation network was limited to cortical auditory areas including bilateral STG and Heschl's gyrus. Moreover, the contrast between speaking vs. playback highlighted a distinct functional network that included bilateral precentral gyrus, SMA, IFG, postcentral gyrus and insula. These findings suggest that speech motor control involves feedback error detection in sensory (e.g. auditory) cortices that subsequently activate motor-related areas for the adjustment of speech parameters during speaking. PMID:25623499

Acoustic Constraints and Musical Consequences: Exploring Composers' Use of Cues for Musical Emotion

PubMed Central

Schutz, Michael

2017-01-01

Emotional communication in music is based in part on the use of pitch and timing, two cues effective in emotional speech. Corpus analyses of natural speech illustrate that happy utterances tend to be higher and faster than sad. Although manipulations altering melodies show that passages changed to be higher and faster sound happier, corpus analyses of unaltered music paralleling those of natural speech have proven challenging. This partly reflects the importance of modality (i.e., major/minor), a powerful musical cue whose use is decidedly imbalanced in Western music. This imbalance poses challenges for creating musical corpora analogous to existing speech corpora for purposes of analyzing emotion. However, a novel examination of music by Bach and Chopin balanced in modality illustrates that, consistent with predictions from speech, their major key (nominally “happy”) pieces are approximately a major second higher and 29% faster than their minor key pieces (Poon and Schutz, 2015). Although this provides useful evidence for parallels in use of emotional cues between these domains, it raises questions about how composers “trade off” cue differentiation in music, suggesting interesting new potential research directions. This Focused Review places those results in a broader context, highlighting their connections with previous work on the natural use of cues for musical emotion. Together, these observational findings based on unaltered music—widely recognized for its artistic significance—complement previous experimental work systematically manipulating specific parameters. In doing so, they also provide a useful musical counterpart to fruitful studies of the acoustic cues for emotion found in natural speech. PMID:29249997

Test–retest repeatability of human speech biomarkers from static and real-time dynamic magnetic resonance imaging

PubMed Central

Töger, Johannes; Sorensen, Tanner; Somandepalli, Krishna; Toutios, Asterios; Lingala, Sajan Goud; Narayanan, Shrikanth; Nayak, Krishna

2017-01-01

Static anatomical and real-time dynamic magnetic resonance imaging (RT-MRI) of the upper airway is a valuable method for studying speech production in research and clinical settings. The test–retest repeatability of quantitative imaging biomarkers is an important parameter, since it limits the effect sizes and intragroup differences that can be studied. Therefore, this study aims to present a framework for determining the test–retest repeatability of quantitative speech biomarkers from static MRI and RT-MRI, and apply the framework to healthy volunteers. Subjects (n = 8, 4 females, 4 males) are imaged in two scans on the same day, including static images and dynamic RT-MRI of speech tasks. The inter-study agreement is quantified using intraclass correlation coefficient (ICC) and mean within-subject standard deviation (σe). Inter-study agreement is strong to very strong for static measures (ICC: min/median/max 0.71/0.89/0.98, σe: 0.90/2.20/6.72 mm), poor to strong for dynamic RT-MRI measures of articulator motion range (ICC: 0.26/0.75/0.90, σe: 1.6/2.5/3.6 mm), and poor to very strong for velocities (ICC: 0.21/0.56/0.93, σe: 2.2/4.4/16.7 cm/s). In conclusion, this study characterizes repeatability of static and dynamic MRI-derived speech biomarkers using state-of-the-art imaging. The introduced framework can be used to guide future development of speech biomarkers. Test–retest MRI data are provided free for research use. PMID:28599561

Acoustic Constraints and Musical Consequences: Exploring Composers' Use of Cues for Musical Emotion.

PubMed

Schutz, Michael

2017-01-01

Emotional communication in music is based in part on the use of pitch and timing, two cues effective in emotional speech. Corpus analyses of natural speech illustrate that happy utterances tend to be higher and faster than sad. Although manipulations altering melodies show that passages changed to be higher and faster sound happier, corpus analyses of unaltered music paralleling those of natural speech have proven challenging. This partly reflects the importance of modality (i.e., major/minor), a powerful musical cue whose use is decidedly imbalanced in Western music. This imbalance poses challenges for creating musical corpora analogous to existing speech corpora for purposes of analyzing emotion. However, a novel examination of music by Bach and Chopin balanced in modality illustrates that, consistent with predictions from speech, their major key (nominally "happy") pieces are approximately a major second higher and 29% faster than their minor key pieces (Poon and Schutz, 2015). Although this provides useful evidence for parallels in use of emotional cues between these domains, it raises questions about how composers "trade off" cue differentiation in music, suggesting interesting new potential research directions. This Focused Review places those results in a broader context, highlighting their connections with previous work on the natural use of cues for musical emotion. Together, these observational findings based on unaltered music-widely recognized for its artistic significance-complement previous experimental work systematically manipulating specific parameters. In doing so, they also provide a useful musical counterpart to fruitful studies of the acoustic cues for emotion found in natural speech.

A low-power and high-quality implementation of the discrete cosine transformation

NASA Astrophysics Data System (ADS)

Heyne, B.; Götze, J.

2007-06-01

In this paper a computationally efficient and high-quality preserving DCT architecture is presented. It is obtained by optimizing the Loeffler DCT based on the Cordic algorithm. The computational complexity is reduced from 11 multiply and 29 add operations (Loeffler DCT) to 38 add and 16 shift operations (which is similar to the complexity of the binDCT). The experimental results show that the proposed DCT algorithm not only reduces the computational complexity significantly, but also retains the good transformation quality of the Loeffler DCT. Therefore, the proposed Cordic based Loeffler DCT is especially suited for low-power and high-quality CODECs in battery-based systems.

Binaural sluggishness in the perception of tone sequences and speech in noise.

PubMed

Culling, J F; Colburn, H S

2000-01-01

The binaural system is well-known for its sluggish response to changes in the interaural parameters to which it is sensitive. Theories of binaural unmasking have suggested that detection of signals in noise is mediated by detection of differences in interaural correlation. If these theories are correct, improvements in the intelligibility of speech in favorable binaural conditions is most likely mediated by spectro-temporal variations in interaural correlation of the stimulus which mirror the spectro-temporal amplitude modulations of the speech. However, binaural sluggishness should limit the temporal resolution of the representation of speech recovered by this means. The present study tested this prediction in two ways. First, listeners' masked discrimination thresholds for ascending vs descending pure-tone arpeggios were measured as a function of rate of frequency change in the NoSo and NoSpi binaural configurations. Three-tone arpeggios were presented repeatedly and continuously for 1.6 s, masked by a 1.6-s burst of noise. In a two-interval task, listeners determined the interval in which the arpeggios were ascending. The results showed a binaural advantage of 12-14 dB for NoSpi at 3.3 arpeggios per s (arp/s), which reduced to 3-5 dB at 10.4 arp/s. This outcome confirmed that the discrimination of spectro-temporal patterns in noise is susceptible to the effects of binaural sluggishness. Second, listeners' masked speech-reception thresholds were measured in speech-shaped noise using speech which was 1, 1.5, and 2 times the original articulation rate. The articulation rate was increased using a phase-vocoder technique which increased all the modulation frequencies in the speech without altering its pitch. Speech-reception thresholds were, on average, 5.2 dB lower for the NoSpi than for the NoSo configuration, at the original articulation rate. This binaural masking release was reduced to 2.8 dB when the articulation rate was doubled, but the most notable effect was a 6-8 dB increase in thresholds with articulation rate for both configurations. These results suggest that higher modulation frequencies in masked signals cannot be temporally resolved by the binaural system, but that the useful modulation frequencies in speech are sufficiently low (<5 Hz) that they are invulnerable to the effects of binaural sluggishness, even at elevated articulation rates.

Soundscape elaboration from anthrophonic adaptation of community noise

NASA Astrophysics Data System (ADS)

Teddy Badai Samodra, FX

2018-03-01

Under the situation of an urban environment, noise has been a critical issue in affecting the indoor environment. A reliable approach is required for evaluation of the community noise as one factor of anthrophonic in the urban environment. This research investigates the level of noise exposure from different community noise sources and elaborates the advantage of the noise disadvantages for soundscape innovation. Integrated building element design as a protector for noise control and speech intelligibility compliance using field experiment and MATLAB programming and modeling are also carried out. Meanwhile, for simulation analysis and building acoustic optimization, Sound Reduction-Speech Intelligibility and Reverberation Time are the main parameters for identifying tropical building model as case study object. The results show that the noise control should consider its integration with the other critical issue, thermal control, in an urban environment. The 1.1 second of reverberation time for speech activities and noise reduction more than 28.66 dBA for critical frequency (20 Hz), the speech intelligibility index could be reached more than fair assessment, 0.45. Furthermore, the environmental psychology adaptation result “Close The Opening” as the best method in high noise condition and personal adjustment as the easiest and the most adaptable way.

Production and perception of whispered vowels

NASA Astrophysics Data System (ADS)

Kiefte, Michael

2005-09-01

Information normally associated with pitch, such as intonation, can still be conveyed in whispered speech despite the absence of voicing. For example, it is possible to whisper the question ``You are going today?'' without any syntactic information to distinguish this sentence from a simple declarative. It has been shown that pitch change in whispered speech is correlated with the simultaneous raising or lowering of several formants [e.g., M. Kiefte, J. Acoust. Soc. Am. 116, 2546 (2004)]. However, spectral peak frequencies associated with formants have been identified as important correlates to vowel identity. Spectral peak frequencies may serve two roles in the perception of whispered speech: to indicate both vowel identity and intended pitch. Data will be presented to examine the relative importance of several acoustic properties including spectral peak frequencies and spectral shape parameters in both the production and perception of whispered vowels. Speakers were asked to phonate and whisper vowels at three different pitches across a range of roughly a musical fifth. It will be shown that relative spectral change is preserved within vowels across intended pitches in whispered speech. In addition, several models of vowel identification by listeners will be presented. [Work supported by SSHRC.

Automatic Conversational Scene Analysis in Children with Asperger Syndrome/High-Functioning Autism and Typically Developing Peers

PubMed Central

Tavano, Alessandro; Pesarin, Anna; Murino, Vittorio; Cristani, Marco

2014-01-01

Individuals with Asperger syndrome/High Functioning Autism fail to spontaneously attribute mental states to the self and others, a life-long phenotypic characteristic known as mindblindness. We hypothesized that mindblindness would affect the dynamics of conversational interaction. Using generative models, in particular Gaussian mixture models and observed influence models, conversations were coded as interacting Markov processes, operating on novel speech/silence patterns, termed Steady Conversational Periods (SCPs). SCPs assume that whenever an agent's process changes state (e.g., from silence to speech), it causes a general transition of the entire conversational process, forcing inter-actant synchronization. SCPs fed into observed influence models, which captured the conversational dynamics of children and adolescents with Asperger syndrome/High Functioning Autism, and age-matched typically developing participants. Analyzing the parameters of the models by means of discriminative classifiers, the dialogs of patients were successfully distinguished from those of control participants. We conclude that meaning-free speech/silence sequences, reflecting inter-actant synchronization, at least partially encode typical and atypical conversational dynamics. This suggests a direct influence of theory of mind abilities onto basic speech initiative behavior. PMID:24489674

Florida Journal of Communication Disorders, 1995.

ERIC Educational Resources Information Center

Langhans, Joseph J., Ed.

1995-01-01

This annual volume is a compilation of articles addressing evaluation, management, professional affairs, practice parameters, and clinical application of speech and language services. Featured articles include: (1) "Comparison of Pressure, Flow, and Resistance for Modal and Loft Register Productions" (Joseph L. Langhans and Peter J.…

Recognition of speaker-dependent continuous speech with KEAL

NASA Astrophysics Data System (ADS)

Mercier, G.; Bigorgne, D.; Miclet, L.; Le Guennec, L.; Querre, M.

1989-04-01

A description of the speaker-dependent continuous speech recognition system KEAL is given. An unknown utterance, is recognized by means of the followng procedures: acoustic analysis, phonetic segmentation and identification, word and sentence analysis. The combination of feature-based, speaker-independent coarse phonetic segmentation with speaker-dependent statistical classification techniques is one of the main design features of the acoustic-phonetic decoder. The lexical access component is essentially based on a statistical dynamic programming technique which aims at matching a phonemic lexical entry containing various phonological forms, against a phonetic lattice. Sentence recognition is achieved by use of a context-free grammar and a parsing algorithm derived from Earley's parser. A speaker adaptation module allows some of the system parameters to be adjusted by matching known utterances with their acoustical representation. The task to be performed, described by its vocabulary and its grammar, is given as a parameter of the system. Continuously spoken sentences extracted from a 'pseudo-Logo' language are analyzed and results are presented.

Sensory-motor networks involved in speech production and motor control: an fMRI study.

PubMed

Behroozmand, Roozbeh; Shebek, Rachel; Hansen, Daniel R; Oya, Hiroyuki; Robin, Donald A; Howard, Matthew A; Greenlee, Jeremy D W

2015-04-01

Speaking is one of the most complex motor behaviors developed to facilitate human communication. The underlying neural mechanisms of speech involve sensory-motor interactions that incorporate feedback information for online monitoring and control of produced speech sounds. In the present study, we adopted an auditory feedback pitch perturbation paradigm and combined it with functional magnetic resonance imaging (fMRI) recordings in order to identify brain areas involved in speech production and motor control. Subjects underwent fMRI scanning while they produced a steady vowel sound /a/ (speaking) or listened to the playback of their own vowel production (playback). During each condition, the auditory feedback from vowel production was either normal (no perturbation) or perturbed by an upward (+600 cents) pitch-shift stimulus randomly. Analysis of BOLD responses during speaking (with and without shift) vs. rest revealed activation of a complex network including bilateral superior temporal gyrus (STG), Heschl's gyrus, precentral gyrus, supplementary motor area (SMA), Rolandic operculum, postcentral gyrus and right inferior frontal gyrus (IFG). Performance correlation analysis showed that the subjects produced compensatory vocal responses that significantly correlated with BOLD response increases in bilateral STG and left precentral gyrus. However, during playback, the activation network was limited to cortical auditory areas including bilateral STG and Heschl's gyrus. Moreover, the contrast between speaking vs. playback highlighted a distinct functional network that included bilateral precentral gyrus, SMA, IFG, postcentral gyrus and insula. These findings suggest that speech motor control involves feedback error detection in sensory (e.g. auditory) cortices that subsequently activate motor-related areas for the adjustment of speech parameters during speaking. Copyright © 2015 Elsevier Inc. All rights reserved.

Dual-Layer Video Encryption using RSA Algorithm

NASA Astrophysics Data System (ADS)

Chadha, Aman; Mallik, Sushmit; Chadha, Ankit; Johar, Ravdeep; Mani Roja, M.

2015-04-01

This paper proposes a video encryption algorithm using RSA and Pseudo Noise (PN) sequence, aimed at applications requiring sensitive video information transfers. The system is primarily designed to work with files encoded using the Audio Video Interleaved (AVI) codec, although it can be easily ported for use with Moving Picture Experts Group (MPEG) encoded files. The audio and video components of the source separately undergo two layers of encryption to ensure a reasonable level of security. Encryption of the video component involves applying the RSA algorithm followed by the PN-based encryption. Similarly, the audio component is first encrypted using PN and further subjected to encryption using the Discrete Cosine Transform. Combining these techniques, an efficient system, invulnerable to security breaches and attacks with favorable values of parameters such as encryption/decryption speed, encryption/decryption ratio and visual degradation; has been put forth. For applications requiring encryption of sensitive data wherein stringent security requirements are of prime concern, the system is found to yield negligible similarities in visual perception between the original and the encrypted video sequence. For applications wherein visual similarity is not of major concern, we limit the encryption task to a single level of encryption which is accomplished by using RSA, thereby quickening the encryption process. Although some similarity between the original and encrypted video is observed in this case, it is not enough to comprehend the happenings in the video.

Is blunted cardiovascular reactivity in depression mood-state dependent? A comparison of major depressive disorder remitted depression and healthy controls.

PubMed

Salomon, Kristen; Bylsma, Lauren M; White, Kristi E; Panaite, Vanessa; Rottenberg, Jonathan

2013-10-01

Prior work has repeatedly demonstrated that people who have current major depression exhibit blunted cardiovascular reactivity to acute stressors (e.g., Salomon et al., 2009). A key question regards the psychobiological basis for these deficits, including whether such deficits are depressed mood-state dependent or whether these effects are trait-like and are observed outside of depression episodes in vulnerable individuals. To examine this issue, we assessed cardiovascular reactivity to a speech stressor task and a forehead cold pressor in 50 individuals with current major depressive disorder (MDD), 25 with remitted major depression (RMD), and 45 healthy controls. Heart rate (HR), blood pressure and impedance cardiography were assessed and analyses controlled for BMI and sex. Significant group effects were found for SBP, HR, and PEP for the speech preparation period and HR, CO, and PEP during the speech. For each of these parameters, only the MDD group exhibited attenuated reactivity as well as impaired SBP recovery. Reactivity and recovery in the RMD group more closely resembled the healthy controls. Speeches given by the MDD group were rated as less persuasive than the RMD or healthy controls' speeches. No significant differences were found for the cold pressor. Blunted cardiovascular reactivity and impaired recovery in current major depression may be mood-state dependent phenomena and may be more reflective of motivational deficits than deficits in the physiological integrity of the cardiovascular system. Copyright © 2013 Elsevier B.V. All rights reserved.

Spectral and temporal resolutions of information-bearing acoustic changes for understanding vocoded sentencesa)

PubMed Central

Stilp, Christian E.; Goupell, Matthew J.

2015-01-01

Short-time spectral changes in the speech signal are important for understanding noise-vocoded sentences. These information-bearing acoustic changes, measured using cochlea-scaled entropy in cochlear implant simulations [CSECI; Stilp et al. (2013). J. Acoust. Soc. Am. 133(2), EL136–EL141; Stilp (2014). J. Acoust. Soc. Am. 135(3), 1518–1529], may offer better understanding of speech perception by cochlear implant (CI) users. However, perceptual importance of CSECI for normal-hearing listeners was tested at only one spectral resolution and one temporal resolution, limiting generalizability of results to CI users. Here, experiments investigated the importance of these informational changes for understanding noise-vocoded sentences at different spectral resolutions (4–24 spectral channels; Experiment 1), temporal resolutions (4–64 Hz cutoff for low-pass filters that extracted amplitude envelopes; Experiment 2), or when both parameters varied (6–12 channels, 8–32 Hz; Experiment 3). Sentence intelligibility was reduced more by replacing high-CSECI intervals with noise than replacing low-CSECI intervals, but only when sentences had sufficient spectral and/or temporal resolution. High-CSECI intervals were more important for speech understanding as spectral resolution worsened and temporal resolution improved. Trade-offs between CSECI and intermediate spectral and temporal resolutions were minimal. These results suggest that signal processing strategies that emphasize information-bearing acoustic changes in speech may improve speech perception for CI users. PMID:25698018

Is blunted cardiovascular reactivity in depression mood-state dependent? A comparison of major depressive disorder remitted depression and healthy controls

PubMed Central

Salomon, Kristen; Bylsma, Lauren M.; White, Kristi E.; Panaite, Vanessa; Rottenberg, Jonathan

2015-01-01

Prior work has repeatedly demonstrated that people who have current major depression exhibit blunted cardiovascular reactivity to acute stressors (e.g., Salomon et al., 2009). A key question regards the psychobiological basis for these deficits, including whether such deficits are depressed mood-state dependent or whether these effects are trait-like and are observed outside of depression episodes invulnerable individuals. To examine this issue, we assessed cardiovascular reactivity to a speech stressor task and a forehead cold pressor in 50 individuals with current major depressive disorder (MDD), 25 with remitted major depression (RMD), and 45 healthy controls. Heart rate (HR), blood pressure and impedance cardiography were assessed and analyses controlled for BMI and sex. Significant group effects were found for SBP, HR, and PEP for the speech preparation period and HR, CO, and PEP during the speech. For each of these parameters, only the MDD group exhibited attenuated reactivity as well as impaired SBP recovery. Reactivity and recovery in the RMD group more closely resembled the healthy controls. Speeches given by the MDD group were rated as less persuasive than the RMD or healthy controls' speeches. No significant differences were found for the cold pressor. Blunted cardiovascular reactivity and impaired recovery in current major depression may be mood-state dependent phenomena and may be more reflective of motivational deficits than deficits in the physiological integrity of the cardiovascular system. PMID:23756147

The Influence of Native Language on Auditory-Perceptual Evaluation of Vocal Samples Completed by Brazilian and Canadian SLPs.

