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检索条件"主题词=Linear predictive coding"

共 2458 条记录，以下是491-500 订阅

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ESTIMATION OF FRAME INDEPENDENT AND ENHANCEMENT COMPONENTS FOR SPEECH COMMUNICATION OVER PACKET NETWORKS

ESTIMATION OF FRAME INDEPENDENT AND ENHANCEMENT COMPONENTS F...

引用

2010 IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Giacobello, Daniele Murthi, Manohar N. Christensen, Mads Graesboll Jensen, Soren Holdt Moonen, Marc Aalborg Univ Dept Elect Syst Aalborg Denmark Miami Univ Dept Elect & Comp Engn Oxford OH 45056 USA Katholieke Univ Leuven Dept Elect Engn Leuven Belgium

ISBN: (纸本)9781424442966

In this paper, we describe a new approach to cope with packet loss in speech coders. The idea is to split the information present in each speech packet into two components, one to independently decode the given speech frame and one to enhance it by exploiting interframe dependencies. The scheme is based on sparse linear prediction and a redefinition of the analysis-by-synthesis process. We present Mean Opinion Scores for the presented coder with different degrees of packet loss and show that it performs similarly to frame dependent coders for low packet loss probability and similarly to frame independent coders for high packet loss probability. We also present ideas on how to make the coder work synergistically with the channel loss estimate.

关键词： Speech coding Voice over IP (VoIP) linear predictive coding analysis-by-synthesis

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Voice-dial by statistical recognition of continuous speech

Voice-dial by statistical recognition of continuous speech

引用

International Symposium on Signals, Circuits and Systems

作者： Dumitru, CO Gavat, I Univ Politehn Bucharest Fac Elect & Telecommun Bucharest Romania

ISBN: (纸本)0780379799

In this paper is presented our work concerning continuous speech recognition in a telephone numbers voice-dialing task realized by statistical modeling. The speech is parameterized using the computational inexpensive linear predictive coding (LPC) to determine the LPC, the cepstral LPC and the reflection coefficients. In our tests, the recognizer based on Hidden Markov Models (HMMs) performs better for the cepstral LPC coefficients then for the reflection coefficients or the LPC coefficients.

关键词： speech recognition linear predictive coding cepstral analysis hidden Markov models statistical recognition continuous speech recognition telephone numbers voice-dialing task statistical modeling cepstral LPC reflection

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Real Time Implementation of MELP Speech Compression Algorithm using Blackfin Processors 9

Real Time Implementation of MELP Speech Compression Algorith...

引用

9th International Symposium on Image and Signal Processing and Analysis (ISPA)

作者： Duta, Cristina-Loredana Gheorghe, Laura Tapus, Nicolae Univ Politehn Bucuresti Dept Comp Sci & Engn Bucharest Romania

ISBN: (纸本)9781467380324

A large part of the latest research in speech coding algorithms is motivated by the need of obtaining secure military communications, to allow effective operation in a hostile environment. Since the bandwidth of the communication channel is a sensitive problem in military applications, low bit-rate speech compression methods are mostly used. Several speech processing applications such as Mixed Excitation linear Prediction are characterized by very strict requirements in power consumption, size, and voltage supply. These requirements are difficult to fulfill, given the complexity and number of functions to be implemented, together with the real time requirement and large dynamic range of the input signals. To meet these constraints, careful optimization should be done at all levels, ranging from algorithmic level, through system and circuit architecture, to layout and design of the cell library. The key points of this optimization are among others, the choice of the algorithms, the modification of the algorithms to reduce computational complexity, the choice of a fixed-point arithmetic unit, the minimization of the number of bits required at every node of the algorithm, and a careful match between algorithms and architecture. This paper concentrates on low bit rate speech coding technology, mainly in MELP and solved the problem of optimizing the program of MELP on Digital Signal Processor platform. The algorithm was ported onto a fixed point DSP, Blackfin 537, and stage by stage optimization was performed to meet the real time requirements. The main functions involved were analysis, parameter encoding, parameter decoding and synthesis. The fixed point source code at the MELP front end was also thoroughly optimized at the C Level. Memory optimization techniques such as data placement and caching were also used to reduce the processing time. The results we obtained show that real-time implementations of a speech vocoder based on the MELP standard for low bit rate commu