PubMed

Chaves, Cristiane Ribeiro; Campbell, Melanie; Côrtes Gama, Ana Cristina

2017-03-01

This study aimed to determine the influence of native language on the auditory-perceptual assessment of voice, as completed by Brazilian and Anglo-Canadian listeners using Brazilian vocal samples and the grade, roughness, breathiness, asthenia, strain (GRBAS) scale. This is an analytical, observational, comparative, and transversal study conducted at the Speech Language Pathology Department of the Federal University of Minas Gerais in Brazil, and at the Communication Sciences and Disorders Department of the University of Alberta in Canada. The GRBAS scale, connected speech, and a sustained vowel were used in this study. The vocal samples were drawn randomly from a database of recorded speech of Brazilian adults, some with healthy voices and some with voice disorders. The database is housed at the Federal University of Minas Gerais. Forty-six samples of connected speech (recitation of days of the week), produced by 35 women and 11 men, and 46 samples of the sustained vowel /a/, produced by 37 women and 9 men, were used in this study. The listeners were divided into two groups of three speech therapists, according to nationality: Brazilian or Anglo-Canadian. The groups were matched according to the years of professional experience of participants. The weighted kappa was used to calculate the intra- and inter-rater agreements, with 95% confidence intervals, respectively. An analysis of the intra-rater agreement showed that Brazilians and Canadians had similar results in auditory-perceptual evaluation of sustained vowel and connected speech. The results of the inter-rater agreement of connected speech and sustained vowel indicated that Brazilians and Canadians had, respectively, moderate agreement on the overall severity (0.57 and 0.50), breathiness (0.45 and 0.45), and asthenia (0.50 and 0.46); poor correlation on roughness (0.19 and 0.007); and weak correlation on strain to connected speech (0.22), and moderate correlation to sustained vowel (0.50). In general, auditory-perceptual evaluation is not influenced by the native language on most dimensions of the perceptual parameters of the GRBAS scale. Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

Interlabial contact pressures exhibited in dysarthria following traumatic brain injury during speech and nonspeech tasks.

PubMed

Goozée, Justine V; Murdoch, Bruce E; Theodoros, Deborah G

2002-01-01

A miniature pressure transducer was used to assess the interlabial contact pressures produced by a group of 19 adults (mean age 30.6 years) with dysarthria following severe traumatic brain injury (TBI) during a set of speech and nonspeech tasks. Ten parameters relating to lip strength, endurance, rate of movement and lip pressure accuracy and stability were measured from the nonspeech tasks. The results attained by the TBI group were compared against a group of 19 age- and sex-matched control subjects. Significant differences between the groups were found for maximum interlabial contact pressure, maximum rate of repetition of maximum pressure, and lip pressure accuracy at 50 and 10% levels of maximum pressure. In regards to speech, the interlabial contact pressures generated by the TBI group and control group did not differ significantly. When expressed as percentages of maximum pressure, however, the TBI group's interlabial pressures appeared to have been generated with greater physiological effort. Copyright 2002 S. Karger AG, Basel

The influence of otitis media with effusion on speech and language development and psycho-intellectual behaviour of the preschool child--results of a cross-sectional study in 1,512 children.

PubMed

Van Cauwenberge, P; Van Cauwenberge, K; Kluyskens, P

1985-01-01

To investigate the influence of otitis media with effusion (OME) on the psychological, social and intellectual development of preschool children, a cross-sectional study in 1,512 apparently healthy children, aged 25-80 months, attending kindergarten (infant school) was performed. Tympanometry and evaluation of the various psychological, social and intellectual parameters by the infant school teacher (assisted by a sociologist) were the most important diagnostical tools in this study. It was demonstrated that OME had a negative influence on speech/language development, intelligence, attention at school, activity at school, manual skill and social behaviour of the 2 to 6 year old child. For speech and language the negative influence was most clearly demonstrated in the youngest age group (less than 47 months), for intelligence and activity in the older age groups. Early detection and appropriate treatment of OME are recommended to avoid these complications.

Got EQ?: Increasing Cultural and Clinical Competence through Emotional Intelligence

ERIC Educational Resources Information Center

Robertson, Shari A.

2007-01-01

Cultural intelligence has been described across three parameters of human behavior: cognitive intelligence, emotional intelligence (EQ), and physical intelligence. Each contributes a unique and important perspective to the ability of speech-language pathologists and audiologists to provide benefits to their clients regardless of cultural…

A Research Program in Computer Technology

DTIC Science & Technology

1976-07-01

K PROGRAM VERIFICATION 12 [Shaw76b] Shaw, M., W. A. Wulf, and R. L. London, Abstraction and Verification ain Aiphard: Iteration and Generators...millisecond trame of speech: pitch, gain, and 10 k -parameters (often called reflection coefficients). The 12 parameters from each frame are encoded into...del rey, CA 90291 Program Code 3D30 & 3P1O I,%’POLLING OFFICE NAME AND ADDRESS 12 REPORT DATE Defense Advanced Research Projects Agency July 1976 1400

[Relationships between electrophysiological characteristic of speech evoked auditory brainstem response and Mandarin monosyllable discriminative ability at different hearing impairment].

PubMed

Fu, Q Y; Liang, Y; Zou, A; Wang, T; Zhao, X D; Wan, J

2016-04-07

To investigate the relationships between electrophysiological characteristic of speech evoked auditory brainstem response(s-ABR) and Mandarin phonetically balanced maximum(PBmax) at different hearing impairment, so as to provide more clues for the mechanism of speech cognitive behavior. Forty-one ears in 41 normal hearing adults(NH), thirty ears in 30 conductive hearing loss patients(CHL) and twenty-seven ears in 27 sensorineural hearing loss patients(SNHL) were included in present study. The speech discrimination scores were obtained by Mandarin phonemic-balanced monosyllable lists via speech audiometric software. Their s-ABRs were recorded with speech syllables /da/ with the intensity of phonetically balanced maximum(PBmax). The electrophysiological characteristic of s-ABR, as well as the relationships between PBmax and s-ABR parameters including latency in time domain, fundamental frequency(F0) and first formant(F1) in frequency domain were analyzed statistically. All subjects completed good speech perception tests and PBmax of CHL and SNHL had no significant difference (P>0.05), but both significantly less than that of NH (P<0.05). While divided the subjects into three groups by 90%

Speaker verification system using acoustic data and non-acoustic data

DOEpatents

Gable, Todd J [Walnut Creek, CA; Ng, Lawrence C [Danville, CA; Holzrichter, John F [Berkeley, CA; Burnett, Greg C [Livermore, CA

2006-03-21

A method and system for speech characterization. One embodiment includes a method for speaker verification which includes collecting data from a speaker, wherein the data comprises acoustic data and non-acoustic data. The data is used to generate a template that includes a first set of "template" parameters. The method further includes receiving a real-time identity claim from a claimant, and using acoustic data and non-acoustic data from the identity claim to generate a second set of parameters. The method further includes comparing the first set of parameters to the set of parameters to determine whether the claimant is the speaker. The first set of parameters and the second set of parameters include at least one purely non-acoustic parameter, including a non-acoustic glottal shape parameter derived from averaging multiple glottal cycle waveforms.

Realization of guitar audio effects using methods of digital signal processing

NASA Astrophysics Data System (ADS)

Buś, Szymon; Jedrzejewski, Konrad

2015-09-01

The paper is devoted to studies on possibilities of realization of guitar audio effects by means of methods of digital signal processing. As a result of research, some selected audio effects corresponding to the specifics of guitar sound were realized as the real-time system called Digital Guitar Multi-effect. Before implementation in the system, the selected effects were investigated using the dedicated application with a graphical user interface created in Matlab environment. In the second stage, the real-time system based on a microcontroller and an audio codec was designed and realized. The system is designed to perform audio effects on the output signal of an electric guitar.

Robust Transmission of H.264/AVC Streams Using Adaptive Group Slicing and Unequal Error Protection

NASA Astrophysics Data System (ADS)

Thomos, Nikolaos; Argyropoulos, Savvas; Boulgouris, Nikolaos V.; Strintzis, Michael G.

2006-12-01

We present a novel scheme for the transmission of H.264/AVC video streams over lossy packet networks. The proposed scheme exploits the error-resilient features of H.264/AVC codec and employs Reed-Solomon codes to protect effectively the streams. A novel technique for adaptive classification of macroblocks into three slice groups is also proposed. The optimal classification of macroblocks and the optimal channel rate allocation are achieved by iterating two interdependent steps. Dynamic programming techniques are used for the channel rate allocation process in order to reduce complexity. Simulations clearly demonstrate the superiority of the proposed method over other recent algorithms for transmission of H.264/AVC streams.

Multi-modal highlight generation for sports videos using an information-theoretic excitability measure

NASA Astrophysics Data System (ADS)

Hasan, Taufiq; Bořil, Hynek; Sangwan, Abhijeet; L Hansen, John H.

2013-12-01

The ability to detect and organize `hot spots' representing areas of excitement within video streams is a challenging research problem when techniques rely exclusively on video content. A generic method for sports video highlight selection is presented in this study which leverages both video/image structure as well as audio/speech properties. Processing begins where the video is partitioned into small segments and several multi-modal features are extracted from each segment. Excitability is computed based on the likelihood of the segmental features residing in certain regions of their joint probability density function space which are considered both exciting and rare. The proposed measure is used to rank order the partitioned segments to compress the overall video sequence and produce a contiguous set of highlights. Experiments are performed on baseball videos based on signal processing advancements for excitement assessment in the commentators' speech, audio energy, slow motion replay, scene cut density, and motion activity as features. Detailed analysis on correlation between user excitability and various speech production parameters is conducted and an effective scheme is designed to estimate the excitement level of commentator's speech from the sports videos. Subjective evaluation of excitability and ranking of video segments demonstrate a higher correlation with the proposed measure compared to well-established techniques indicating the effectiveness of the overall approach.

Invariant principles of speech motor control that are not language-specific.

PubMed

Chakraborty, Rahul

2012-12-01

Bilingual speakers must learn to modify their speech motor control mechanism based on the linguistic parameters and rules specified by the target language. This study examines if there are aspects of speech motor control which remain invariant regardless of the first (L1) and second (L2) language targets. Based on the age of academic exposure and proficiency in L2, 21 Bengali-English bilingual participants were classified into high (n = 11) and low (n = 10) L2 (English) proficiency groups. Using the Optotrak 3020 motion sensitive camera system, the lips and jaw movements were recorded while participants produced Bengali (L1) and English (L2) sentences. Based on kinematic analyses of the lip and jaw movements, two different variability measures (i.e., lip aperture and lower lip/jaw complex) were computed for English and Bengali sentences. Analyses demonstrated that the two groups of bilingual speakers produced lip aperture complexes (a higher order synergy) that were more consistent in co-ordination than were the lower lip/jaw complexes (a lower order synergy). Similar findings were reported earlier in monolingual English speakers by Smith and Zelaznik. Thus, this hierarchical organization may be viewed as a fundamental principle of speech motor control, since it is maintained even in bilingual speakers.

Influence of auditory attention on sentence recognition captured by the neural phase.

PubMed

Müller, Jana Annina; Kollmeier, Birger; Debener, Stefan; Brand, Thomas

2018-03-07

The aim of this study was to investigate whether attentional influences on speech recognition are reflected in the neural phase entrained by an external modulator. Sentences were presented in 7 Hz sinusoidally modulated noise while the neural response to that modulation frequency was monitored by electroencephalogram (EEG) recordings in 21 participants. We implemented a selective attention paradigm including three different attention conditions while keeping physical stimulus parameters constant. The participants' task was either to repeat the sentence as accurately as possible (speech recognition task), to count the number of decrements implemented in modulated noise (decrement detection task), or to do both (dual task), while the EEG was recorded. Behavioural analysis revealed reduced performance in the dual task condition for decrement detection, possibly reflecting limited cognitive resources. EEG analysis revealed no significant differences in power for the 7 Hz modulation frequency, but an attention-dependent phase difference between tasks. Further phase analysis revealed a significant difference 500 ms after sentence onset between trials with correct and incorrect responses for speech recognition, indicating that speech recognition performance and the neural phase are linked via selective attention mechanisms, at least shortly after sentence onset. However, the neural phase effects identified were small and await further investigation. © 2018 Federation of European Neuroscience Societies and John Wiley & Sons Ltd.

Gender differences in identifying emotions from auditory and visual stimuli.

PubMed

Waaramaa, Teija

2017-12-01

The present study focused on gender differences in emotion identification from auditory and visual stimuli produced by two male and two female actors. Differences in emotion identification from nonsense samples, language samples and prolonged vowels were investigated. It was also studied whether auditory stimuli can convey the emotional content of speech without visual stimuli, and whether visual stimuli can convey the emotional content of speech without auditory stimuli. The aim was to get a better knowledge of vocal attributes and a more holistic understanding of the nonverbal communication of emotion. Females tended to be more accurate in emotion identification than males. Voice quality parameters played a role in emotion identification in both genders. The emotional content of the samples was best conveyed by nonsense sentences, better than by prolonged vowels or shared native language of the speakers and participants. Thus, vocal non-verbal communication tends to affect the interpretation of emotion even in the absence of language. The emotional stimuli were better recognized from visual stimuli than auditory stimuli by both genders. Visual information about speech may not be connected to the language; instead, it may be based on the human ability to understand the kinetic movements in speech production more readily than the characteristics of the acoustic cues.

Dysarthria of Motor Neuron Disease: Clinician Judgments of Severity.

ERIC Educational Resources Information Center

Seikel, J. Anthony; And Others

1990-01-01

This study investigated the relationship between the temporal-acoustic parameters of the speech of 15 adults with motor neuron disease. Differences in predictions of the progression of the disease and clinician judgments of dysarthria severity were found to relate to the linguistic systems of both speaker and judge. (Author/JDD)

Educational Information Quantization for Improving Content Quality in Learning Management Systems

ERIC Educational Resources Information Center

Rybanov, Alexander Aleksandrovich

2014-01-01

The article offers the educational information quantization method for improving content quality in Learning Management Systems. The paper considers questions concerning analysis of quality of quantized presentation of educational information, based on quantitative text parameters: average frequencies of parts of speech, used in the text; formal…

The Effects of Different Noise Types on Heart Rate Variability in Men

PubMed Central

Sim, Chang Sun; Sung, Joo Hyun; Cheon, Sang Hyeon; Lee, Jang Myung; Lee, Jae Won

2015-01-01

Purpose To determine the impact of noise on heart rate variability (HRV) in men, with a focus on the noise type rather than on noise intensity. Materials and Methods Forty college-going male volunteers were enrolled in this study and were randomly divided into four groups according to the type of noise they were exposed to: background, traffic, speech, or mixed (traffic and speech) noise. All groups except the background group (35 dB) were exposed to 45 dB sound pressure levels. We collected data on age, smoking status, alcohol consumption, and disease status from responses to self-reported questionnaires and medical examinations. We also measured HRV parameters and blood pressure levels before and after exposure to noise. The HRV parameters were evaluated while patients remained seated for 5 minutes, and frequency and time domain analyses were then performed. Results After noise exposure, only the speech noise group showed a reduced low frequency (LF) value, reflecting the activity of both the sympathetic and parasympathetic nervous systems. The low-to-high frequency (LF/HF) ratio, which reflected the activity of the autonomic nervous system (ANS), became more stable, decreasing from 5.21 to 1.37; however, this change was not statistically significant. Conclusion These results indicate that 45 dB(A) of noise, 10 dB(A) higher than background noise, affects the ANS. Additionally, the impact on HRV activity might differ according to the noise quality. Further studies will be required to ascertain the role of noise type. PMID:25510770

Acoustic Quality Levels of Mosques in Batu Pahat

NASA Astrophysics Data System (ADS)

Azizah Adnan, Nor; Nafida Raja Shahminan, Raja; Khair Ibrahim, Fawazul; Tami, Hannifah; Yusuff, M. Rizal M.; Murniwaty Samsudin, Emedya; Ismail, Isham

2018-04-01

Every Friday, Muslims has been required to perform a special prayer known as the Friday prayers which involve the delivery of a brief lecture (Khutbah). Speech intelligibility in oral communications presented by the preacher affected all the congregation and determined the level of acoustic quality in the interior of the mosque. Therefore, this study intended to assess the level of acoustic quality of three public mosques in Batu Pahat. Good acoustic quality is essential in contributing towards appreciation in prayers and increasing khusyu’ during the worship, which is closely related to the speech intelligibility corresponding to the actual function of the mosque according to Islam. Acoustic parameters measured includes noise criteria (NC), reverberation time (RT) and speech transmission index (STI), and was performed using the sound level meter and sound measurement instruments. This test is carried out through the physical observation with the consideration of space and volume design as a factor affecting acoustic parameters. Results from all 3 mosques as the showed that the acoustic quality level inside these buildings are slightly poor which is at below 0.45 coefficients based on the standard. Among the factors that influencing the low acoustical quality are location, building materials, installation of sound absorption material and the number of occupants inside the mosque. As conclusion, the acoustic quality level of a mosque is highly depends on physical factors of the mosque such as the architectural design and space volume besides other factors as been identified by this study.

Acoustic and Perceptual Analyses of Adductor Spasmodic Dysphonia in Mandarin-speaking Chinese.

PubMed

Chen, Zhipeng; Li, Jingyuan; Ren, Qingyi; Ge, Pingjiang

2018-02-12

The objective of this study was to examine the perceptual structure and acoustic characteristics of speech of patients with adductor spasmodic dysphonia (ADSD) in Mandarin. Case-Control Study MATERIALS AND METHODS: For the estimation of dysphonia level, perceptual and acoustic analysis were used for patients with ADSD (N = 20) and the control group (N = 20) that are Mandarin-Chinese speakers. For both subgroups, a sustained vowel and connected speech samples were obtained. The difference of perceptual and acoustic parameters between the two subgroups was assessed and analyzed. For acoustic assessment, the percentage of phonatory breaks (PBs) of connected reading and the percentage of aperiodic segments and frequency shifts (FS) of vowel and reading in patients with ADSD were significantly worse than controls, the mean harmonics-to-noise ratio and the fundamental frequency standard deviation of vowel as well. For perceptual evaluation, the rating of speech and vowel in patients with ADSD are significantly higher than controls. The percentage of aberrant acoustic events (PB, frequency shift, and aperiodic segment) and the fundamental frequency standard deviation and mean harmonics-to-noise ratio were significantly correlated with the perceptual rating in the vowel and reading productions. The perceptual and acoustic parameters of connected vowel and reading in patients with ADSD are worse than those in normal controls, and could validly and reliably estimate dysphonia of ADSD in Mandarin-speaking Chinese. Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

The effects of different noise types on heart rate variability in men.

PubMed

Sim, Chang Sun; Sung, Joo Hyun; Cheon, Sang Hyeon; Lee, Jang Myung; Lee, Jae Won; Lee, Jiho

2015-01-01

To determine the impact of noise on heart rate variability (HRV) in men, with a focus on the noise type rather than on noise intensity. Forty college-going male volunteers were enrolled in this study and were randomly divided into four groups according to the type of noise they were exposed to: background, traffic, speech, or mixed (traffic and speech) noise. All groups except the background group (35 dB) were exposed to 45 dB sound pressure levels. We collected data on age, smoking status, alcohol consumption, and disease status from responses to self-reported questionnaires and medical examinations. We also measured HRV parameters and blood pressure levels before and after exposure to noise. The HRV parameters were evaluated while patients remained seated for 5 minutes, and frequency and time domain analyses were then performed. After noise exposure, only the speech noise group showed a reduced low frequency (LF) value, reflecting the activity of both the sympathetic and parasympathetic nervous systems. The low-to-high frequency (LF/HF) ratio, which reflected the activity of the autonomic nervous system (ANS), became more stable, decreasing from 5.21 to 1.37; however, this change was not statistically significant. These results indicate that 45 dB(A) of noise, 10 dB(A) higher than background noise, affects the ANS. Additionally, the impact on HRV activity might differ according to the noise quality. Further studies will be required to ascertain the role of noise type.

A new model of sensorimotor coupling in the development of speech.

PubMed

Westermann, Gert; Reck Miranda, Eduardo

2004-05-01

We present a computational model that learns a coupling between motor parameters and their sensory consequences in vocal production during a babbling phase. Based on the coupling, preferred motor parameters and prototypically perceived sounds develop concurrently. Exposure to an ambient language modifies perception to coincide with the sounds from the language. The model develops motor mirror neurons that are active when an external sound is perceived. An extension to visual mirror neurons for oral gestures is suggested.

Motif Discovery in Speech: Application to Monitoring Alzheimer's Disease.

PubMed

Garrard, Peter; Nemes, Vanda; Nikolic, Dragana; Barney, Anna

2017-01-01

Perseveration - repetition of words, phrases or questions in speech - is commonly described in Alzheimer's disease (AD). Measuring perseveration is difficult, but may index cognitive performance, aiding diagnosis and disease monitoring. Continuous recording of speech would produce a large quantity of data requiring painstaking manual analysis, and risk violating patients' and others' privacy. A secure record and an automated approach to analysis are required. To record bone-conducted acoustic energy fluctuations from a subject's vocal apparatus using an accelerometer, to describe the recording and analysis stages in detail, and demonstrate that the approach is feasible in AD. Speech-related vibration was captured by an accelerometer, affixed above the temporomandibular joint. Healthy subjects read a script with embedded repetitions. Features were extracted from recorded signals and combined using Principal Component Analysis to obtain a one-dimensional representation of the feature vector. Motif discovery techniques were used to detect repeated segments. The equipment was tested in AD patients to determine device acceptability and recording quality. Comparison with the known location of embedded motifs suggests that, with appropriate parameter tuning, the motif discovery method can detect repetitions. The device was acceptable to patients and produced adequate signal quality in their home environments. We established that continuously recording bone-conducted speech and detecting perseverative patterns were both possible. In future studies we plan to associate the frequency of verbal repetitions with stage, progression and type of dementia. It is possible that the method could contribute to the assessment of disease-modifying treatments. Copyright© Bentham Science Publishers; For any queries, please email at epub@benthamscience.org.

Acoustic voice analysis of prelingually deaf adults before and after cochlear implantation.

PubMed

Evans, Maegan K; Deliyski, Dimitar D

2007-11-01

It is widely accepted that many severe to profoundly deaf adults have benefited from cochlear implants (CIs). However, limited research has been conducted to investigate changes in voice and speech of prelingually deaf adults who receive CIs, a population well known for presenting with a variety of voice and speech abnormalities. The purpose of this study was to use acoustic analysis to explore changes in voice and speech for three prelingually deaf males pre- and postimplantation over 6 months. The following measurements, some measured in varying contexts, were obtained: fundamental frequency (F0), jitter, shimmer, noise-to-harmonic ratio, voice turbulence index, soft phonation index, amplitude- and F0-variation, F0-range, speech rate, nasalance, and vowel production. Characteristics of vowel production were measured by determining the first formant (F1) and second formant (F2) of vowels in various contexts, magnitude of F2-variation, and rate of F2-variation. Perceptual measurements of pitch, pitch variability, loudness variability, speech rate, and intonation were obtained for comparison. Results are reported using descriptive statistics. The results showed patterns of change for some of the parameters while there was considerable variation across the subjects. All participants demonstrated a decrease in F0 in at least one context and demonstrated a change in nasalance toward the norm as compared to their normal hearing control. The two participants who were oral-language communicators were judged to produce vowels with an average of 97.2% accuracy and the sign-language user demonstrated low percent accuracy for vowel production.

New simple evaluation method of the monosyllable /sa/ using a psychoacoustic system in maxillectomy patients.

PubMed

Chowdhury, Nafees Uddin; Otomaru, Takafumi; Murase, Mai; Inohara, Ken; Hattori, Mariko; Sumita, Yuka I; Taniguchi, Hisashi

2011-01-01

An objective assessment of speech would benefit the prosthetic rehabilitation of maxillectomy patients. This study aimed to establish a simple, objective evaluation of monosyllable /sa/ utterances in maxillectomy patients by using a psychoacoustic system typically used in industry. This study comprised two experiments. Experiment 1 involved analysis of the psychoacoustic parameters (loudness, sharpness and roughness) in monosyllable /sa/ utterances by 18 healthy subjects (9 males, 9 females). The utterances were recorded in a sound-treated room. The coefficient of variation (CV) for each parameter was compared to identify the most suitable parameter for objective evaluation of speech. Experiment 2 involved analysis of /sa/ utterances by 18 maxillectomy patients (9 males, 9 females) with and without prosthesis, and comparisons of the psychoacoustic data between the healthy subjects and maxillectomy patients without prosthesis, between the maxillectomy patients with and without prosthesis, and between the healthy subjects and maxillectomy patients with prosthesis. The CV for sharpness was the lowest among the three psychoacoustic parameters in both the healthy males and females. There were significant differences in the sharpness of /sa/ between the healthy subjects and the maxillectomy patients without prosthesis (but not with prosthesis), and between the maxillectomy patients with and without prosthesis. We found that the psychoacoustic parameters typically adopted in industrial research could also be applied to evaluate the psychoacoustics of the monosyllable /sa/ utterance, and distinguished the monosyllable /sa/ in maxillectomy patients with an obturator from that without an obturator using the system. Copyright © 2010 Japan Prosthodontic Society. Published by Elsevier Ltd. All rights reserved.

Sub-band/transform compression of video sequences

NASA Technical Reports Server (NTRS)

Sauer, Ken; Bauer, Peter

1992-01-01

The progress on compression of video sequences is discussed. The overall goal of the research was the development of data compression algorithms for high-definition television (HDTV) sequences, but most of our research is general enough to be applicable to much more general problems. We have concentrated on coding algorithms based on both sub-band and transform approaches. Two very fundamental issues arise in designing a sub-band coder. First, the form of the signal decomposition must be chosen to yield band-pass images with characteristics favorable to efficient coding. A second basic consideration, whether coding is to be done in two or three dimensions, is the form of the coders to be applied to each sub-band. Computational simplicity is of essence. We review the first portion of the year, during which we improved and extended some of the previous grant period's results. The pyramid nonrectangular sub-band coder limited to intra-frame application is discussed. Perhaps the most critical component of the sub-band structure is the design of bandsplitting filters. We apply very simple recursive filters, which operate at alternating levels on rectangularly sampled, and quincunx sampled images. We will also cover the techniques we have studied for the coding of the resulting bandpass signals. We discuss adaptive three-dimensional coding which takes advantage of the detection algorithm developed last year. To this point, all the work on this project has been done without the benefit of motion compensation (MC). Motion compensation is included in many proposed codecs, but adds significant computational burden and hardware expense. We have sought to find a lower-cost alternative featuring a simple adaptation to motion in the form of the codec. In sequences of high spatial detail and zooming or panning, it appears that MC will likely be necessary for the proposed quality and bit rates.

[Hemisphere dominance of cortical speech processing and its dynamics in the course of aphasia therapy. A psychophysiologic contribution to the discussion of the neuronal plasticity hypothesis].

PubMed

Maier, R; Lünser, W

1991-01-01

Examinations of aphasic patients by using cognitive tasks were based on the hypothesis that semantically evoked potentials correlate to the processing of information in the cortical areas of Broca and Wernicke. Some of the examined right-handed patients with ischemic lesions of the left hemisphere produced characteristic potentials in the right temporal lobe, and not in the dominant left lobe as was expected. These case histories suggest that in these patients speech processing moved into the subdominant hemisphere as a result of compensation after cerebral damage by using the faculty of neuroplasticity. Furthermore an extended classification of aphasias is presented, illustrated by a three-parameter model.

Real time speech formant analyzer and display

DOEpatents

Holland, George E.; Struve, Walter S.; Homer, John F.

1987-01-01

A speech analyzer for interpretation of sound includes a sound input which converts the sound into a signal representing the sound. The signal is passed through a plurality of frequency pass filters to derive a plurality of frequency formants. These formants are converted to voltage signals by frequency-to-voltage converters and then are prepared for visual display in continuous real time. Parameters from the inputted sound are also derived and displayed. The display may then be interpreted by the user. The preferred embodiment includes a microprocessor which is interfaced with a television set for displaying of the sound formants. The microprocessor software enables the sound analyzer to present a variety of display modes for interpretive and therapeutic used by the user.

Author's Rebuttal to Smits et al. (2018), "Comment on 'Sensitivity of the Speech Intelligibility Index to the Assumed Dynamic Range' by Jin et al. (2017)".

PubMed

Jin, In-Ki; Kates, James M; Arehart, Kathryn H

2018-01-22

The purpose of this letter is to refute the comments written by Smits, Goverts, and Versfeld (2018). Refutations to each issue including the fixed mathematical relationship between dynamic range (DR) and a fitting constant (Q value), deviating results for small DRs, and determination of Speech Intelligibility Index (SII) model parameters are described. Although Smits et al. (2018) correctly identified several issues, those comments do not diminish the results of the original article (Jin, Kates, & Arehart, 2017) in providing new insights for the SII. Jin et al. (2017) clearly provided the impact of languages and DR on the SII, which was the main result of the study.

Real time speech formant analyzer and display

DOEpatents

Holland, G.E.; Struve, W.S.; Homer, J.F.

1987-02-03

A speech analyzer for interpretation of sound includes a sound input which converts the sound into a signal representing the sound. The signal is passed through a plurality of frequency pass filters to derive a plurality of frequency formants. These formants are converted to voltage signals by frequency-to-voltage converters and then are prepared for visual display in continuous real time. Parameters from the inputted sound are also derived and displayed. The display may then be interpreted by the user. The preferred embodiment includes a microprocessor which is interfaced with a television set for displaying of the sound formants. The microprocessor software enables the sound analyzer to present a variety of display modes for interpretive and therapeutic used by the user. 19 figs.

Speaker normalization and adaptation using second-order connectionist networks.

PubMed

Watrous, R L

1993-01-01

A method for speaker normalization and adaption using connectionist networks is developed. A speaker-specific linear transformation of observations of the speech signal is computed using second-order network units. Classification is accomplished by a multilayer feedforward network that operates on the normalized speech data. The network is adapted for a new talker by modifying the transformation parameters while leaving the classifier fixed. This is accomplished by backpropagating classification error through the classifier to the second-order transformation units. This method was evaluated for the classification of ten vowels for 76 speakers using the first two formant values of the Peterson-Barney data. The results suggest that rapid speaker adaptation resulting in high classification accuracy can be accomplished by this method.

Repair Sequences in Dysarthric Conversational Speech: A Study in Interactional Phonetics

ERIC Educational Resources Information Center

Rutter, Ben

2009-01-01

This paper presents some findings from a case study of repair sequences in conversations between a dysarthric speaker, Chris, and her interactional partners. It adopts the methodology of interactional phonetics, where turn design, sequence organization, and variation in phonetic parameters are analysed in unison. The analysis focused on the use of…

Analysing Normal and Partial Glossectomee Tongues Using Ultrasound

ERIC Educational Resources Information Center

Bressmann, Tim; Uy, Catherine; Irish, Jonathan C.

2005-01-01

The present study aimed at identifying underlying parameters that govern the shape of the tongue. A functional topography of the tongue surface was developed based on three-dimensional ultrasound scans of sustained speech sounds in ten normal subjects. A principal component analysis extracted three components that explained 89.2% of the variance…

The relationship between the Nasality Severity Index 2.0 and perceptual judgments of hypernasality.

PubMed

Bettens, Kim; De Bodt, Marc; Maryn, Youri; Luyten, Anke; Wuyts, Floris L; Van Lierde, Kristiane M

2016-01-01

The Nasality Severity Index 2.0 (NSI 2.0) forms a new, multiparametric approach in the identification of hypernasality. The present study aimed to investigate the correlation between the NSI 2.0 scores and the perceptual assessment of hypernasality. Speech samples of 35 patients, representing a range of nasality from normal to severely hypernasal, were rated by four expert speech-language pathologists using visual analogue scaling (VAS) judging the degree of hypernasality, audible nasal airflow (ANA) and speech intelligibility. Inter- and intra-listener reliability was verified using intraclass correlation coefficients. Correlations between NSI 2.0 scores and its parameters (i.e. nasalance score of an oral text and vowel /u/, voice low tone to high tone ratio of the vowel /i/) and the degree of hypernasality were determined using Pearson correlation coefficients. Multiple linear regression analysis was used to investigate the possible influence of ANA and speech intelligibility on the NSI 2.0 scores. Overall good to excellent inter- and intra-listener reliability was found for the perceptual ratings. A moderate, but significant negative correlation between NSI 2.0 scores and perceived hypernasality (r=-0.64) was found, in which a more negative NSI 2.0 score indicates the presence of more severe hypernasality. No significant influence of ANA or intelligibility on the NSI 2.0 was observed based on the regression analysis. Because the NSI 2.0 correlates significantly with perceived hypernasality, it provides an easy-to-interpret severity score of hypernasality which will facilitate the evaluation of therapy outcomes, communication to the patient and other clinicians, and decisions for treatment planning, based on a multiparametric approach. However, research is still necessary to further explore the instrumental correlates of perceived hypernasality. The reader will be able to (1) describe and discuss current issues and influencing variables regarding perceptual ratings of hypernasality; (2) describe and discuss the relationship between the Nasality Severity Index 2.0, a new multiparametric approach to hypernasality, and perceptual judgments of hypernasality based on visual analogue scale ratings; (3) compare these results with the correlations based on a single parameter approach and (4) describe and discuss the possible influence of audible nasal airflow and speech intelligibility on the NSI 2.0 scores. Copyright © 2016 Elsevier Inc. All rights reserved.

Acoustics in Halls for Speech and Music

NASA Astrophysics Data System (ADS)

Gade, Anders C.

This chapter deals specifically with concepts, tools, and architectural variables of importance when designing auditoria for speech and music. The focus will be on cultivating the useful components of the sound in the room rather than on avoiding noise from outside or from installations, which is dealt with in Chap. 11. The chapter starts by presenting the subjective aspects of the room acoustic experience according to consensus at the time of writing. Then follows a description of their objective counterparts, the objective room acoustic parameters, among which the classical reverberation time measure is only one of many, but still of fundamental value. After explanations on how these parameters can be measured and predicted during the design phase, the remainder of the chapter deals with how the acoustic properties can be controlled by the architectural design of auditoria. This is done by presenting the influence of individual design elements as well as brief descriptions of halls designed for specific purposes, such as drama, opera, and symphonic concerts. Finally, some important aspects of loudspeaker installations in auditoria are briefly touched upon.

Effects of sound source location and direction on acoustic parameters in Japanese churches.

PubMed

Soeta, Yoshiharu; Ito, Ken; Shimokura, Ryota; Sato, Shin-ichi; Ohsawa, Tomohiro; Ando, Yoichi

2012-02-01

In 1965, the Catholic Church liturgy changed to allow priests to face the congregation. Whereas Church tradition, teaching, and participation have been much discussed with respect to priest orientation at Mass, the acoustical changes in this regard have not yet been examined scientifically. To discuss acoustic desired within churches, it is necessary to know the acoustical characteristics appropriate for each phase of the liturgy. In this study, acoustic measurements were taken at various source locations and directions using both old and new liturgies performed in Japanese churches. A directional loudspeaker was used as the source to provide vocal and organ acoustic fields, and impulse responses were measured. Various acoustical parameters such as reverberation time and early decay time were analyzed. The speech transmission index was higher for the new Catholic liturgy, suggesting that the change in liturgy has improved speech intelligibility. Moreover, the interaural cross-correlation coefficient and early lateral energy fraction were higher and lower, respectively, suggesting that the change in liturgy has made the apparent source width smaller. © 2012 Acoustical Society of America

Mild Developmental Foreign Accent Syndrome and Psychiatric Comorbidity: Altered White Matter Integrity in Speech and Emotion Regulation Networks

PubMed Central

Berthier, Marcelo L.; Roé-Vellvé, Núria; Moreno-Torres, Ignacio; Falcon, Carles; Thurnhofer-Hemsi, Karl; Paredes-Pacheco, José; Torres-Prioris, María J.; De-Torres, Irene; Alfaro, Francisco; Gutiérrez-Cardo, Antonio L.; Baquero, Miquel; Ruiz-Cruces, Rafael; Dávila, Guadalupe

2016-01-01

Foreign accent syndrome (FAS) is a speech disorder that is defined by the emergence of a peculiar manner of articulation and intonation which is perceived as foreign. In most cases of acquired FAS (AFAS) the new accent is secondary to small focal lesions involving components of the bilaterally distributed neural network for speech production. In the past few years FAS has also been described in different psychiatric conditions (conversion disorder, bipolar disorder, and schizophrenia) as well as in developmental disorders (specific language impairment, apraxia of speech). In the present study, two adult males, one with atypical phonetic production and the other one with cluttering, reported having developmental FAS (DFAS) since their adolescence. Perceptual analysis by naïve judges could not confirm the presence of foreign accent, possibly due to the mildness of the speech disorder. However, detailed linguistic analysis provided evidence of prosodic and segmental errors previously reported in AFAS cases. Cognitive testing showed reduced communication in activities of daily living and mild deficits related to psychiatric disorders. Psychiatric evaluation revealed long-lasting internalizing disorders (neuroticism, anxiety, obsessive-compulsive disorder, social phobia, depression, alexithymia, hopelessness, and apathy) in both subjects. Diffusion tensor imaging (DTI) data from each subject with DFAS were compared with data from a group of 21 age- and gender-matched healthy control subjects. Diffusion parameters (MD, AD, and RD) in predefined regions of interest showed changes of white matter microstructure in regions previously related with AFAS and psychiatric disorders. In conclusion, the present findings militate against the possibility that these two subjects have FAS of psychogenic origin. Rather, our findings provide evidence that mild DFAS occurring in the context of subtle, yet persistent, developmental speech disorders may be associated with structural brain anomalies. We suggest that the simultaneous involvement of speech and emotion regulation networks might result from disrupted neural organization during development, or compensatory or maladaptive plasticity. Future studies are required to examine whether the interplay between biological trait-like diathesis (shyness, neuroticism) and the stressful experience of living with mild DFAS lead to the development of internalizing psychiatric disorders. PMID:27555813

Investigations in mechanisms and strategies to enhance hearing with cochlear implants

NASA Astrophysics Data System (ADS)

Churchill, Tyler H.

Cochlear implants (CIs) produce hearing sensations by stimulating the auditory nerve (AN) with current pulses whose amplitudes are modulated by filtered acoustic temporal envelopes. While this technology has provided hearing for multitudinous CI recipients, even bilaterally-implanted listeners have more difficulty understanding speech in noise and localizing sounds than normal hearing (NH) listeners. Three studies reported here have explored ways to improve electric hearing abilities. Vocoders are often used to simulate CIs for NH listeners. Study 1 was a psychoacoustic vocoder study examining the effects of harmonic carrier phase dispersion and simulated CI current spread on speech intelligibility in noise. Results showed that simulated current spread was detrimental to speech understanding and that speech vocoded with carriers whose components' starting phases were equal was the least intelligible. Cross-correlogram analyses of AN model simulations confirmed that carrier component phase dispersion resulted in better neural envelope representation. Localization abilities rely on binaural processing mechanisms in the brainstem and mid-brain that are not fully understood. In Study 2, several potential mechanisms were evaluated based on the ability of metrics extracted from stereo AN simulations to predict azimuthal locations. Results suggest that unique across-frequency patterns of binaural cross-correlation may provide a strong cue set for lateralization and that interaural level differences alone cannot explain NH sensitivity to lateral position. While it is known that many bilateral CI users are sensitive to interaural time differences (ITDs) in low-rate pulsatile stimulation, most contemporary CI processing strategies use high-rate, constant-rate pulse trains. In Study 3, we examined the effects of pulse rate and pulse timing on ITD discrimination, ITD lateralization, and speech recognition by bilateral CI listeners. Results showed that listeners were able to use low-rate pulse timing cues presented redundantly on multiple electrodes for ITD discrimination and lateralization of speech stimuli even when mixed with high rates on other electrodes. These results have contributed to a better understanding of those aspects of the auditory system that support speech understanding and binaural hearing, suggested vocoder parameters that may simulate aspects of electric hearing, and shown that redundant, low-rate pulse timing supports improved spatial hearing for bilateral CI listeners.

Rating, ranking, and understanding acoustical quality in university classrooms

NASA Astrophysics Data System (ADS)

Hodgson, Murray

2002-08-01

Nonoptimal classroom acoustical conditions directly affect speech perception and, thus, learning by students. Moreover, they may lead to voice problems for the instructor, who is forced to raise his/her voice when lecturing to compensate for poor acoustical conditions. The project applied previously developed simplified methods to predict speech intelligibility in occupied classrooms from measurements in unoccupied and occupied university classrooms. The methods were used to predict the speech intelligibility at various positions in 279 University of British Columbia (UBC) classrooms, when 70% occupied, and for four instructor voice levels. Classrooms were classified and rank ordered by acoustical quality, as determined by the room-average speech intelligibility. This information was used by UBC to prioritize classrooms for renovation. Here, the statistical results are reported to illustrate the range of acoustical qualities found at a typical university. Moreover, the variations of quality with relevant classroom acoustical parameters were studied to better understand the results. In particular, the factors leading to the best and worst conditions were studied. It was found that 81% of the 279 classrooms have "good," "very good," or "excellent" acoustical quality with a "typical" (average-male) instructor. However, 50 (18%) of the classrooms had "fair" or "poor" quality, and two had "bad" quality, due to high ventilation-noise levels. Most rooms were "very good" or "excellent" at the front, and "good" or "very good" at the back. Speech quality varied strongly with the instructor voice level. In the worst case considered, with a quiet female instructor, most of the classrooms were "bad" or "poor." Quality also varies with occupancy, with decreased occupancy resulting in decreased quality. The research showed that a new classroom acoustical design and renovation should focus on limiting background noise. They should promote high instructor speech levels at the back of the classrooms. This involves, in part, limiting the amount of sound absorption that is introduced into classrooms to control reverberation. Speech quality is not very sensitive to changes in reverberation, so controlling it for its own sake should not be a design priority. copyright 2002 Acoustical Society of America.