关键词： speech signal speech processing linear predictive coding digital signal processor Blackfin processor

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Qualitative Spectral Parameter coding for Speech Signals 7

Qualitative Spectral Parameter Coding for Speech Signals

引用

7th International Conference on Computational Intelligence, Communication Systems and Networks

作者： Charu Sharma, Sukriti Manav Rachna Coll Engn Dept Elect & Commun Faridabad India

ISBN: (纸本)9781467370165

Speech signal is the unique and special signal in communication system so it must be analyzed in order to extract its important parameters and to compress it for maximum utilization of available bandwidth. For this, there are various kinds of speech analysis and synthesis techniques that have been effectively used. Among all these techniques, linear predictive coding (LPC) is the most powerful one to represent the speech signal at reduced bit rates while preserving the quality of the signal and also provides accurate estimation of speech parameters and is computationally effective. Voice-excited LPC is the technique proposed in this paper. This technique has been implemented using both male and female voices and trade-offs between bit rates, delay, power signal to noise ratio and complexity are analyzed. It results in a low bit rate and a better signal to noise ratio.

关键词： Autocorrelation Bit-Rates Discrete cosine transform linear predictive coding Mean Square Error Power Signal to Noise Ratio

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Back Propagation Feed forward neural network approach for Speech Recognition 3

Back Propagation Feed forward neural network approach for Sp...

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International Conference on Reliability, Infocom Technologies and Optimization (ICRITO) (Trends and Future Directions)

作者： Rajput, Neelima Verma, S. K. GBPEC Dept CSE Pauri Gharwal Uttrakhand India

ISBN: (纸本)9781479968961

In this paper a biologically motivated approach for the English alphabet speech recognition is implemented by using a self-organized neural network. The designing of an accurate and effective speech recognition system is a challenging task in the area of human computer interface. linear predictive coding (LPC) is used for learn Feature extraction of input audio signals. Back propagation (BP) is a feed forward neural network and it propagates the error in backward direction to update the weights of hidden layers. The error is difference of actual output and target output computed on the basis of gradient descent method. The performance of the system is evaluated on the basis of recognition rate. We have used BP neural network architecture to recognize the time varying input data. The proposed provides better accurate results than the existing systems for the English Alphabet speech recognition.

关键词： Artificial Neural Networks Back Propagation Feed Forward Neural Network Gradient Descent Method linear predictive coding Speech Recognition System

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An Acoustic Signal Based Language Independent Lip Synchronization Method and Its Implementation via Extended LPC 28

An Acoustic Signal Based Language Independent Lip Synchroniz...

引用

28th Signal Processing and Communications Applications Conference (SIU)

作者： Cankurtaran, Halil Said Boyaci, Ali Yarkan, Serhan Istanbul Ticaret Univ Elekt Elekt Muhendisligi Bolumu Kucukyali E5 KavsagiInonu Cad4 Kucukyali TR-34840 Istanbul Turkey Istanbul Ticaret Univ Bilgisayar Muhendisligi Bolumu Kucukyali E5 KavsagiInonu Cad4 Kucukyali TR-34840 Istanbul Turkey

ISBN: (纸本)9781728172064

Processing human speech with the use of digital technologies leads to several important fields of research. Speech-to-text and lip-syncing are among the instances of relevant prominent research areas. In this regard, audio-visualization of acoustic signals, providing visual aid in real-time for disabled people, and realization of text-free animation applications are just to name a few. Therefore, in this study, a language-independent lip-sync method that is based on extended linear predictive coding is proposed. The proposed method operates on baseband electrical signal that is acquired by a standard single-channel off-the-shelf microphone and exploits the statistical characteristics of acoustic signals produced by human speech. In addition, the proposed method is implemented on an embedded system, tested, and its performance is evaluated. Results are given along with discussions and future directions.