[Rehabilitative measures in hearing-impaired children].

PubMed

von Wedel, H; von Wedel, U C; Zorowka, P

1991-12-01

On the basis of certain fundamental data on the maturation processes of the central auditory pathways in early childhood the importance of early intervention with hearing aids is discussed and emphasized. Pathological hearing, that is acoustical deprivation in early childhood will influence the maturation process. Very often speech development is delayed if diagnosis and therapy or rehabilitation are not early enough. Anamnesis, early diagnosis and clinical differential diagnosis are required before a hearing aid can be fitted. Selection criteria and adjustment parameters are discussed, showing that the hearing aid fitting procedure must be embedded in a complex matrix of requirements related to the development of speech as well as to the cognitive, emotional and social development of the child. As a rule, finding and preparing the "best" hearing aids (binaural fitting is obligatory) for a child is a long and often difficult process, which can only be performed by specialists who are pedo-audiologists. After the binaural fitting of hearing aids an intensive hearing and speech education in close cooperation between parents, pedo-audiologist and teacher must support the whole development of the child.

Signal Prediction With Input Identification

NASA Technical Reports Server (NTRS)

Juang, Jer-Nan; Chen, Ya-Chin

1999-01-01

A novel coding technique is presented for signal prediction with applications including speech coding, system identification, and estimation of input excitation. The approach is based on the blind equalization method for speech signal processing in conjunction with the geometric subspace projection theory to formulate the basic prediction equation. The speech-coding problem is often divided into two parts, a linear prediction model and excitation input. The parameter coefficients of the linear predictor and the input excitation are solved simultaneously and recursively by a conventional recursive least-squares algorithm. The excitation input is computed by coding all possible outcomes into a binary codebook. The coefficients of the linear predictor and excitation, and the index of the codebook can then be used to represent the signal. In addition, a variable-frame concept is proposed to block the same excitation signal in sequence in order to reduce the storage size and increase the transmission rate. The results of this work can be easily extended to the problem of disturbance identification. The basic principles are outlined in this report and differences from other existing methods are discussed. Simulations are included to demonstrate the proposed method.

Kinematic analysis of jaw function in children following traumatic brain injury.

PubMed

Loh, E W L; Goozée, J V; Murdoch, B E

2005-07-01

To investigate jaw movements in children following traumatic brain injury (TBI) during speech using electromagnetic articulography (EMA). Jaw movements of two non-dysarthric children (aged 12.75 and 13.08 years) who had sustained a TBI were recorded using the AG-100 EMA system (Carstens Medizineletronik) during word-initial consonant productions. Mean quantitative kinematic parameters and coefficient of variation (variability) values were calculated and individually compared to the mean values obtained by a group of six control children (mean age 12.57 years, SD 1.52). The two children with TBI exhibited word-initial consonant jaw movement durations that were comparable to the control children, with sub-clinical reductions in speed being offset by reduced distances. Differences were observed between the two children in jaw kinematic variability, with one child exhibiting increased variability, while the other child demonstrated reduced or comparable variability compared to the control group. Possible sub-clinical impairments of jaw movement for speech were exhibited by two children who had sustained a TBI, providing insight into the consequences of TBI on speech motor control development.

The Interaction of Lexical Characteristics and Speech Production in Parkinson's Disease.

PubMed

Chiu, Yi-Fang; Forrest, Karen

2017-01-01

This study sought to investigate the interaction of speech movement execution with higher order lexical parameters. The authors examined how lexical characteristics affect speech output in individuals with Parkinson's disease (PD) and healthy control (HC) speakers. Twenty speakers with PD and 12 healthy speakers read sentences with target words that varied in word frequency and neighborhood density. The formant transitions (F2 slopes) of the diphthongs in the target words were compared across lexical categories between PD and HC groups. Both groups of speakers produced steeper F2 slopes for the diphthongs in less frequent words and words from sparse neighborhoods. The magnitude of the increase in F2 slopes was significantly less in the PD than HC group. The lexical effect on the F2 slope differed among the diphthongs and between the 2 groups. PD and healthy speakers varied their acoustic output on the basis of word frequency and neighborhood density. F2 slope variations can be traced to higher level lexical differences. This lexical effect on articulation, however, appears to be constrained by PD.

Hypertrophied tonsils impair velopharyngeal function after palatoplasty.

PubMed

Abdel-Aziz, Mosaad

2012-03-01

When tonsillar hypertrophy obstructing the airway is encountered in a child with a repaired cleft palate and velopharyngeal insufficiency, the surgeon may opt for tonsillectomy to relieve the airway obstruction, with possible effects on velopharyngeal closure. The aim of this study was to assess the impact of hypertrophied tonsils on velopharyngeal function in children with repaired cleft palate and to measure the effect of tonsillectomy on velopharyngeal closure and speech resonance. Case series. Twelve children with repaired cleft palate and tonsillar hypertrophy underwent tonsillectomy to relieve airway obstruction. Preoperative and postoperative evaluation of velopharyngeal function was performed. Auditory perceptual assessment of speech and nasalance scores were measured, and velopharyngeal closure was evaluated by flexible nasopharyngoscopy. Preoperative impairment of velopharyngeal function was detected. However, significant postoperative improvement of speech parameters (hypernasality, nasal emission of air, and weak pressure consonants measured with auditory perceptual assessment) was achieved, and the overall postoperative nasalance score was improved significantly for nasal and oral sentences. Reduction of velopharyngeal gap size was detected after removal of hypertrophied tonsils. Although the improvement of velopharyngeal closure was not significant, three cases demonstrated complete postoperative closure with no gap. Hypertrophied tonsils may impair velopharyngeal function in children with repaired cleft palate, and tonsillectomy is beneficial for such patients as it can improve the velopharyngeal closure and speech resonance. Secondary corrective surgery may be avoided in some cases after tonsillectomy. Copyright © 2012 The American Laryngological, Rhinological, and Otological Society, Inc.

Perceptual, auditory and acoustic vocal analysis of speech and singing in choir conductors.

PubMed

Rehder, Maria Inês Beltrati Cornacchioni; Behlau, Mara

2008-01-01

the voice of choir conductors. to evaluate the vocal quality of choir conductors based on the production of a sustained vowel during singing and when speaking in order to observe auditory and acoustic differences. participants of this study were 100 choir conductors, with an equal distribution between genders. Participants were asked to produce the sustained vowel "é" using a singing and speaking voice. Speech samples were analyzed based on auditory-perceptive and acoustic parameters. The auditory-perceptive analysis was carried out by two speech-language pathologist, specialists in this field of knowledge. The acoustic analysis was carried out with the support of the computer software Doctor Speech (Tiger Electronics, SRD, USA, version 4.0), using the Real Analysis module. the auditory-perceptive analysis of the vocal quality indicated that most conductors have adapted voices, presenting more alterations in their speaking voice. The acoustic analysis indicated different values between genders and between the different production modalities. The fundamental frequency was higher in the singing voice, as well as the values for the first formant; the second formant presented lower values in the singing voice, with statistically significant results only for women. the voice of choir conductors is adapted, presenting fewer deviations in the singing voice when compared to the speaking voice. Productions differ based the voice modality, singing or speaking.

Pre- and posttreatment voice and speech outcomes in patients with advanced head and neck cancer treated with chemoradiotherapy: expert listeners' and patient's perception.

PubMed

van der Molen, Lisette; van Rossum, Maya A; Jacobi, Irene; van Son, Rob J J H; Smeele, Ludi E; Rasch, Coen R N; Hilgers, Frans J M

2012-09-01

Perceptual judgments and patients' perception of voice and speech after concurrent chemoradiotherapy (CCRT) for advanced head and neck cancer. Prospective clinical trial. A standard Dutch text and a diadochokinetic task were recorded. Expert listeners rated voice and speech quality (based on Grade, Roughness, Breathiness, Asthenia, and Strain), articulation (overall, [p], [t], [k]), and comparative mean opinion scores of voice and speech at three assessment points calculated. A structured study-specific questionnaire evaluated patients' perception pretreatment (N=55), at 10-week (N=49) and 1-year posttreatment (N=37). At 10 weeks, perceptual voice quality is significantly affected. The parameters overall voice quality (mean, -0.24; P=0.008), strain (mean, -0.12; P=0.012), nasality (mean, -0.08; P=0.009), roughness (mean, -0.22; P=0.001), and pitch (mean, -0.03; P=0.041) improved over time but not beyond baseline levels, except for asthenia at 1-year posttreatment (voice is less asthenic than at baseline; mean, +0.20; P=0.03). Perceptual analyses of articulation showed no significant differences. Patients judge their voice quality as good (score, 18/20) at all assessment points, but at 1-year posttreatment, most of them (70%) judge their "voice not as it used to be." In the 1-year versus 10-week posttreatment comparison, the larynx-hypopharynx tumor group was more strained, whereas nonlarynx tumor voices were judged less strained (mean, -0.33 and +0.07, respectively; P=0.031). Patients' perceived changes in voice and speech quality at 10-week post- versus pretreatment correlate weakly with expert judgments. Overall, perceptual CCRT effects on voice and speech seem to peak at 10-week posttreatment but level off at 1-year posttreatment. However, at that assessment point, most patients still perceive their voice as different from baseline. Copyright © 2012 The Voice Foundation. Published by Mosby, Inc. All rights reserved.

Acoustic characteristics of modern Greek Orthodox Church music.

PubMed

Delviniotis, Dimitrios S

2013-09-01

Some acoustic characteristics of the two types of vocal music of the Greek Orthodox Church Music, the Byzantine chant (BC) and ecclesiastical speech (ES), are studied in relation to the common Greek speech and the Western opera. Vocal samples were obtained, and their acoustic parameters of sound pressure level (SPL), fundamental frequency (F0), and the long-time average spectrum (LTAS) characteristics were analyzed. Twenty chanters, including two chanters-singers of opera, sang (BC) and read (ES) the same hymn of Byzantine music (BM), the two opera singers sang the same aria of opera, and common speech samples were obtained, and all audio were analyzed. The distribution of SPL values showed that the BC and ES have higher SPL by 9 and 12 dB, respectively, than common speech. The average F0 in ES tends to be lower than the common speech, and the smallest standard deviation (SD) of F0 values characterizes its monotonicity. The tone-scale intervals of BC are close enough to the currently accepted theory with SD equal to 0.24 semitones. The rate and extent of vibrato, which is rare in BC, equals 4.1 Hz and 0.6 semitones, respectively. The average LTAS slope is greatest in BC (+4.5 dB) but smaller than in opera (+5.7 dB). In both BC and ES, instead of a singer's formant appearing in an opera voice, a speaker's formant (SPF) was observed around 3300 Hz, with relative levels of +6.3 and +4.6 dB, respectively. The two vocal types of BM, BC, and ES differ both to each other and common Greek speech and opera style regarding SPL, the mean and SD of F0, the LTAS slope, and the relative level of SPF. Copyright © 2013 The Voice Foundation. Published by Mosby, Inc. All rights reserved.

Implicit prosody mining based on the human eye image capture technology

NASA Astrophysics Data System (ADS)

Gao, Pei-pei; Liu, Feng

2013-08-01

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

Flexible digital modulation and coding synthesis for satellite communications

NASA Technical Reports Server (NTRS)

Vanderaar, Mark; Budinger, James; Hoerig, Craig; Tague, John

1991-01-01

An architecture and a hardware prototype of a flexible trellis modem/codec (FTMC) transmitter are presented. The theory of operation is built upon a pragmatic approach to trellis-coded modulation that emphasizes power and spectral efficiency. The system incorporates programmable modulation formats, variations of trellis-coding, digital baseband pulse-shaping, and digital channel precompensation. The modulation formats examined include (uncoded and coded) binary phase shift keying (BPSK), quatenary phase shift keying (QPSK), octal phase shift keying (8PSK), 16-ary quadrature amplitude modulation (16-QAM), and quadrature quadrature phase shift keying (Q squared PSK) at programmable rates up to 20 megabits per second (Mbps). The FTMC is part of the developing test bed to quantify modulation and coding concepts.

Behavioural and neuroanatomical correlates of auditory speech analysis in primary progressive aphasias.

PubMed

Hardy, Chris J D; Agustus, Jennifer L; Marshall, Charles R; Clark, Camilla N; Russell, Lucy L; Bond, Rebecca L; Brotherhood, Emilie V; Thomas, David L; Crutch, Sebastian J; Rohrer, Jonathan D; Warren, Jason D

2017-07-27

Non-verbal auditory impairment is increasingly recognised in the primary progressive aphasias (PPAs) but its relationship to speech processing and brain substrates has not been defined. Here we addressed these issues in patients representing the non-fluent variant (nfvPPA) and semantic variant (svPPA) syndromes of PPA. We studied 19 patients with PPA in relation to 19 healthy older individuals. We manipulated three key auditory parameters-temporal regularity, phonemic spectral structure and prosodic predictability (an index of fundamental information content, or entropy)-in sequences of spoken syllables. The ability of participants to process these parameters was assessed using two-alternative, forced-choice tasks and neuroanatomical associations of task performance were assessed using voxel-based morphometry of patients' brain magnetic resonance images. Relative to healthy controls, both the nfvPPA and svPPA groups had impaired processing of phonemic spectral structure and signal predictability while the nfvPPA group additionally had impaired processing of temporal regularity in speech signals. Task performance correlated with standard disease severity and neurolinguistic measures. Across the patient cohort, performance on the temporal regularity task was associated with grey matter in the left supplementary motor area and right caudate, performance on the phoneme processing task was associated with grey matter in the left supramarginal gyrus, and performance on the prosodic predictability task was associated with grey matter in the right putamen. Our findings suggest that PPA syndromes may be underpinned by more generic deficits of auditory signal analysis, with a distributed cortico-subcortical neuraoanatomical substrate extending beyond the canonical language network. This has implications for syndrome classification and biomarker development.

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

PubMed

Boyd, Paul J

2006-12-01

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

RecoverNow: Feasibility of a Mobile Tablet-Based Rehabilitation Intervention to Treat Post-Stroke Communication Deficits in the Acute Care Setting

PubMed Central

Corbett, Dale; Finestone, Hillel M.; Hatcher, Simon; Lumsden, Jim; Momoli, Franco; Shamy, Michel C. F.; Stotts, Grant; Swartz, Richard H.; Yang, Christine

2016-01-01

Background Approximately 40% of patients diagnosed with stroke experience some degree of aphasia. With limited health care resources, patients’ access to speech and language therapies is often delayed. We propose using mobile-platform technology to initiate early speech-language therapy in the acute care setting. For this pilot, our objective was to assess the feasibility of a tablet-based speech-language therapy for patients with communication deficits following acute stroke. Methods We enrolled consecutive patients admitted with a stroke and communication deficits with NIHSS score ≥1 on the best language and/or dysarthria parameters. We excluded patients with severe comprehension deficits where communication was not possible. Following baseline assessment by a speech-language pathologist (SLP), patients were provided with a mobile tablet programmed with individualized therapy applications based on the assessment, and instructed to use it for at least one hour per day. Our objective was to establish feasibility by measuring recruitment rate, adherence rate, retention rate, protocol deviations and acceptability. Results Over 6 months, 143 patients were admitted with a new diagnosis of stroke: 73 had communication deficits, 44 met inclusion criteria, and 30 were enrolled into RecoverNow (median age 62, 26.6% female) for a recruitment rate of 68% of eligible participants. Participants received mobile tablets at a mean 6.8 days from admission [SEM 1.6], and used them for a mean 149.8 minutes/day [SEM 19.1]. In-hospital retention rate was 97%, and 96% of patients scored the mobile tablet-based communication therapy as at least moderately convenient 3/5 or better with 5/5 being most “convenient”. Conclusions Individualized speech-language therapy delivered by mobile tablet technology is feasible in acute care. PMID:28002479

RecoverNow: Feasibility of a Mobile Tablet-Based Rehabilitation Intervention to Treat Post-Stroke Communication Deficits in the Acute Care Setting.

PubMed

Mallet, Karen H; Shamloul, Rany M; Corbett, Dale; Finestone, Hillel M; Hatcher, Simon; Lumsden, Jim; Momoli, Franco; Shamy, Michel C F; Stotts, Grant; Swartz, Richard H; Yang, Christine; Dowlatshahi, Dar

2016-01-01

Approximately 40% of patients diagnosed with stroke experience some degree of aphasia. With limited health care resources, patients' access to speech and language therapies is often delayed. We propose using mobile-platform technology to initiate early speech-language therapy in the acute care setting. For this pilot, our objective was to assess the feasibility of a tablet-based speech-language therapy for patients with communication deficits following acute stroke. We enrolled consecutive patients admitted with a stroke and communication deficits with NIHSS score ≥1 on the best language and/or dysarthria parameters. We excluded patients with severe comprehension deficits where communication was not possible. Following baseline assessment by a speech-language pathologist (SLP), patients were provided with a mobile tablet programmed with individualized therapy applications based on the assessment, and instructed to use it for at least one hour per day. Our objective was to establish feasibility by measuring recruitment rate, adherence rate, retention rate, protocol deviations and acceptability. Over 6 months, 143 patients were admitted with a new diagnosis of stroke: 73 had communication deficits, 44 met inclusion criteria, and 30 were enrolled into RecoverNow (median age 62, 26.6% female) for a recruitment rate of 68% of eligible participants. Participants received mobile tablets at a mean 6.8 days from admission [SEM 1.6], and used them for a mean 149.8 minutes/day [SEM 19.1]. In-hospital retention rate was 97%, and 96% of patients scored the mobile tablet-based communication therapy as at least moderately convenient 3/5 or better with 5/5 being most "convenient". Individualized speech-language therapy delivered by mobile tablet technology is feasible in acute care.

A model of mandibular movements during speech: normative pilot study for the Brazilian Portuguese language.

PubMed

Bianchini, Esther M G; de Andrade, Cláudia R F

2006-07-01

The precision of speech articulation is related to the possibility and freedom of the mandibular movements, modifying the spaces in order to allow the different articulatory positions of each sound. Electrognathography allows the objective delineation and registration of the mandibular movements, determining the level of opening, translations and velocity of these movements. Its use is a resource that can establish quantitative diagnostic parameters. The aim of this study was to verify the amplitude, velocity and characterization of the mandibular movements during speech using computerized electrognathography. Participants were 40 adults, male and female, with no temporomandibular disorders; with no missing teeth; with no dental occlusion alterations or dentofacial deformities; with no dental prostheses; and with no communication, neurological or cognitive deficits. The mandibular movements were observed during the sequential naming of pictures containing all the phonemes of the Brazilian Portuguese language. The registrations were obtained using electrognathography (BioENG-BioPak system), assessing the spatial position, course and velocity of the mandibular movements. The mean values of velocity were: 88.65 mm/sec during opening and 89.90mm/sec during closing. The mean values of amplitude were: sagittal opening: 12.77 mm, frontal opening: 11.21 mm, protrusion: 1.22 mm; retrusion 5.67 mm; translations to the right: 1.49 mm and to the left: 1.59 mm. The velocity of opening is directly related to that of closing. The amplitude of opening demonstrates a direct correlation with the velocity of opening and closing. All participants presented lateral translations during the course of the jaw. The assessment of speech in normal individuals is characterized by: discreet mandibular movements with an anteroposterior component and lateral translations. This study allowed for the delineation of a profile of the mandibular movements during speech in asymptomatic individuals.

Do /s/-Initial Clusters Imply CVCC Sequences? Evidence from Disordered Speech

ERIC Educational Resources Information Center

Pan, Ning; Roussel, Nancye

2008-01-01

The structure of /s/-initial clusters is debated in developmental phonology. Pan and Snyder (2004) took the Government Phonology (GP) framework and proposed that production of /s/-initial clusters requires the positive setting of two binary parameters [+/-Branching rhyme (BR)] and [+/-Magic empty nucleus (MEN)] and the initial /s/ is treated as a…

Influence of Analogy Instruction for Pitch Variation on Perceptual Ratings of Other Speech Parameters

ERIC Educational Resources Information Center

Tse, Andy C. Y.; Wong, Andus W-K.; Ma, Estella P-M.; Whitehill, Tara L.; Masters, Rich S. W.

2013-01-01

Purpose: "Analogy" is the similarity of different concepts on which a comparison can be based. Recently, an analogy of "waves at sea" was shown to be effective in modulating fundamental frequency (F[subscript 0]) variation. Perceptions of intonation were not examined, as the primary aim of the work was to determine whether…

Speaker verification using committee neural networks.

PubMed

Reddy, Narender P; Buch, Ojas A

2003-10-01

Security is a major problem in web based access or remote access to data bases. In the present study, the technique of committee neural networks was developed for speech based speaker verification. Speech data from the designated speaker and several imposters were obtained. Several parameters were extracted in the time and frequency domains, and fed to neural networks. Several neural networks were trained and the five best performing networks were recruited into the committee. The committee decision was based on majority voting of the member networks. The committee opinion was evaluated with further testing data. The committee correctly identified the designated speaker in (50 out of 50) 100% of the cases and rejected imposters in (150 out of 150) 100% of the cases. The committee decision was not unanimous in majority of the cases tested.

Enhancement and Bandwidth Compression of Noisy Speech by Estimation of Speech and Its Model Parameters

DTIC Science & Technology

1978-08-01

110 V.4.1 Motivation for the Revision 110 V.4.2 Estimation of s(i)’s(j) by E[s(i) s(j) la ,v 111 V.4.3 RLMAP Estimation Procedure 114 V.5 Extension to...and .ne distinction will be left to the context in which it is used. -83- P(Sola,g,si) = p~s(N-1,0) la ,g,s(-!,-p)) N-i = i p (s (n) Ia,g, s(n-1, -p...s(n) la ,g,s(n-!,n-p)) - 1 (si) T 2 2 exp[- -I- (s(n)-a *s(n-1,n-p)) (4-5) (2"rg /2g From equations (4-4) and (4-5), a 1 NI T 2 (Ig ) 22 N/Iexp

Performance of wavelet analysis and neural networks for pathological voices identification

NASA Astrophysics Data System (ADS)

Salhi, Lotfi; Talbi, Mourad; Abid, Sabeur; Cherif, Adnane

2011-09-01

Within the medical environment, diverse techniques exist to assess the state of the voice of the patient. The inspection technique is inconvenient for a number of reasons, such as its high cost, the duration of the inspection, and above all, the fact that it is an invasive technique. This study focuses on a robust, rapid and accurate system for automatic identification of pathological voices. This system employs non-invasive, non-expensive and fully automated method based on hybrid approach: wavelet transform analysis and neural network classifier. First, we present the results obtained in our previous study while using classic feature parameters. These results allow visual identification of pathological voices. Second, quantified parameters drifting from the wavelet analysis are proposed to characterise the speech sample. On the other hand, a system of multilayer neural networks (MNNs) has been developed which carries out the automatic detection of pathological voices. The developed method was evaluated using voice database composed of recorded voice samples (continuous speech) from normophonic or dysphonic speakers. The dysphonic speakers were patients of a National Hospital 'RABTA' of Tunis Tunisia and a University Hospital in Brussels, Belgium. Experimental results indicate a success rate ranging between 75% and 98.61% for discrimination of normal and pathological voices using the proposed parameters and neural network classifier. We also compared the average classification rate based on the MNN, Gaussian mixture model and support vector machines.

78 FR 49693 - Speech-to-Speech and Internet Protocol (IP) Speech-to-Speech Telecommunications Relay Services...

Federal Register 2010, 2011, 2012, 2013, 2014

2013-08-15

...] Speech-to-Speech and Internet Protocol (IP) Speech-to-Speech Telecommunications Relay Services...: This is a summary of the Commission's Speech-to-Speech and Internet Protocol (IP) Speech-to-Speech...), Internet Protocol Relay (IP Relay), and IP captioned telephone service (IP CTS) as compensable forms of TRS...

Speech adjustments for room acoustics and their effects on vocal effort

PubMed Central

Bottalico, Pasquale

2016-01-01

Objectives The aims of the present study are: (1) to analyze the effects of the acoustical environment and the voice style on time dose (Dt_p,) and fundamental frequency (mean fo and standard deviation std_fo), while taking into account the effect of short term vocal fatigue; (2) to predict the self-reported vocal effort from the voice acoustical parameters. Methods Ten male and ten female subjects were recorded while reading a text in normal and loud styles, in three rooms - anechoic, semi-reverberant and reverberant –with and without acrylic glass panels 0.5 m from the mouth, which increased external auditory feedback. Subjects quantified how much effort was required to speak in each condition on a visual analogue scale after each task. Results (Aim1) In the loud style, Dt_p, fo and std_fo increased. The Dt_p was higher in the reverberant room compared to the other two rooms. Both genders tended to increase fo in less reverberant environments, while a more monotonous speech was produced in rooms with greater reverberation. All three voice parameters increased with short-term vocal fatigue. (Aim2) A model of the vocal effort to acoustic vocal parameters is proposed. The SPL (Sound Pressure Level) contributed to 66% of the variance explained by the model, followed by the fundamental frequency (30%) and the modulation in amplitude (4%). Conclusions The results provide insight into how voice acoustical parameters can predict vocal effort. In particular, it increased when SPL and fo increased and when the amplitude voice modulation (std_ΔSPL) decreased. PMID:28029555

Improving the speech intelligibility in classrooms

NASA Astrophysics Data System (ADS)

Lam, Choi Ling Coriolanus

One of the major acoustical concerns in classrooms is the establishment of effective verbal communication between teachers and students. Non-optimal acoustical conditions, resulting in reduced verbal communication, can cause two main problems. First, they can lead to reduce learning efficiency. Second, they can also cause fatigue, stress, vocal strain and health problems, such as headaches and sore throats, among teachers who are forced to compensate for poor acoustical conditions by raising their voices. Besides, inadequate acoustical conditions can induce the usage of public address system. Improper usage of such amplifiers or loudspeakers can lead to impairment of students' hearing systems. The social costs of poor classroom acoustics will be large to impair the learning of children. This invisible problem has far reaching implications for learning, but is easily solved. Many researches have been carried out that they have accurately and concisely summarized the research findings on classrooms acoustics. Though, there is still a number of challenging questions remaining unanswered. Most objective indices for speech intelligibility are essentially based on studies of western languages. Even several studies of tonal languages as Mandarin have been conducted, there is much less on Cantonese. In this research, measurements have been done in unoccupied rooms to investigate the acoustical parameters and characteristics of the classrooms. The speech intelligibility tests, which based on English, Mandarin and Cantonese, and the survey were carried out on students aged from 5 years old to 22 years old. It aims to investigate the differences in intelligibility between English, Mandarin and Cantonese of the classrooms in Hong Kong. The significance on speech transmission index (STI) related to Phonetically Balanced (PB) word scores will further be developed. Together with developed empirical relationship between the speech intelligibility in classrooms with the variations of the reverberation time, the indoor ambient noise (or background noise level), the signal-to-noise ratio, and the speech transmission index, it aims to establish a guideline for improving the speech intelligibility in classrooms for any countries and any environmental conditions. The study showed that the acoustical conditions of most of the measured classrooms in Hong Kong are unsatisfactory. The selection of materials inside a classroom is important for improving speech intelligibility at design stage, especially the acoustics ceiling, to shorten the reverberation time inside the classroom. The signal-to-noise should be higher than 11dB(A) for over 70% of speech perception, either tonal or non-tonal languages, without the usage of address system. The unexpected results bring out a call to revise the standard design and to devise acceptable standards for classrooms in Hong Kong. It is also demonstrated a method for assessment on the classroom in other cities with similar environmental conditions.

Universal and language-specific sublexical cues in speech perception: a novel electroencephalography-lesion approach.

PubMed

Obrig, Hellmuth; Mentzel, Julia; Rossi, Sonja

2016-06-01

SEE CAPPA DOI101093/BRAIN/AWW090 FOR A SCIENTIFIC COMMENTARY ON THIS ARTICLE  : The phonological structure of speech supports the highly automatic mapping of sound to meaning. While it is uncontroversial that phonotactic knowledge acts upon lexical access, it is unclear at what stage these combinatorial rules, governing phonological well-formedness in a given language, shape speech comprehension. Moreover few studies have investigated the neuronal network affording this important step in speech comprehension. Therefore we asked 70 participants-half of whom suffered from a chronic left hemispheric lesion-to listen to 252 different monosyllabic pseudowords. The material models universal preferences of phonotactic well-formedness by including naturally spoken pseudowords and digitally reversed exemplars. The latter partially violate phonological structure of all human speech and are rich in universally dispreferred phoneme sequences while preserving basic auditory parameters. Language-specific constraints were modelled in that half of the naturally spoken pseudowords complied with the phonotactics of the native language of the monolingual participants (German) while the other half did not. To ensure universal well-formedness and naturalness, the latter stimuli comply with Slovak phonotactics and all stimuli were produced by an early bilingual speaker. To maximally attenuate lexico-semantic influences, transparent pseudowords were avoided and participants had to detect immediate repetitions, a task orthogonal to the contrasts of interest. The results show that phonological 'well-formedness' modulates implicit processing of speech at different levels: universally dispreferred phonological structure elicits early, medium and late latency differences in the evoked potential. On the contrary, the language-specific phonotactic contrast selectively modulates a medium latency component of the event-related potentials around 400 ms. Using a novel event-related potential-lesion approach allowed us to furthermore supply first evidence that implicit processing of these different phonotactic levels relies on partially separable brain areas in the left hemisphere: contrasting forward to reversed speech the approach delineated an area comprising supramarginal and angular gyri. Conversely, the contrast between legal versus illegal phonotactics consistently projected to anterior and middle portions of the middle temporal and superior temporal gyri. Our data support the notion that phonological structure acts on different stages of phonologically and lexically driven steps of speech comprehension. In the context of previous work we propose context-dependent sensitivity to different levels of phonotactic well-formedness. © The Author (2016). Published by Oxford University Press on behalf of the Guarantors of Brain. All rights reserved. For Permissions, please email: journals.permissions@oup.com.

Perceptual evaluation and acoustic analysis of pneumatic artificial larynx.

PubMed

Xu, Jie Jie; Chen, Xi; Lu, Mei Ping; Qiao, Ming Zhe

2009-12-01

To investigate the perceptual and acoustic characteristics of the pneumatic artificial larynx (PAL) and evaluate its speech ability and clinical value. Prospective study. The study was conducted in the Voice Lab, Department of Otorhinolaryngology, The First Affiliated Hospital of Nanjing Medical University. Forty-six laryngectomy patients using the PAL were rated for intelligibility and fluency of speech. The voice signals of sustained vowel /a/ for 40 healthy controls and 42 successful patients using the PAL were measured by a computer system. The acoustic parameters and sound spectrographs were analyzed and compared between the two groups. Forty-two of 46 patients using the PAL (91.3%) acquired successful speech capability. The intelligibility scores of 42 successful PAL speakers ranged from 71 to 95 percent, and the intelligibility range of four unsuccessful speakers was 30 to 50 percent. The fluency was judged as good or excellent in 42 successful patients, and poor or fair in four unsuccessful patients. There was no significant difference in average fundamental frequency, maximum intensity, jitter, shimmer, and normalized noise energy (NNE) between 42 successful PAL speakers and 40 healthy controls, while the maximum phonation time (MPT) of PAL speakers was slightly lower than that of the controls. The sound spectrographs of the patients using the PAL approximated those of the healthy controls. The PAL has the advantage of a high percentage of successful vocal rehabilitation. PAL speech is fluent and intelligible. The acoustic characteristics of the PAL are similar to those of a normal voice.

A simulation framework for auditory discrimination experiments: Revealing the importance of across-frequency processing in speech perception.

PubMed

Schädler, Marc René; Warzybok, Anna; Ewert, Stephan D; Kollmeier, Birger

2016-05-01

A framework for simulating auditory discrimination experiments, based on an approach from Schädler, Warzybok, Hochmuth, and Kollmeier [(2015). Int. J. Audiol. 54, 100-107] which was originally designed to predict speech recognition thresholds, is extended to also predict psychoacoustic thresholds. The proposed framework is used to assess the suitability of different auditory-inspired feature sets for a range of auditory discrimination experiments that included psychoacoustic as well as speech recognition experiments in noise. The considered experiments were 2 kHz tone-in-broadband-noise simultaneous masking depending on the tone length, spectral masking with simultaneously presented tone signals and narrow-band noise maskers, and German Matrix sentence test reception threshold in stationary and modulated noise. The employed feature sets included spectro-temporal Gabor filter bank features, Mel-frequency cepstral coefficients, logarithmically scaled Mel-spectrograms, and the internal representation of the Perception Model from Dau, Kollmeier, and Kohlrausch [(1997). J. Acoust. Soc. Am. 102(5), 2892-2905]. The proposed framework was successfully employed to simulate all experiments with a common parameter set and obtain objective thresholds with less assumptions compared to traditional modeling approaches. Depending on the feature set, the simulated reference-free thresholds were found to agree with-and hence to predict-empirical data from the literature. Across-frequency processing was found to be crucial to accurately model the lower speech reception threshold in modulated noise conditions than in stationary noise conditions.

Parameterization of the Voice Source by Combining Spectral Decay and Amplitude Features of the Glottal Flow.

ERIC Educational Resources Information Center

Alku, Paavo; Vilkman, Erkki; Laukkanen, Anne-Maria

1998-01-01

A new method is presented for the parameterization of glottal volume velocity waveforms that have been estimated by inverse filtering acoustic speech pressure signals. The new technique combines two features of voice production: the AC value and the spectral decay of the glottal flow. Testing found the new parameter correlates strongly with the…

Brief Report: Dysregulated Immune System in Children with Autism: Beneficial Effects of Intravenous Immune Globulin on Autistic Characteristics.

ERIC Educational Resources Information Center

Gupta, Sudhir; And Others

1996-01-01

Children (ages 3-12) with autism (n=25) were given intravenous immune globulin (IVIG) treatments at 4-week intervals for at least 6 months. Marked abnormality of immune parameters was observed in subjects, compared to age-matched controls. IVIG treatment resulted in improved eye contact, speech, behavior, echolalia, and other autistic features.…

Linear degrees of freedom in speech production: analysis of cineradio- and labio-film data and articulatory-acoustic modeling.

PubMed

Beautemps, D; Badin, P; Bailly, G

2001-05-01

The following contribution addresses several issues concerning speech degrees of freedom in French oral vowels, stop, and fricative consonants based on an analysis of tongue and lip shapes extracted from cineradio- and labio-films. The midsagittal tongue shapes have been submitted to a linear decomposition where some of the loading factors were selected such as jaw and larynx position while four other components were derived from principal component analysis (PCA). For the lips, in addition to the more traditional protrusion and opening components, a supplementary component was extracted to explain the upward movement of both the upper and lower lips in [v] production. A linear articulatory model was developed; the six tongue degrees of freedom were used as the articulatory control parameters of the midsagittal tongue contours and explained 96% of the tongue data variance. These control parameters were also used to specify the frontal lip width dimension derived from the labio-film front views. Finally, this model was complemented by a conversion model going from the midsagittal to the area function, based on a fitting of the midsagittal distances and the formant frequencies for both vowels and consonants.

Logarithmic temporal axis manipulation and its application for measuring auditory contributions in F0 control using a transformed auditory feedback procedure

NASA Astrophysics Data System (ADS)

Yanaga, Ryuichiro; Kawahara, Hideki

2003-10-01

A new parameter extraction procedure based on logarithmic transformation of the temporal axis was applied to investigate auditory effects on voice F0 control to overcome artifacts due to natural fluctuations and nonlinearities in speech production mechanisms. The proposed method may add complementary information to recent findings reported by using frequency shift feedback method [Burnett and Larson, J. Acoust. Soc. Am. 112 (2002)], in terms of dynamic aspects of F0 control. In a series of experiments, dependencies of system parameters in F0 control on subjects, F0 and style (musical expressions and speaking) were tested using six participants. They were three male and three female students specialized in musical education. They were asked to sustain a Japanese vowel /a/ for about 10 s repeatedly up to 2 min in total while hearing F0 modulated feedback speech, that was modulated using an M-sequence. The results replicated qualitatively the previous finding [Kawahara and Williams, Vocal Fold Physiology, (1995)] and provided more accurate estimates. Relations with designing an artificial singer also will be discussed. [Work partly supported by the grant in aids in scientific research (B) 14380165 and Wakayama University.

Fundamental frequency estimation of singing voice

NASA Astrophysics Data System (ADS)

de Cheveigné, Alain; Henrich, Nathalie

2002-05-01

A method of fundamental frequency (F0) estimation recently developped for speech [de Cheveigné and Kawahara, J. Acoust. Soc. Am. (to be published)] was applied to singing voice. An electroglottograph signal recorded together with the microphone provided a reference by which estimates could be validated. Using standard parameter settings as for speech, error rates were low despite the wide range of F0s (about 100 to 1600 Hz). Most ``errors'' were due to irregular vibration of the vocal folds, a sharp formant resonance that reduced the waveform to a single harmonic, or fast F0 changes such as in high-amplitude vibrato. Our database (18 singers from baritone to soprano) included examples of diphonic singing for which melody is carried by variations of the frequency of a narrow formant rather than F0. Varying a parameter (ratio of inharmonic to total power) the algorithm could be tuned to follow either frequency. Although the method has not been formally tested on a wide range of instruments, it seems appropriate for musical applications because it is accurate, accepts a wide range of F0s, and can be implemented with low latency for interactive applications. [Work supported by the Cognitique programme of the French Ministry of Research and Technology.

Analysis and synthesis of laughter

NASA Astrophysics Data System (ADS)

Sundaram, Shiva; Narayanan, Shrikanth

2004-10-01

There is much enthusiasm in the text-to-speech community for synthesis of emotional and natural speech. One idea being proposed is to include emotion dependent paralinguistic cues during synthesis to convey emotions effectively. This requires modeling and synthesis techniques of various cues for different emotions. Motivated by this, a technique to synthesize human laughter is proposed. Laughter is a complex mechanism of expression and has high variability in terms of types and usage in human-human communication. People have their own characteristic way of laughing. Laughter can be seen as a controlled/uncontrolled physiological process of a person resulting from an initial excitation in context. A parametric model based on damped simple harmonic motion to effectively capture these diversities and also maintain the individuals characteristics is developed here. Limited laughter/speech data from actual humans and synthesis ease are the constraints imposed on the accuracy of the model. Analysis techniques are also developed to determine the parameters of the model for a given individual or laughter type. Finally, the effectiveness of the model to capture the individual characteristics and naturalness compared to real human laughter has been analyzed. Through this the factors involved in individual human laughter and their importance can be better understood.

Non-fluent aphasia and neural reorganization after speech therapy: insights from human sleep electrophysiology and functional magnetic resonance imaging.

PubMed

Sarasso, S; Santhanam, P; Määtta, S; Poryazova, R; Ferrarelli, F; Tononi, G; Small, S L

2010-09-01

Stroke is associated with long-term functional deficits. Behavioral interventions are often effective in promoting functional recovery and plastic changes. Recent studies in normal subjects have shown that sleep, and particularly slow wave activity (SWA), is tied to local brain plasticity and may be used as a sensitive marker of local cortical reorganization after stroke. In a pilot study, we assessed the local changes induced by a single exposure to a therapeutic session of IMITATE (Intensive Mouth Imitation and Talking for Aphasia Therapeutic Effects), a behavioral therapy used for recovery in patients with post-stroke aphasia. In addition, we measured brain activity changes with functional magnetic resonance imaging (fMRI) in a language observation task before, during and after the full IMITATE rehabilitative program. Speech production improved both after a single exposure and the full therapy program as measured by the Western Aphasia Battery (WAB) Repetition subscale. We found that IMITATE induced reorganization in functionally-connected, speech-relevant areas in the left hemisphere. These preliminary results suggest that sleep hd-EEGs, and the topographical analysis of SWA parameters, are well suited to investigate brain plastic changes underpinning functional recovery in neurological disorders.

On Short-Time Estimation of Vocal Tract Length from Formant Frequencies

PubMed Central

Lammert, Adam C.; Narayanan, Shrikanth S.

2015-01-01

Vocal tract length is highly variable across speakers and determines many aspects of the acoustic speech signal, making it an essential parameter to consider for explaining behavioral variability. A method for accurate estimation of vocal tract length from formant frequencies would afford normalization of interspeaker variability and facilitate acoustic comparisons across speakers. A framework for considering estimation methods is developed from the basic principles of vocal tract acoustics, and an estimation method is proposed that follows naturally from this framework. The proposed method is evaluated using acoustic characteristics of simulated vocal tracts ranging from 14 to 19 cm in length, as well as real-time magnetic resonance imaging data with synchronous audio from five speakers whose vocal tracts range from 14.5 to 18.0 cm in length. Evaluations show improvements in accuracy over previously proposed methods, with 0.631 and 1.277 cm root mean square error on simulated and human speech data, respectively. Empirical results show that the effectiveness of the proposed method is based on emphasizing higher formant frequencies, which seem less affected by speech articulation. Theoretical predictions of formant sensitivity reinforce this empirical finding. Moreover, theoretical insights are explained regarding the reason for differences in formant sensitivity. PMID:26177102

Low-level neural auditory discrimination dysfunctions in specific language impairment-A review on mismatch negativity findings.