关键词： formant frequency linear predictive coding lip sync

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HIGH-QUALITY AUDIO coding USING MULTIPULSE LPC

HIGH-QUALITY AUDIO CODING USING MULTIPULSE LPC

引用

1990 INTERNATIONAL CONF ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING ( ICASSP 90 )

作者： SINGHAL, S Bell Communications Research Inc. Morristown NJ USA

Experiments are described in coding broadband audio using multipulse linear predictive coding (LPC). It is possible to obtain stable LPC filters that model sinusoids closely and to include perceptual masking in these coders. The quantization of both the LPC and multipulse parameters is also examined, and it is found that multipulse can compensate for quantization error in LPC filters. With appropriate perceptual masking, these coders can provide high quality and audio output. At 128 kb/s, the coders achieved typical SNR values of 35-40 dB in simulations.< >

关键词： Audio coding linear predictive coding Filters Quantization

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An information theoretic perspective on the speech spectrum process

An information theoretic perspective on the speech spectrum ...

引用

7th IEEE Workshop on Speech coding

作者： Nordén, F Eriksson, T Hedelin, P Chalmers Univ Technol Dept Signals & Syst Informat Theory Lab S-41296 Gothenburg Sweden

ISBN: (纸本)0780364163

In this paper, an information theoretic study of properties of the speech spectrum process is performed. Various techniques to model the probability density function are applied to the spectrum source to compute rate-distortion functions. We estimate the difference in the required rate to achieve a given distortion for three different scenarios: interframe gain exploitation, low-pass filtering of LPC vectors and increased speech signal bandwidth. We obtain fairly consistent results for the different methods of calculating rate-distortion functions. The results show that for close to transparent LPC quantization we, gain 4-6 bits per frame by exploiting first order interframe correlation. The new idea of using low-pass filtered LPC vectors has shown to decrease the coding cost with 1-3 bits per frame, depending on the cutoff frequency.

关键词： Bit rate Costs Distortion Filtering linear predictive coding Low pass filters Probability density function Quantization Rate-distortion Speech processing

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Application of DSP Speech Synthesis System on Service Robots

Application of DSP Speech Synthesis System on Service Robots

引用

CACS International Automatic Control Conference (CACS 2014)

作者： Huang, Guo-Shing Lu, Yan-An Natl Chin Yi Univ Technol Inst Elect Engn Taichung Taiwan

ISBN: (纸本)9781479945849

The main purpose of this paper is to perform the voice signal processing and synthesis to apply to service robots using DSP TMS320C6713. In this study, we select the C language program written with DSP;the voice signal is restored to play. The speech synthesis is implemented using linear predictive coding (LPC) approach in the paper. Because the LPC synthesis is a coding technique of time waveform, one can reduce the transmission rate of signal in time domain;save perfect voice messages. It can be obtained a very high naturalness and clarity for the synthesis of a group of words. We scored the synthesis results by subjective listening test ways.

关键词： DSP speech recognition service robots linear predictive coding speech synthesis

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A low-complexity doubletalk detector for acoustic echo cancellers in packet-based telephony

A low-complexity doubletalk detector for acoustic echo cance...

引用

Workshop on Applications of Signal Processing to Audio and Acoustics

作者： Gordy, JD Goubran, RA Carleton Univ Dept Syst & Comp Engn Ottawa ON K1S 5B6 Canada

ISBN: (纸本)0780391543

This paper investigates a normalized cross-correlation-based doubletalk detector for acoustic echo cancellers. A novel low-complexity version is presented suitable for implementation on IP-enabled telephones employing LPC-based speech coders. In particular. the algorithm obtains a decorrelated input signal and decorrelation filter coefficients directly from the speech decoder. The proposed algorithm is implemented using ITU G.729. and simulation results are collected for a typical room. Calibration data are obtained and shown to be independent of the speech input signal. It is also shown that the proposed algorithm has the same performance as a full-complexity version employing the same decorrelation filter order.

关键词： Internet telephony decorrelation echo suppression filtering theory linear predictive coding speech coding vocoders IP-enabled telephones ITU G.729 LPC acoustic echo cancellers decorrelation filter coefficients low-complexity doubletalk detector packet-ba

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