PubMed

Kujala, Teija; Leminen, Miika

2017-12-01

In specific language impairment (SLI), there is a delay in the child's oral language skills when compared with nonverbal cognitive abilities. The problems typically relate to phonological and morphological processing and word learning. This article reviews studies which have used mismatch negativity (MMN) in investigating low-level neural auditory dysfunctions in this disorder. With MMN, it is possible to tap the accuracy of neural sound discrimination and sensory memory functions. These studies have found smaller response amplitudes and longer latencies for speech and non-speech sound changes in children with SLI than in typically developing children, suggesting impaired and slow auditory discrimination in SLI. Furthermore, they suggest shortened sensory memory duration and vulnerability of the sensory memory to masking effects. Importantly, some studies reported associations between MMN parameters and language test measures. In addition, it was found that language intervention can influence the abnormal MMN in children with SLI, enhancing its amplitude. These results suggest that the MMN can shed light on the neural basis of various auditory and memory impairments in SLI, which are likely to influence speech perception. Copyright © 2017. Published by Elsevier Ltd.

The effect of deep brain stimulation on the speech motor system.

PubMed

Mücke, Doris; Becker, Johannes; Barbe, Michael T; Meister, Ingo; Liebhart, Lena; Roettger, Timo B; Dembek, Till; Timmermann, Lars; Grice, Martine

2014-08-01

Chronic deep brain stimulation of the nucleus ventralis intermedius is an effective treatment for individuals with medication-resistant essential tremor. However, these individuals report that stimulation has a deleterious effect on their speech. The present study investigates one important factor leading to these effects: the coordination of oral and glottal articulation. Sixteen native-speaking German adults with essential tremor, between 26 and 86 years old, with and without chronic deep brain stimulation of the nucleus ventralis intermedius and 12 healthy, age-matched subjects were recorded performing a fast syllable repetition task (/papapa/, /tatata/, /kakaka/). Syllable duration and voicing-to-syllable ratio as well as parameters related directly to consonant production, voicing during constriction, and frication during constriction were measured. Voicing during constriction was greater in subjects with essential tremor than in controls, indicating a perseveration of voicing into the voiceless consonant. Stimulation led to fewer voiceless intervals (voicing-to-syllable ratio), indicating a reduced degree of glottal abduction during the entire syllable cycle. Stimulation also induced incomplete oral closures (frication during constriction), indicating imprecise oral articulation. The detrimental effect of stimulation on the speech motor system can be quantified using acoustic measures at the subsyllabic level.

Spatial resolution dependence on spectral frequency in human speech cortex electrocorticography.

PubMed

Muller, Leah; Hamilton, Liberty S; Edwards, Erik; Bouchard, Kristofer E; Chang, Edward F

2016-10-01

Electrocorticography (ECoG) has become an important tool in human neuroscience and has tremendous potential for emerging applications in neural interface technology. Electrode array design parameters are outstanding issues for both research and clinical applications, and these parameters depend critically on the nature of the neural signals to be recorded. Here, we investigate the functional spatial resolution of neural signals recorded at the human cortical surface. We empirically derive spatial spread functions to quantify the shared neural activity for each frequency band of the electrocorticogram. Five subjects with high-density (4 mm center-to-center spacing) ECoG grid implants participated in speech perception and production tasks while neural activity was recorded from the speech cortex, including superior temporal gyrus, precentral gyrus, and postcentral gyrus. The cortical surface field potential was decomposed into traditional EEG frequency bands. Signal similarity between electrode pairs for each frequency band was quantified using a Pearson correlation coefficient. The correlation of neural activity between electrode pairs was inversely related to the distance between the electrodes; this relationship was used to quantify spatial falloff functions for cortical subdomains. As expected, lower frequencies remained correlated over larger distances than higher frequencies. However, both the envelope and phase of gamma and high gamma frequencies (30-150 Hz) are largely uncorrelated (<90%) at 4 mm, the smallest spacing of the high-density arrays. Thus, ECoG arrays smaller than 4 mm have significant promise for increasing signal resolution at high frequencies, whereas less additional gain is achieved for lower frequencies. Our findings quantitatively demonstrate the dependence of ECoG spatial resolution on the neural frequency of interest. We demonstrate that this relationship is consistent across patients and across cortical areas during activity.

Spatial resolution dependence on spectral frequency in human speech cortex electrocorticography

NASA Astrophysics Data System (ADS)

Muller, Leah; Hamilton, Liberty S.; Edwards, Erik; Bouchard, Kristofer E.; Chang, Edward F.

2016-10-01

Objective. Electrocorticography (ECoG) has become an important tool in human neuroscience and has tremendous potential for emerging applications in neural interface technology. Electrode array design parameters are outstanding issues for both research and clinical applications, and these parameters depend critically on the nature of the neural signals to be recorded. Here, we investigate the functional spatial resolution of neural signals recorded at the human cortical surface. We empirically derive spatial spread functions to quantify the shared neural activity for each frequency band of the electrocorticogram. Approach. Five subjects with high-density (4 mm center-to-center spacing) ECoG grid implants participated in speech perception and production tasks while neural activity was recorded from the speech cortex, including superior temporal gyrus, precentral gyrus, and postcentral gyrus. The cortical surface field potential was decomposed into traditional EEG frequency bands. Signal similarity between electrode pairs for each frequency band was quantified using a Pearson correlation coefficient. Main results. The correlation of neural activity between electrode pairs was inversely related to the distance between the electrodes; this relationship was used to quantify spatial falloff functions for cortical subdomains. As expected, lower frequencies remained correlated over larger distances than higher frequencies. However, both the envelope and phase of gamma and high gamma frequencies (30-150 Hz) are largely uncorrelated (<90%) at 4 mm, the smallest spacing of the high-density arrays. Thus, ECoG arrays smaller than 4 mm have significant promise for increasing signal resolution at high frequencies, whereas less additional gain is achieved for lower frequencies. Significance. Our findings quantitatively demonstrate the dependence of ECoG spatial resolution on the neural frequency of interest. We demonstrate that this relationship is consistent across patients and across cortical areas during activity.

Cepstral analysis of normal and pathological voice in Spanish adults. Smoothed cepstral peak prominence in sustained vowels versus connected speech.

PubMed

Delgado-Hernández, Jonathan; León-Gómez, Nieves M; Izquierdo-Arteaga, Laura M; Llanos-Fumero, Yanira

In recent years, the use of cepstral measures for acoustic evaluation of voice has increased. One of the most investigated parameters is smoothed cepstral peak prominence (CPPs). The objectives of this paper are to establish the usefulness of this acoustic measure in the objective evaluation of alterations of the voice in Spanish and to determine what type of voice sample (sustained vowels or connected speech) is the most sensitive in evaluating the severity of dysphonia. Forty subjects participated in this study 40, 20 controls and 20 with dysphonia. Two voice samples were recorded for each subject (one sustained vowel/a/and four phonetically balanced sentences) and the CPPs was calculated using the Praat programme. Three raters perceptually evaluated the voice sample with the Grade parameter of GRABS scale. Significantly lower values were found in the dysphonic voices, both for/a/(t [38] = 4.85, P<.000) and for phrases (t [38] = 5,75, P<.000). In relation to the type of voice sample most suitable for evaluating the severity of voice alterations, a strong correlation was found with the acoustic-perceptual scale of CPPs calculated from connected speech (r s = -0.73) and moderate correlation with that calculated from the sustained vowel (r s = -0,56). The results of this preliminary study suggest that CPPs is a good measure to detect dysphonia and to objectively assess the severity of alterations in the voice. Copyright © 2017 Elsevier España, S.L.U. and Sociedad Española de Otorrinolaringología y Cirugía de Cabeza y Cuello. All rights reserved.

78 FR 49717 - Speech-to-Speech and Internet Protocol (IP) Speech-to-Speech Telecommunications Relay Services...

Federal Register 2010, 2011, 2012, 2013, 2014

2013-08-15

...] Speech-to-Speech and Internet Protocol (IP) Speech-to-Speech Telecommunications Relay Services... Internet Protocol (IP) Speech-to-Speech Telecommunications Relay Services; Telecommunications Relay... (IP Relay) and video relay service (VRS), the Commission should bundle national STS outreach efforts...

An Efficient VLSI Architecture of the Enhanced Three Step Search Algorithm

NASA Astrophysics Data System (ADS)

Biswas, Baishik; Mukherjee, Rohan; Saha, Priyabrata; Chakrabarti, Indrajit

2016-09-01

The intense computational complexity of any video codec is largely due to the motion estimation unit. The Enhanced Three Step Search is a popular technique that can be adopted for fast motion estimation. This paper proposes a novel VLSI architecture for the implementation of the Enhanced Three Step Search Technique. A new addressing mechanism has been introduced which enhances the speed of operation and reduces the area requirements. The proposed architecture when implemented in Verilog HDL on Virtex-5 Technology and synthesized using Xilinx ISE Design Suite 14.1 achieves a critical path delay of 4.8 ns while the area comes out to be 2.9K gate equivalent. It can be incorporated in commercial devices like smart-phones, camcorders, video conferencing systems etc.

Direct migration motion estimation and mode decision to decoder for a low-complexity decoder Wyner-Ziv video coding

NASA Astrophysics Data System (ADS)

Lei, Ted Chih-Wei; Tseng, Fan-Shuo

2017-07-01

This paper addresses the problem of high-computational complexity decoding in traditional Wyner-Ziv video coding (WZVC). The key focus is the migration of two traditionally high-computationally complex encoder algorithms, namely motion estimation and mode decision. In order to reduce the computational burden in this process, the proposed architecture adopts the partial boundary matching algorithm and four flexible types of block mode decision at the decoder. This approach does away with the need for motion estimation and mode decision at the encoder. The experimental results show that the proposed padding block-based WZVC not only decreases decoder complexity to approximately one hundredth that of the state-of-the-art DISCOVER decoding but also outperforms DISCOVER codec by up to 3 to 4 dB.

Speech coding, reconstruction and recognition using acoustics and electromagnetic waves

DOEpatents

Holzrichter, J.F.; Ng, L.C.

1998-03-17

The use of EM radiation in conjunction with simultaneously recorded acoustic speech information enables a complete mathematical coding of acoustic speech. The methods include the forming of a feature vector for each pitch period of voiced speech and the forming of feature vectors for each time frame of unvoiced, as well as for combined voiced and unvoiced speech. The methods include how to deconvolve the speech excitation function from the acoustic speech output to describe the transfer function each time frame. The formation of feature vectors defining all acoustic speech units over well defined time frames can be used for purposes of speech coding, speech compression, speaker identification, language-of-speech identification, speech recognition, speech synthesis, speech translation, speech telephony, and speech teaching. 35 figs.

Speech coding, reconstruction and recognition using acoustics and electromagnetic waves

DOEpatents

Holzrichter, John F.; Ng, Lawrence C.

1998-01-01

The use of EM radiation in conjunction with simultaneously recorded acoustic speech information enables a complete mathematical coding of acoustic speech. The methods include the forming of a feature vector for each pitch period of voiced speech and the forming of feature vectors for each time frame of unvoiced, as well as for combined voiced and unvoiced speech. The methods include how to deconvolve the speech excitation function from the acoustic speech output to describe the transfer function each time frame. The formation of feature vectors defining all acoustic speech units over well defined time frames can be used for purposes of speech coding, speech compression, speaker identification, language-of-speech identification, speech recognition, speech synthesis, speech translation, speech telephony, and speech teaching.

Exploring Australian speech-language pathologists' use and perceptions ofnon-speech oral motor exercises.

PubMed

Rumbach, Anna F; Rose, Tanya A; Cheah, Mynn

2018-01-29

To explore Australian speech-language pathologists' use of non-speech oral motor exercises, and rationales for using/not using non-speech oral motor exercises in clinical practice. A total of 124 speech-language pathologists practising in Australia, working with paediatric and/or adult clients with speech sound difficulties, completed an online survey. The majority of speech-language pathologists reported that they did not use non-speech oral motor exercises when working with paediatric or adult clients with speech sound difficulties. However, more than half of the speech-language pathologists working with adult clients who have dysarthria reported using non-speech oral motor exercises with this population. The most frequently reported rationale for using non-speech oral motor exercises in speech sound difficulty management was to improve awareness/placement of articulators. The majority of speech-language pathologists agreed there is no clear clinical or research evidence base to support non-speech oral motor exercise use with clients who have speech sound difficulties. This study provides an overview of Australian speech-language pathologists' reported use and perceptions of non-speech oral motor exercises' applicability and efficacy in treating paediatric and adult clients who have speech sound difficulties. The research findings provide speech-language pathologists with insight into how and why non-speech oral motor exercises are currently used, and adds to the knowledge base regarding Australian speech-language pathology practice of non-speech oral motor exercises in the treatment of speech sound difficulties. Implications for Rehabilitation Non-speech oral motor exercises refer to oral motor activities which do not involve speech, but involve the manipulation or stimulation of oral structures including the lips, tongue, jaw, and soft palate. Non-speech oral motor exercises are intended to improve the function (e.g., movement, strength) of oral structures. The majority of speech-language pathologists agreed there is no clear clinical or research evidence base to support non-speech oral motor exercise use with clients who have speech sound disorders. Non-speech oral motor exercise use was most frequently reported in the treatment of dysarthria. Non-speech oral motor exercise use when targeting speech sound disorders is not widely endorsed in the literature.

Speech coding, reconstruction and recognition using acoustics and electromagnetic waves

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

Holzrichter, J.F.; Ng, L.C.

The use of EM radiation in conjunction with simultaneously recorded acoustic speech information enables a complete mathematical coding of acoustic speech. The methods include the forming of a feature vector for each pitch period of voiced speech and the forming of feature vectors for each time frame of unvoiced, as well as for combined voiced and unvoiced speech. The methods include how to deconvolve the speech excitation function from the acoustic speech output to describe the transfer function each time frame. The formation of feature vectors defining all acoustic speech units over well defined time frames can be used formore » purposes of speech coding, speech compression, speaker identification, language-of-speech identification, speech recognition, speech synthesis, speech translation, speech telephony, and speech teaching. 35 figs.« less

Analysis of Optimal Jitter Buffer Size for VoIP QoS under WiMAX Power-Saving Mode

NASA Astrophysics Data System (ADS)

Kim, Hyungsuk; Kim, Taehyoun

VoIP service is expected as one of the key applications of Mobile WiMAX, but the speech quality of VoIP service often suffers deterioration due to the fluctuating transmission delay called jitter. This is commonly ameliorated by a de-jitter buffer, and we aim to find the optimal size of de-jitter buffer to achieve speech quality comparable to PSTN. We developed a new model of the packet drops at the de-jitter buffer and the end-to-end packet delay which takes account of the additional delay introduced by the WiMAX power-saving mode. Using our model, we analyzed the optimal size of the de-jitter buffer for various network parameters, and showed that the results obtained by analysis accord with simulation results.

[Speech rhythm disorders due to synchronization induced in coupled inhibitory neurons].

PubMed

Skliarov, O P

2007-01-01

Leaked-integrate-and-fire coupled oscillators (LIFs) were used as a model of electrophysiological activity. The activity of these oscillators determines the speech rhythm, which is governed by the square-law map with inhibition as a controlling parameter. Regular rhythms of convulsive repetitions at early stuttering are changed, however, by a mixture of repetitions and neurotic pauses. This mixture is a "stumbling block" for clinicians. Due to delays, only inhibitory LIFs are capable to create the synchronic activity in-phase or in-anti-phase at medial or at low coupling. This activity has the form of slow oscillations damping to the background level. Splashes of the activity above or below the level lead to neurotic disorders or to convulsive repetitions. Really, increased due to GABA the coupling leads to a reduction of stuttering.

Acoustic analysis of speech variables during depression and after improvement.

PubMed

Nilsonne, A

1987-09-01

Speech recordings were made of 16 depressed patients during depression and after clinical improvement. The recordings were analyzed using a computer program which extracts acoustic parameters from the fundamental frequency contour of the voice. The percent pause time, the standard deviation of the voice fundamental frequency distribution, the standard deviation of the rate of change of the voice fundamental frequency and the average speed of voice change were found to correlate to the clinical state of the patient. The mean fundamental frequency, the total reading time and the average rate of change of the voice fundamental frequency did not differ between the depressed and the improved group. The acoustic measures were more strongly correlated to the clinical state of the patient as measured by global depression scores than to single depressive symptoms such as retardation or agitation.

Physiological, Psychological, and Social Effects of Noise

NASA Technical Reports Server (NTRS)

Kryter, K. D.

1984-01-01

The physiological, and behavioral effects of noise on man are investigated. Basic parameters such as definitions of noise, measuring techniques of noise, and the physiology of the ear are presented prior to the development of topics on hearing loss, speech communication in noise, social effects of noise, and the health effects of noise pollution. Recommendations for the assessment and subsequent control of noise is included.

Speech processing using maximum likelihood continuity mapping

DOEpatents

Hogden, John E.

2000-01-01

Speech processing is obtained that, given a probabilistic mapping between static speech sounds and pseudo-articulator positions, allows sequences of speech sounds to be mapped to smooth sequences of pseudo-articulator positions. In addition, a method for learning a probabilistic mapping between static speech sounds and pseudo-articulator position is described. The method for learning the mapping between static speech sounds and pseudo-articulator position uses a set of training data composed only of speech sounds. The said speech processing can be applied to various speech analysis tasks, including speech recognition, speaker recognition, speech coding, speech synthesis, and voice mimicry.

Speech processing using maximum likelihood continuity mapping

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

Hogden, J.E.

Speech processing is obtained that, given a probabilistic mapping between static speech sounds and pseudo-articulator positions, allows sequences of speech sounds to be mapped to smooth sequences of pseudo-articulator positions. In addition, a method for learning a probabilistic mapping between static speech sounds and pseudo-articulator position is described. The method for learning the mapping between static speech sounds and pseudo-articulator position uses a set of training data composed only of speech sounds. The said speech processing can be applied to various speech analysis tasks, including speech recognition, speaker recognition, speech coding, speech synthesis, and voice mimicry.

[Acoustic and aerodynamic characteristics of the oesophageal voice].

PubMed

Vázquez de la Iglesia, F; Fernández González, S

2005-12-01

The aim of the study is to determine the physiology and pathophisiology of esophageal voice according to objective aerodynamic and acoustic parameters (quantitative and qualitative parameters). Our subjects were comprised of 33 laryngectomized patients (all male) that underwent aerodynamic, acoustic and perceptual protocol. There is a statistical association between acoustic and aerodynamic qualitative parameters (phonation flow chart type, sound spectrum, perceptual analysis) among quantitative parameters (neoglotic pressure, phonation flow, phonation time, fundamental frequency, maximum intensity sound level, speech rate). Nevertheles, not always such observations bring practical resources to clinical practice. We consider that the facts studied may enable us to add, pragmatically, new resources to the more effective vocal rehabilitation to these patients. The physiology of esophageal voice is well understood by the method we have applied, also seeking for rehabilitation, improving oral communication skills in the laryngectomee population.

Novel modes and adaptive block scanning order for intra prediction in AV1

NASA Astrophysics Data System (ADS)

Hadar, Ofer; Shleifer, Ariel; Mukherjee, Debargha; Joshi, Urvang; Mazar, Itai; Yuzvinsky, Michael; Tavor, Nitzan; Itzhak, Nati; Birman, Raz

2017-09-01

The demand for streaming video content is on the rise and growing exponentially. Networks bandwidth is very costly and therefore there is a constant effort to improve video compression rates and enable the sending of reduced data volumes while retaining quality of experience (QoE). One basic feature that utilizes the spatial correlation of pixels for video compression is Intra-Prediction, which determines the codec's compression efficiency. Intra prediction enables significant reduction of the Intra-Frame (I frame) size and, therefore, contributes to efficient exploitation of bandwidth. In this presentation, we propose new Intra-Prediction algorithms that improve the AV1 prediction model and provide better compression ratios. Two (2) types of methods are considered: )1( New scanning order method that maximizes spatial correlation in order to reduce prediction error; and )2( New Intra-Prediction modes implementation in AVI. Modern video coding standards, including AVI codec, utilize fixed scan orders in processing blocks during intra coding. The fixed scan orders typically result in residual blocks with high prediction error mainly in blocks with edges. This means that the fixed scan orders cannot fully exploit the content-adaptive spatial correlations between adjacent blocks, thus the bitrate after compression tends to be large. To reduce the bitrate induced by inaccurate intra prediction, the proposed approach adaptively chooses the scanning order of blocks according to criteria of firstly predicting blocks with maximum number of surrounding, already Inter-Predicted blocks. Using the modified scanning order method and the new modes has reduced the MSE by up to five (5) times when compared to conventional TM mode / Raster scan and up to two (2) times when compared to conventional CALIC mode / Raster scan, depending on the image characteristics (which determines the percentage of blocks predicted with Inter-Prediction, which in turn impacts the efficiency of the new scanning method). For the same cases, the PSNR was shown to improve by up to 7.4dB and up to 4 dB, respectively. The new modes have yielded 5% improvement in BD-Rate over traditionally used modes, when run on K-Frame, which is expected to yield 1% of overall improvement.

Real-time demonstration hardware for enhanced DPCM video compression algorithm

NASA Technical Reports Server (NTRS)

Bizon, Thomas P.; Whyte, Wayne A., Jr.; Marcopoli, Vincent R.

1992-01-01

The lack of available wideband digital links as well as the complexity of implementation of bandwidth efficient digital video CODECs (encoder/decoder) has worked to keep the cost of digital television transmission too high to compete with analog methods. Terrestrial and satellite video service providers, however, are now recognizing the potential gains that digital video compression offers and are proposing to incorporate compression systems to increase the number of available program channels. NASA is similarly recognizing the benefits of and trend toward digital video compression techniques for transmission of high quality video from space and therefore, has developed a digital television bandwidth compression algorithm to process standard National Television Systems Committee (NTSC) composite color television signals. The algorithm is based on differential pulse code modulation (DPCM), but additionally utilizes a non-adaptive predictor, non-uniform quantizer and multilevel Huffman coder to reduce the data rate substantially below that achievable with straight DPCM. The non-adaptive predictor and multilevel Huffman coder combine to set this technique apart from other DPCM encoding algorithms. All processing is done on a intra-field basis to prevent motion degradation and minimize hardware complexity. Computer simulations have shown the algorithm will produce broadcast quality reconstructed video at an average transmission rate of 1.8 bits/pixel. Hardware implementation of the DPCM circuit, non-adaptive predictor and non-uniform quantizer has been completed, providing realtime demonstration of the image quality at full video rates. Video sampling/reconstruction circuits have also been constructed to accomplish the analog video processing necessary for the real-time demonstration. Performance results for the completed hardware compare favorably with simulation results. Hardware implementation of the multilevel Huffman encoder/decoder is currently under development along with implementation of a buffer control algorithm to accommodate the variable data rate output of the multilevel Huffman encoder. A video CODEC of this type could be used to compress NTSC color television signals where high quality reconstruction is desirable (e.g., Space Station video transmission, transmission direct-to-the-home via direct broadcast satellite systems or cable television distribution to system headends and direct-to-the-home).

Schizophrenia alters intra-network functional connectivity in the caudate for detecting speech under informational speech masking conditions.

PubMed

Zheng, Yingjun; Wu, Chao; Li, Juanhua; Li, Ruikeng; Peng, Hongjun; She, Shenglin; Ning, Yuping; Li, Liang

2018-04-04

Speech recognition under noisy "cocktail-party" environments involves multiple perceptual/cognitive processes, including target detection, selective attention, irrelevant signal inhibition, sensory/working memory, and speech production. Compared to health listeners, people with schizophrenia are more vulnerable to masking stimuli and perform worse in speech recognition under speech-on-speech masking conditions. Although the schizophrenia-related speech-recognition impairment under "cocktail-party" conditions is associated with deficits of various perceptual/cognitive processes, it is crucial to know whether the brain substrates critically underlying speech detection against informational speech masking are impaired in people with schizophrenia. Using functional magnetic resonance imaging (fMRI), this study investigated differences between people with schizophrenia (n = 19, mean age = 33 ± 10 years) and their matched healthy controls (n = 15, mean age = 30 ± 9 years) in intra-network functional connectivity (FC) specifically associated with target-speech detection under speech-on-speech-masking conditions. The target-speech detection performance under the speech-on-speech-masking condition in participants with schizophrenia was significantly worse than that in matched healthy participants (healthy controls). Moreover, in healthy controls, but not participants with schizophrenia, the strength of intra-network FC within the bilateral caudate was positively correlated with the speech-detection performance under the speech-masking conditions. Compared to controls, patients showed altered spatial activity pattern and decreased intra-network FC in the caudate. In people with schizophrenia, the declined speech-detection performance under speech-on-speech masking conditions is associated with reduced intra-caudate functional connectivity, which normally contributes to detecting target speech against speech masking via its functions of suppressing masking-speech signals.

Speech Adjustments for Room Acoustics and Their Effects on Vocal Effort.

PubMed

Bottalico, Pasquale

2017-05-01

The aims of the present study are (1) to analyze the effects of the acoustical environment and the voice style on time dose (D t_p ) and fundamental frequency (mean f 0 and standard deviation std_f 0 ) while taking into account the effect of short-term vocal fatigue and (2) to predict the self-reported vocal effort from the voice acoustical parameters. Ten male and ten female subjects were recorded while reading a text in normal and loud styles, in three rooms-anechoic, semi-reverberant, and reverberant-with and without acrylic glass panels 0.5 m from the mouth, which increased external auditory feedback. Subjects quantified how much effort was required to speak in each condition on a visual analogue scale after each task. (Aim1) In the loud style, D t_p , f 0 , and std_f 0 increased. The D t_p was higher in the reverberant room compared to the other two rooms. Both genders tended to increase f 0 in less reverberant environments, whereas a more monotonous speech was produced in rooms with greater reverberation. All three voice parameters increased with short-term vocal fatigue. (Aim2) A model of the vocal effort to acoustic vocal parameters is proposed. The sound pressure level contributed to 66% of the variance explained by the model, followed by the f 0 (30%) and the modulation in amplitude (4%). The results provide insight into how voice acoustical parameters can predict vocal effort. In particular, it increased when SPL and f 0 increased and when the amplitude voice modulation decreased. Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

The Relationship Between Speech Production and Speech Perception Deficits in Parkinson's Disease.

PubMed

De Keyser, Kim; Santens, Patrick; Bockstael, Annelies; Botteldooren, Dick; Talsma, Durk; De Vos, Stefanie; Van Cauwenberghe, Mieke; Verheugen, Femke; Corthals, Paul; De Letter, Miet

2016-10-01

This study investigated the possible relationship between hypokinetic speech production and speech intensity perception in patients with Parkinson's disease (PD). Participants included 14 patients with idiopathic PD and 14 matched healthy controls (HCs) with normal hearing and cognition. First, speech production was objectified through a standardized speech intelligibility assessment, acoustic analysis, and speech intensity measurements. Second, an overall estimation task and an intensity estimation task were addressed to evaluate overall speech perception and speech intensity perception, respectively. Finally, correlation analysis was performed between the speech characteristics of the overall estimation task and the corresponding acoustic analysis. The interaction between speech production and speech intensity perception was investigated by an intensity imitation task. Acoustic analysis and speech intensity measurements demonstrated significant differences in speech production between patients with PD and the HCs. A different pattern in the auditory perception of speech and speech intensity was found in the PD group. Auditory perceptual deficits may influence speech production in patients with PD. The present results suggest a disturbed auditory perception related to an automatic monitoring deficit in PD.

Visual Context Enhanced: The Joint Contribution of Iconic Gestures and Visible Speech to Degraded Speech Comprehension.

PubMed

Drijvers, Linda; Özyürek, Asli

2017-01-01

This study investigated whether and to what extent iconic co-speech gestures contribute to information from visible speech to enhance degraded speech comprehension at different levels of noise-vocoding. Previous studies of the contributions of these 2 visual articulators to speech comprehension have only been performed separately. Twenty participants watched videos of an actress uttering an action verb and completed a free-recall task. The videos were presented in 3 speech conditions (2-band noise-vocoding, 6-band noise-vocoding, clear), 3 multimodal conditions (speech + lips blurred, speech + visible speech, speech + visible speech + gesture), and 2 visual-only conditions (visible speech, visible speech + gesture). Accuracy levels were higher when both visual articulators were present compared with 1 or none. The enhancement effects of (a) visible speech, (b) gestural information on top of visible speech, and (c) both visible speech and iconic gestures were larger in 6-band than 2-band noise-vocoding or visual-only conditions. Gestural enhancement in 2-band noise-vocoding did not differ from gestural enhancement in visual-only conditions. When perceiving degraded speech in a visual context, listeners benefit more from having both visual articulators present compared with 1. This benefit was larger at 6-band than 2-band noise-vocoding, where listeners can benefit from both phonological cues from visible speech and semantic cues from iconic gestures to disambiguate speech.

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

PubMed

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

2017-09-01

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

Treatment for Vocal Polyps: Lips and Tongue Trill.

PubMed

de Vasconcelos, Daniela; Gomes, Adriana de Oliveira Camargo; de Araújo, Cláudia Marina Tavares

2017-03-01

Vocal polyps do not have a well-defined therapeutic indication. The recommended treatment is often laryngeal microsurgery, followed by postoperative speech therapy. Speech therapy as the initial treatment for polyps is a new concept and aims to modify inappropriate vocal behavior, adjust the voice quality, and encourage regression of the lesion. This study aimed to determine the effectiveness of the sonorous lips and tongue trill technique in the treatment of vocal polyps. The sample consisted of 10 adults diagnosed with a polyp who were divided into two subgroups: treatment and control. Ten speech therapy sessions were conducted, each lasting 30-45 minutes, based on the sonorous lips and tongue trill technique, accompanied by continuous guidance about vocal health. Speech therapy was effective in three of the five participants. The number of symptoms presented by the participants decreased significantly after voice therapy (P = 0.034) and vocal self-evaluation (P = 0.034). The acoustic evaluation showed improvements in parameters of noise values (P = 0.028) and jitter (P = 0.034). The size of the polyp and the degree of severity of dysphonia, hoarseness, and breathiness showed a significant reduction after treatment (P = 0.043). Among the remaining two participants, one opted out of laryngeal surgery, indicating that the improvement obtained was sufficient to avoid surgery. The sonorous lips and tongue trill technique was thus considered effective in 60% of the participants, and as laryngeal surgery was avoided in 80% of them, it should be considered a treatment option for vocal polyps. Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

Adapted cuing technique: facilitating sequential phoneme production.

PubMed

Klick, S L

1994-09-01

ACT is a visual cuing technique designed to facilitate dyspraxic speech by highlighting the sequential production of phonemes. In using ACT, cues are presented in such a way as to suggest sequential, coarticulatory movement in an overall pattern of motion. While using ACT, the facilitator's hand moves forward and back along the side of her (or his) own face. Finger movements signal specific speech sounds in formations loosely based on the manual alphabet for the hearing impaired. The best movements suggest the flowing, interactive nature of coarticulated phonemes. The synergistic nature of speech is suggested by coordinated hand motions which tighten and relax, move quickly or slowly, reflecting the motions of the vocal tract at various points during production of phonemic sequences. General principles involved in using ACT include a primary focus on speech-in-motion, the monitoring and fading of cues, and the presentation of stimuli based on motor-task analysis of phonemic sequences. Phonemic sequences are cued along three dimensions: place, manner, and vowel-related mandibular motion. Cuing vowels is a central feature of ACT. Two parameters of vowel production, focal point of resonance and mandibular closure, are cued. The facilitator's hand motions reflect the changing shape of the vocal tract and the trajectory of the tongue that result from the coarticulation of vowels and consonants. Rigid presentation of the phonemes is secondary to the facilitator's primary focus on presenting the overall sequential movement. The facilitator's goal is to self-tailor ACT in response to the changing needs and abilities of the client.(ABSTRACT TRUNCATED AT 250 WORDS)

Excitability of the motor system: A transcranial magnetic stimulation study on singing and speaking.

PubMed

Royal, Isabelle; Lidji, Pascale; Théoret, Hugo; Russo, Frank A; Peretz, Isabelle

2015-08-01

The perception of movements is associated with increased activity in the human motor cortex, which in turn may underlie our ability to understand actions, as it may be implicated in the recognition, understanding and imitation of actions. Here, we investigated the involvement and lateralization of the primary motor cortex (M1) in the perception of singing and speech. Transcranial magnetic stimulation (TMS) was applied independently for both hemispheres over the mouth representation of the motor cortex in healthy participants while they watched 4-s audiovisual excerpts of singers producing a 2-note ascending interval (singing condition) or 4-s audiovisual excerpts of a person explaining a proverb (speech condition). Subjects were instructed to determine whether a sung interval/written proverb, matched a written interval/proverb. During both tasks, motor evoked potentials (MEPs) were recorded from the contralateral mouth muscle (orbicularis oris) of the stimulated motor cortex compared to a control task. Moreover, to investigate the time course of motor activation, TMS pulses were randomly delivered at 7 different time points (ranging from 500 to 3500 ms after stimulus onset). Results show that stimulation of the right hemisphere had a similar effect on the MEPs for both the singing and speech perception tasks, whereas stimulation of the left hemisphere significantly differed in the speech perception task compared to the singing perception task. Furthermore, analysis of the MEPs in the singing task revealed that they decreased for small musical intervals, but increased for large musical intervals, regardless of which hemisphere was stimulated. Overall, these results suggest a dissociation between the lateralization of M1 activity for speech perception and for singing perception, and that in the latter case its activity can be modulated by musical parameters such as the size of a musical interval. Copyright © 2015 Elsevier Ltd. All rights reserved.

Non-fluent speech following stroke is caused by impaired efference copy.

PubMed

Feenaughty, Lynda; Basilakos, Alexandra; Bonilha, Leonardo; den Ouden, Dirk-Bart; Rorden, Chris; Stark, Brielle; Fridriksson, Julius

2017-09-01

Efference copy is a cognitive mechanism argued to be critical for initiating and monitoring speech: however, the extent to which breakdown of efference copy mechanisms impact speech production is unclear. This study examined the best mechanistic predictors of non-fluent speech among 88 stroke survivors. Objective speech fluency measures were subjected to a principal component analysis (PCA). The primary PCA factor was then entered into a multiple stepwise linear regression analysis as the dependent variable, with a set of independent mechanistic variables. Participants' ability to mimic audio-visual speech ("speech entrainment response") was the best independent predictor of non-fluent speech. We suggest that this "speech entrainment" factor reflects integrity of internal monitoring (i.e., efference copy) of speech production, which affects speech initiation and maintenance. Results support models of normal speech production and suggest that therapy focused on speech initiation and maintenance may improve speech fluency for individuals with chronic non-fluent aphasia post stroke.

Autistic traits and attention to speech: Evidence from typically developing individuals.

PubMed

Korhonen, Vesa; Werner, Stefan

2017-04-01

Individuals with autism spectrum disorder have a preference for attending to non-speech stimuli over speech stimuli. We are interested in whether non-speech preference is only a feature of diagnosed individuals, and whether we can we test implicit preference experimentally. In typically developed individuals, serial recall is disrupted more by speech stimuli than by non-speech stimuli. Since behaviour of individuals with autistic traits resembles that of individuals with autism, we have used serial recall to test whether autistic traits influence task performance during irrelevant speech sounds. The errors made on the serial recall task during speech or non-speech sounds were counted as a measure of speech or non-speech preference in relation to no sound condition. We replicated the serial order effect and found the speech to be more disruptive than the non-speech sounds, but were unable to find any associations between the autism quotient scores and the non-speech sounds. Our results may indicate a learnt behavioural response to speech sounds.

Electrophysiological evidence for speech-specific audiovisual integration.

PubMed

Baart, Martijn; Stekelenburg, Jeroen J; Vroomen, Jean

2014-01-01

Lip-read speech is integrated with heard speech at various neural levels. Here, we investigated the extent to which lip-read induced modulations of the auditory N1 and P2 (measured with EEG) are indicative of speech-specific audiovisual integration, and we explored to what extent the ERPs were modulated by phonetic audiovisual congruency. In order to disentangle speech-specific (phonetic) integration from non-speech integration, we used Sine-Wave Speech (SWS) that was perceived as speech by half of the participants (they were in speech-mode), while the other half was in non-speech mode. Results showed that the N1 obtained with audiovisual stimuli peaked earlier than the N1 evoked by auditory-only stimuli. This lip-read induced speeding up of the N1 occurred for listeners in speech and non-speech mode. In contrast, if listeners were in speech-mode, lip-read speech also modulated the auditory P2, but not if listeners were in non-speech mode, thus revealing speech-specific audiovisual binding. Comparing ERPs for phonetically congruent audiovisual stimuli with ERPs for incongruent stimuli revealed an effect of phonetic stimulus congruency that started at ~200 ms after (in)congruence became apparent. Critically, akin to the P2 suppression, congruency effects were only observed if listeners were in speech mode, and not if they were in non-speech mode. Using identical stimuli, we thus confirm that audiovisual binding involves (partially) different neural mechanisms for sound processing in speech and non-speech mode. © 2013 Published by Elsevier Ltd.

Inner Speech's Relationship With Overt Speech in Poststroke Aphasia.

PubMed

Stark, Brielle C; Geva, Sharon; Warburton, Elizabeth A

2017-09-18

Relatively preserved inner speech alongside poor overt speech has been documented in some persons with aphasia (PWA), but the relationship of overt speech with inner speech is still largely unclear, as few studies have directly investigated these factors. The present study investigates the relationship of relatively preserved inner speech in aphasia with selected measures of language and cognition. Thirty-eight persons with chronic aphasia (27 men, 11 women; average age 64.53 ± 13.29 years, time since stroke 8-111 months) were classified as having relatively preserved inner and overt speech (n = 21), relatively preserved inner speech with poor overt speech (n = 8), or not classified due to insufficient measurements of inner and/or overt speech (n = 9). Inner speech scores (by group) were correlated with selected measures of language and cognition from the Comprehensive Aphasia Test (Swinburn, Porter, & Al, 2004). The group with poor overt speech showed a significant relationship of inner speech with overt naming (r = .95, p < .01) and with mean length of utterance produced during a written picture description (r = .96, p < .01). Correlations between inner speech and language and cognition factors were not significant for the group with relatively good overt speech. As in previous research, we show that relatively preserved inner speech is found alongside otherwise severe production deficits in PWA. PWA with poor overt speech may rely more on preserved inner speech for overt picture naming (perhaps due to shared resources with verbal working memory) and for written picture description (perhaps due to reliance on inner speech due to perceived task difficulty). Assessments of inner speech may be useful as a standard component of aphasia screening, and therapy focused on improving and using inner speech may prove clinically worthwhile. https://doi.org/10.23641/asha.5303542.

Atypical speech versus non-speech detection and discrimination in 4- to 6- yr old children with autism spectrum disorder: An ERP study.

PubMed

Galilee, Alena; Stefanidou, Chrysi; McCleery, Joseph P

2017-01-01

Previous event-related potential (ERP) research utilizing oddball stimulus paradigms suggests diminished processing of speech versus non-speech sounds in children with an Autism Spectrum Disorder (ASD). However, brain mechanisms underlying these speech processing abnormalities, and to what extent they are related to poor language abilities in this population remain unknown. In the current study, we utilized a novel paired repetition paradigm in order to investigate ERP responses associated with the detection and discrimination of speech and non-speech sounds in 4- to 6-year old children with ASD, compared with gender and verbal age matched controls. ERPs were recorded while children passively listened to pairs of stimuli that were either both speech sounds, both non-speech sounds, speech followed by non-speech, or non-speech followed by speech. Control participants exhibited N330 match/mismatch responses measured from temporal electrodes, reflecting speech versus non-speech detection, bilaterally, whereas children with ASD exhibited this effect only over temporal electrodes in the left hemisphere. Furthermore, while the control groups exhibited match/mismatch effects at approximately 600 ms (central N600, temporal P600) when a non-speech sound was followed by a speech sound, these effects were absent in the ASD group. These findings suggest that children with ASD fail to activate right hemisphere mechanisms, likely associated with social or emotional aspects of speech detection, when distinguishing non-speech from speech stimuli. Together, these results demonstrate the presence of atypical speech versus non-speech processing in children with ASD when compared with typically developing children matched on verbal age.

Atypical speech versus non-speech detection and discrimination in 4- to 6- yr old children with autism spectrum disorder: An ERP study

PubMed Central

Stefanidou, Chrysi; McCleery, Joseph P.

2017-01-01

Previous event-related potential (ERP) research utilizing oddball stimulus paradigms suggests diminished processing of speech versus non-speech sounds in children with an Autism Spectrum Disorder (ASD). However, brain mechanisms underlying these speech processing abnormalities, and to what extent they are related to poor language abilities in this population remain unknown. In the current study, we utilized a novel paired repetition paradigm in order to investigate ERP responses associated with the detection and discrimination of speech and non-speech sounds in 4- to 6—year old children with ASD, compared with gender and verbal age matched controls. ERPs were recorded while children passively listened to pairs of stimuli that were either both speech sounds, both non-speech sounds, speech followed by non-speech, or non-speech followed by speech. Control participants exhibited N330 match/mismatch responses measured from temporal electrodes, reflecting speech versus non-speech detection, bilaterally, whereas children with ASD exhibited this effect only over temporal electrodes in the left hemisphere. Furthermore, while the control groups exhibited match/mismatch effects at approximately 600 ms (central N600, temporal P600) when a non-speech sound was followed by a speech sound, these effects were absent in the ASD group. These findings suggest that children with ASD fail to activate right hemisphere mechanisms, likely associated with social or emotional aspects of speech detection, when distinguishing non-speech from speech stimuli. Together, these results demonstrate the presence of atypical speech versus non-speech processing in children with ASD when compared with typically developing children matched on verbal age. PMID:28738063

Emotion Analysis of Telephone Complaints from Customer Based on Affective Computing.

PubMed

Gong, Shuangping; Dai, Yonghui; Ji, Jun; Wang, Jinzhao; Sun, Hai

2015-01-01

Customer complaint has been the important feedback for modern enterprises to improve their product and service quality as well as the customer's loyalty. As one of the commonly used manners in customer complaint, telephone communication carries rich emotional information of speeches, which provides valuable resources for perceiving the customer's satisfaction and studying the complaint handling skills. This paper studies the characteristics of telephone complaint speeches and proposes an analysis method based on affective computing technology, which can recognize the dynamic changes of customer emotions from the conversations between the service staff and the customer. The recognition process includes speaker recognition, emotional feature parameter extraction, and dynamic emotion recognition. Experimental results show that this method is effective and can reach high recognition rates of happy and angry states. It has been successfully applied to the operation quality and service administration in telecom and Internet service company.

The impact of compression of speech signal, background noise and acoustic disturbances on the effectiveness of speaker identification

NASA Astrophysics Data System (ADS)

Kamiński, K.; Dobrowolski, A. P.

2017-04-01

The paper presents the architecture and the results of optimization of selected elements of the Automatic Speaker Recognition (ASR) system that uses Gaussian Mixture Models (GMM) in the classification process. Optimization was performed on the process of selection of individual characteristics using the genetic algorithm and the parameters of Gaussian distributions used to describe individual voices. The system that was developed was tested in order to evaluate the impact of different compression methods used, among others, in landline, mobile, and VoIP telephony systems, on effectiveness of the speaker identification. Also, the results were presented of effectiveness of speaker identification at specific levels of noise with the speech signal and occurrence of other disturbances that could appear during phone calls, which made it possible to specify the spectrum of applications of the presented ASR system.

Emotion Analysis of Telephone Complaints from Customer Based on Affective Computing

PubMed Central

Gong, Shuangping; Ji, Jun; Wang, Jinzhao; Sun, Hai

2015-01-01

Customer complaint has been the important feedback for modern enterprises to improve their product and service quality as well as the customer's loyalty. As one of the commonly used manners in customer complaint, telephone communication carries rich emotional information of speeches, which provides valuable resources for perceiving the customer's satisfaction and studying the complaint handling skills. This paper studies the characteristics of telephone complaint speeches and proposes an analysis method based on affective computing technology, which can recognize the dynamic changes of customer emotions from the conversations between the service staff and the customer. The recognition process includes speaker recognition, emotional feature parameter extraction, and dynamic emotion recognition. Experimental results show that this method is effective and can reach high recognition rates of happy and angry states. It has been successfully applied to the operation quality and service administration in telecom and Internet service company. PMID:26633967

Design of an MSAT-X mobile transceiver and related base and gateway stations

NASA Technical Reports Server (NTRS)

Fang, Russell J. F.; Bhaskar, Udaya; Hemmati, Farhad; Mackenthun, Kenneth M.; Shenoy, Ajit

1987-01-01

This paper summarizes the results of a design study of the mobile transceiver, base station, and gateway station for NASA's proposed Mobile Satellite Experiment (MSAT-X). Major ground segment system design issues such as frequency stability control, modulation method, linear predictive coding vocoder algorithm, and error control technique are addressed. The modular and flexible transceiver design is described in detail, including the core, RF/IF, modem, vocoder, forward error correction codec, amplitude-companded single sideband, and input/output modules, as well as the flexible interface. Designs for a three-carrier base station and a 10-carrier gateway station are also discussed, including the interface with the controllers and with the public-switched telephone networks at the gateway station. Functional specifications are given for the transceiver, the base station, and the gateway station.

NASA. Lewis Research Center Advanced Modulation and Coding Project: Introduction and overview

NASA Technical Reports Server (NTRS)

Budinger, James M.

1992-01-01

The Advanced Modulation and Coding Project at LeRC is sponsored by the Office of Space Science and Applications, Communications Division, Code EC, at NASA Headquarters and conducted by the Digital Systems Technology Branch of the Space Electronics Division. Advanced Modulation and Coding is one of three focused technology development projects within the branch's overall Processing and Switching Program. The program consists of industry contracts for developing proof-of-concept (POC) and demonstration model hardware, university grants for analyzing advanced techniques, and in-house integration and testing of performance verification and systems evaluation. The Advanced Modulation and Coding Project is broken into five elements: (1) bandwidth- and power-efficient modems; (2) high-speed codecs; (3) digital modems; (4) multichannel demodulators; and (5) very high-data-rate modems. At least one contract and one grant were awarded for each element.

Design of an H.264/SVC resilient watermarking scheme

NASA Astrophysics Data System (ADS)

Van Caenegem, Robrecht; Dooms, Ann; Barbarien, Joeri; Schelkens, Peter

2010-01-01

The rapid dissemination of media technologies has lead to an increase of unauthorized copying and distribution of digital media. Digital watermarking, i.e. embedding information in the multimedia signal in a robust and imperceptible manner, can tackle this problem. Recently, there has been a huge growth in the number of different terminals and connections that can be used to consume multimedia. To tackle the resulting distribution challenges, scalable coding is often employed. Scalable coding allows the adaptation of a single bit-stream to varying terminal and transmission characteristics. As a result of this evolution, watermarking techniques that are robust against scalable compression become essential in order to control illegal copying. In this paper, a watermarking technique resilient against scalable video compression using the state-of-the-art H.264/SVC codec is therefore proposed and evaluated.

Caution and Warning Alarm Design and Evaluation for NASA CEV Auditory Displays: SHFE Information Presentation Directed Research Project (DRPP) report 12.07

NASA Technical Reports Server (NTRS)

Begault, Durand R.; Godfroy, Martine; Sandor, Aniko; Holden, Kritina

2008-01-01

The design of caution-warning signals for NASA s Crew Exploration Vehicle (CEV) and other future spacecraft will be based on both best practices based on current research and evaluation of current alarms. A design approach is presented based upon cross-disciplinary examination of psychoacoustic research, human factors experience, aerospace practices, and acoustical engineering requirements. A listening test with thirteen participants was performed involving ranking and grading of current and newly developed caution-warning stimuli under three conditions: (1) alarm levels adjusted for compliance with ISO 7731, "Danger signals for work places - Auditory Danger Signals", (2) alarm levels adjusted to an overall 15 dBA s/n ratio and (3) simulated codec low-pass filtering. Questionnaire data yielded useful insights regarding cognitive associations with the sounds.

Design and analysis of multihypothesis motion-compensated prediction (MHMCP) codec for error-resilient visual communications

NASA Astrophysics Data System (ADS)

Kung, Wei-Ying; Kim, Chang-Su; Kuo, C.-C. Jay

2004-10-01

A multi-hypothesis motion compensated prediction (MHMCP) scheme, which predicts a block from a weighted superposition of more than one reference blocks in the frame buffer, is proposed and analyzed for error resilient visual communication in this research. By combining these reference blocks effectively, MHMCP can enhance the error resilient capability of compressed video as well as achieve a coding gain. In particular, we investigate the error propagation effect in the MHMCP coder and analyze the rate-distortion performance in terms of the hypothesis number and hypothesis coefficients. It is shown that MHMCP suppresses the short-term effect of error propagation more effectively than the intra refreshing scheme. Simulation results are given to confirm the analysis. Finally, several design principles for the MHMCP coder are derived based on the analytical and experimental results.

Optical network security using unipolar Walsh code

NASA Astrophysics Data System (ADS)

Sikder, Somali; Sarkar, Madhumita; Ghosh, Shila

2018-04-01

Optical code-division multiple-access (OCDMA) is considered as a good technique to provide optical layer security. Many research works have been published to enhance optical network security by using optical signal processing. The paper, demonstrates the design of the AWG (arrayed waveguide grating) router-based optical network for spectral-amplitude-coding (SAC) OCDMA networks with Walsh Code to design a reconfigurable network codec by changing signature codes to against eavesdropping. In this paper we proposed a code reconfiguration scheme to improve the network access confidentiality changing the signature codes by cyclic rotations, for OCDMA system. Each of the OCDMA network users is assigned a unique signature code to transmit the information and at the receiving end each receiver correlates its own signature pattern a(n) with the receiving pattern s(n). The signal arriving at proper destination leads to s(n)=a(n).

Design of an MSAT-X mobile transceiver and related base and gateway stations

NASA Astrophysics Data System (ADS)

Fang, Russell J. F.; Bhaskar, Udaya; Hemmati, Farhad; Mackenthun, Kenneth M.; Shenoy, Ajit

This paper summarizes the results of a design study of the mobile transceiver, base station, and gateway station for NASA's proposed Mobile Satellite Experiment (MSAT-X). Major ground segment system design issues such as frequency stability control, modulation method, linear predictive coding vocoder algorithm, and error control technique are addressed. The modular and flexible transceiver design is described in detail, including the core, RF/IF, modem, vocoder, forward error correction codec, amplitude-companded single sideband, and input/output modules, as well as the flexible interface. Designs for a three-carrier base station and a 10-carrier gateway station are also discussed, including the interface with the controllers and with the public-switched telephone networks at the gateway station. Functional specifications are given for the transceiver, the base station, and the gateway station.

Speech Clarity Index (Ψ): A Distance-Based Speech Quality Indicator and Recognition Rate Prediction for Dysarthric Speakers with Cerebral Palsy

NASA Astrophysics Data System (ADS)

Kayasith, Prakasith; Theeramunkong, Thanaruk

It is a tedious and subjective task to measure severity of a dysarthria by manually evaluating his/her speech using available standard assessment methods based on human perception. This paper presents an automated approach to assess speech quality of a dysarthric speaker with cerebral palsy. With the consideration of two complementary factors, speech consistency and speech distinction, a speech quality indicator called speech clarity index (Ψ) is proposed as a measure of the speaker's ability to produce consistent speech signal for a certain word and distinguished speech signal for different words. As an application, it can be used to assess speech quality and forecast speech recognition rate of speech made by an individual dysarthric speaker before actual exhaustive implementation of an automatic speech recognition system for the speaker. The effectiveness of Ψ as a speech recognition rate predictor is evaluated by rank-order inconsistency, correlation coefficient, and root-mean-square of difference. The evaluations had been done by comparing its predicted recognition rates with ones predicted by the standard methods called the articulatory and intelligibility tests based on the two recognition systems (HMM and ANN). The results show that Ψ is a promising indicator for predicting recognition rate of dysarthric speech. All experiments had been done on speech corpus composed of speech data from eight normal speakers and eight dysarthric speakers.

Motor Speech Phenotypes of Frontotemporal Dementia, Primary Progressive Aphasia, and Progressive Apraxia of Speech.

PubMed

Poole, Matthew L; Brodtmann, Amy; Darby, David; Vogel, Adam P

2017-04-14

Our purpose was to create a comprehensive review of speech impairment in frontotemporal dementia (FTD), primary progressive aphasia (PPA), and progressive apraxia of speech in order to identify the most effective measures for diagnosis and monitoring, and to elucidate associations between speech and neuroimaging. Speech and neuroimaging data described in studies of FTD and PPA were systematically reviewed. A meta-analysis was conducted for speech measures that were used consistently in multiple studies. The methods and nomenclature used to describe speech in these disorders varied between studies. Our meta-analysis identified 3 speech measures which differentiate variants or healthy control-group participants (e.g., nonfluent and logopenic variants of PPA from all other groups, behavioral-variant FTD from a control group). Deficits within the frontal-lobe speech networks are linked to motor speech profiles of the nonfluent variant of PPA and progressive apraxia of speech. Motor speech impairment is rarely reported in semantic and logopenic variants of PPA. Limited data are available on motor speech impairment in the behavioral variant of FTD. Our review identified several measures of speech which may assist with diagnosis and classification, and consolidated the brain-behavior associations relating to speech in FTD, PPA, and progressive apraxia of speech.

System and method for characterizing voiced excitations of speech and acoustic signals, removing acoustic noise from speech, and synthesizing speech

DOEpatents

Burnett, Greg C.; Holzrichter, John F.; Ng, Lawrence C.

2002-01-01

Low power EM waves are used to detect motions of vocal tract tissues of the human speech system before, during, and after voiced speech. A voiced excitation function is derived. The excitation function provides speech production information to enhance speech characterization and to enable noise removal from human speech.

Live maternal speech and singing have beneficial effects on hospitalized preterm infants.

PubMed

Filippa, Manuela; Devouche, Emmanuel; Arioni, Cesare; Imberty, Michel; Gratier, Maya

2013-10-01

To study the effects of live maternal speaking and singing on physiological parameters of preterm infants in the NICU and to test the hypothesis that vocal stimulation can have differential effects on preterm infants at a behavioural level. Eighteen mothers spoke and sang to their medically stable preterm infants in their incubators over 6 days, between 1 and 2 pm. Heart rate (HR), oxygen saturation (OxSat), number of critical events (hypoxemia, bradycardia and apnoea) and change in behavioural state were measured. Comparisons of periods with and without maternal vocal stimulation revealed significantly greater oxygen saturation level and heart rate and significantly fewer negative critical events (p < 0.0001) when the mother was speaking and singing. Unexpected findings were the comparable effects of maternal talk and singing on infant physiological parameters and the differential ones on infant behavioural state. A renewed connection to the mother's voice can be an important and significant experience for preterm infants. Exposure to maternal speech and singing shows significant early beneficial effects on physiological state, such as oxygen saturation levels, number of critical events and prevalence of calm alert state. These findings have implications for NICU interventions, encouraging maternal interaction with their medically stable preterm infants. ©2013 Foundation Acta Paediatrica. Published by John Wiley & Sons Ltd.

Modeling phoneme perception. II: A model of stop consonant discrimination.

PubMed

van Hessen, A J; Schouten, M E

1992-10-01

Combining elements from two existing theories of speech sound discrimination, dual process theory (DPT) and trace context theory (TCT), a new theory, called phoneme perception theory, is proposed, consisting of a long-term phoneme memory, a context-coding memory, and a trace memory, each with its own time constants. This theory is tested by means of stop-consonant discrimination data in which interstimulus interval (ISI; values of 100, 300, and 2000 ms) is an important variable. It is shown that discrimination in which labeling plays an important part (2IFC and AX between category) benefits from increased ISI, whereas discrimination in which only sensory traces are compared (AX within category), decreases with increasing ISI. The theory is also tested on speech discrimination data from the literature in which ISI is a variable [Pisoni, J. Acoust. Soc. Am. 36, 277-282 (1964); Cowan and Morse, J. Acoust. Soc. Am. 79, 500-507 (1986)]. It is concluded that the number of parameters in trace context theory is not sufficient to account for most speech-sound discrimination data and that a few additional assumptions are needed, such as a form of sublabeling, in which subjects encode the quality of a stimulus as a member of a category, and which requires processing time.

Recovery from aphasia after hemicraniectomy for infarction of the speech-dominant hemisphere.

PubMed

Kastrau, Frank; Wolter, Marcus; Huber, Walter; Block, Frank

2005-04-01

The space-occupying effect of cerebral edema limits survival chances of patients with severe ischemic stroke. Besides conventional therapies to reduce intracranial pressure, hemicraniectomy can be considered as a therapeutic option after space-occupying cerebral infarction. There is controversy regarding the use of this method in patients with infarction of the speech-dominant hemisphere. In 14 patients with infarction of the dominant hemisphere and subsequent treatment with hemicraniectomy, recovery from aphasic symptoms was evaluated retrospectively. A group of patients who were treated between 1994 and 2003 in our aphasia ward was selected for the study. In all patients, a psychometric quantification was accomplished applying the Aachen Aphasia Test at least twice within a mean observation period of 470 days. A significant improvement of the statistical parameters representing different aspects of aphasia was observed in 13 of 14 patients. Also, an increase of the ability to communicate was evident in 13 patients. Young age at the time of stroke and early poststroke decompressive surgery were identified as main predictors for recovery from aphasia. A significant improvement of aphasic symptoms can be observed in a preselected group of patients after a massive stroke of the speech-dominant hemisphere treated by consecutive hemicraniectomy. Therefore, decompressive surgery can be considered for the treatment of this kind of stroke.

Kernel-Based Sensor Fusion With Application to Audio-Visual Voice Activity Detection

NASA Astrophysics Data System (ADS)

Dov, David; Talmon, Ronen; Cohen, Israel

2016-12-01

In this paper, we address the problem of multiple view data fusion in the presence of noise and interferences. Recent studies have approached this problem using kernel methods, by relying particularly on a product of kernels constructed separately for each view. From a graph theory point of view, we analyze this fusion approach in a discrete setting. More specifically, based on a statistical model for the connectivity between data points, we propose an algorithm for the selection of the kernel bandwidth, a parameter, which, as we show, has important implications on the robustness of this fusion approach to interferences. Then, we consider the fusion of audio-visual speech signals measured by a single microphone and by a video camera pointed to the face of the speaker. Specifically, we address the task of voice activity detection, i.e., the detection of speech and non-speech segments, in the presence of structured interferences such as keyboard taps and office noise. We propose an algorithm for voice activity detection based on the audio-visual signal. Simulation results show that the proposed algorithm outperforms competing fusion and voice activity detection approaches. In addition, we demonstrate that a proper selection of the kernel bandwidth indeed leads to improved performance.

Audiovisual perceptual learning with multiple speakers.

PubMed

Mitchel, Aaron D; Gerfen, Chip; Weiss, Daniel J

2016-05-01

One challenge for speech perception is between-speaker variability in the acoustic parameters of speech. For example, the same phoneme (e.g. the vowel in "cat") may have substantially different acoustic properties when produced by two different speakers and yet the listener must be able to interpret these disparate stimuli as equivalent. Perceptual tuning, the use of contextual information to adjust phonemic representations, may be one mechanism that helps listeners overcome obstacles they face due to this variability during speech perception. Here we test whether visual contextual cues to speaker identity may facilitate the formation and maintenance of distributional representations for individual speakers, allowing listeners to adjust phoneme boundaries in a speaker-specific manner. We familiarized participants to an audiovisual continuum between /aba/ and /ada/. During familiarization, the "b-face" mouthed /aba/ when an ambiguous token was played, while the "D-face" mouthed /ada/. At test, the same ambiguous token was more likely to be identified as /aba/ when paired with a stilled image of the "b-face" than with an image of the "D-face." This was not the case in the control condition when the two faces were paired equally with the ambiguous token. Together, these results suggest that listeners may form speaker-specific phonemic representations using facial identity cues.

Vocal Tract Images Reveal Neural Representations of Sensorimotor Transformation During Speech Imitation

PubMed Central

Carey, Daniel; Miquel, Marc E.; Evans, Bronwen G.; Adank, Patti; McGettigan, Carolyn

2017-01-01

Abstract Imitating speech necessitates the transformation from sensory targets to vocal tract motor output, yet little is known about the representational basis of this process in the human brain. Here, we address this question by using real-time MR imaging (rtMRI) of the vocal tract and functional MRI (fMRI) of the brain in a speech imitation paradigm. Participants trained on imitating a native vowel and a similar nonnative vowel that required lip rounding. Later, participants imitated these vowels and an untrained vowel pair during separate fMRI and rtMRI runs. Univariate fMRI analyses revealed that regions including left inferior frontal gyrus were more active during sensorimotor transformation (ST) and production of nonnative vowels, compared with native vowels; further, ST for nonnative vowels activated somatomotor cortex bilaterally, compared with ST of native vowels. Using test representational similarity analysis (RSA) models constructed from participants’ vocal tract images and from stimulus formant distances, we found that RSA searchlight analyses of fMRI data showed either type of model could be represented in somatomotor, temporal, cerebellar, and hippocampal neural activation patterns during ST. We thus provide the first evidence of widespread and robust cortical and subcortical neural representation of vocal tract and/or formant parameters, during prearticulatory ST. PMID:28334401

EMA analysis of tongue function in children with dysarthria following traumatic brain injury.

PubMed

Murdoch, Bruce E; Goozée, Justine V

2003-01-01

To investigate the speed and accuracy of tongue movements exhibited by a sample of children with dysarthria following severe traumatic brain injury (TBI) during speech using electromagnetic articulography (EMA). Four children, aged between 12.75-17.17 years with dysarthria following TBI, were assessed using the AG-100 electromagnetic articulography system (Carstens Medizinelektronik). The movement trajectories of receiver coils affixed to each child's tongue were examined during consonant productions, together with a range of quantitative kinematic parameters. The children's results were individually compared against the mean values obtained by a group of eight control children (mean age of 14.67 years, SD 1.60). All four TBI children were perceived to exhibit reduced rates of speech and increased word durations. Objective EMA analysis revealed that two of the TBI children exhibited significantly longer consonant durations compared to the control group, resulting from different underlying mechanisms relating to speed generation capabilities and distances travelled. The other two TBI children did not exhibit increased initial consonant movement durations, suggesting that the vowels and/or final consonants may have been contributing to the increased word durations. The finding of different underlying articulatory kinematic profiles has important implications for the treatment of speech rate disturbances in children with dysarthria following TBI.