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检索条件"主题词=Linear Predictive Coding"

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A bitstream scalable audio coder using a hybrid WLPC-wavelet representation

A bitstream scalable audio coder using a hybrid WLPC-wavelet...

引用

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

作者： D. Ning M. Deriche School of Electrical and Electronic Systems Engineering Queensland University of Technology Australia

In this paper, we present a novel bitstream scalable audio coder. In the proposed coder, the full bandwidth of input audio is first split into two. A hybrid WLPC-wavelet representation is used to encode the low frequency components (<11 kHz). In this method, the excitation to the WLPC synthesis filter is decomposed into subbands using a wavelet filterbank, and perceptually encoded. Two stage quantisation of the wavelet coefficients is used to provide scalability. The high frequency components of the input are assumed to be noisy, and efficiently encoded using an LPC noise model. The output bitstream is capable of being decoded at rates between 16 kbit/s and 80 kbit/s. As the bitrate increases, so too does the signal quality. At 80 kbit/s, the quality is near transparent. At the intermediate rates, the coder gives comparable performance to the MPEG layer III coder, when the MPEG coder operates at similar, but fixed, bitrates.

关键词： Frequency Bit rate Bandwidth Filter bank Quantization Wavelet coefficients Scalability Low-frequency noise linear predictive coding Decoding

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A time-scaling approach to detect the closely spaced segments of noisy speech signals

A time-scaling approach to detect the closely spaced segment...

引用

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

作者： M. Pwint F. Sattar School of Electrical and Electronic Engineering Nanyang Technological University Singapore

A segmentation algorithm for noisy speech signal is introduced. The combination of the time scaling technique and binary Walsh transform is the fundamental concept of this method. This approach can also detect the gaps which are closely located among the signal components while segmenting the whole noisy waveform. The location of speech segments and gaps are determined from the reconstructed time-scaled signal. It is found that using Walsh transform at the reconstruction phase of time scaling process provides high capability in separating the signal components from narrow gaps. Although this method is less complex, the experimental results are satisfactory in the absence of a priori knowledge about speech and noise.

关键词： Acoustic noise Speech enhancement Fourier transforms Signal detection Proposals Entropy linear predictive coding Cepstrum Wavelet transforms Background noise

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On the detection of closely-spaced noisy audio segments

On the detection of closely-spaced noisy audio segments

引用

International Conference on Information, Communications and Signal Processing

作者： M. Pwint F. Sattar School of Electrical and Electronic Engineering Nanyang Technological University Singapore

A vocoder based segmentation algorithm for audio signal is presented. The signal to be detected consists of multi-components, which are closely located in time and buried under high level of noise. While detecting the signal components, this method can also estimate the duration of the narrow gaps between them. The combination of a vocoder and binary Walsh transform is the fundamental concept of this method. The location of audio segments and gaps are determined from the output waveform of the vocoder. It is found that using Walsh transform at the synthesis stage of a vocoder provides high capability in separating the signal components from narrow gaps. Although this method is computationally less expensive, the experimental results are satisfactory in the absence of any a priori knowledge about the signal and noise.

关键词： Vocoders Noise level Signal detection Signal synthesis Speech recognition Indexing Cepstral analysis Speech analysis Entropy linear predictive coding

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Reduced complexity LSF vector quantization with switched-adaptive prediction

Reduced complexity LSF vector quantization with switched-ada...

引用

International Symposium on Image and Signal Processing and Analysis

作者： D. Petrinovic Faculty of Electrical Engineering and Computing University of Zagreb Croatia

A modification of a classical predictive vector quantization (PVQ) technique with switched-adaptive prediction for line spectrum frequencies (LSF) quantization is proposed in this paper, enabling significant reduction in complexity. Lower complexity is achieved through use of higher number of switched prediction matrices but with reduced number of their nonzero elements. The structures of such matrices and optimal matrix elements are obtained to maximize the quantizer closed-loop prediction gain. A comparison of the proposed quantizer to the ones with full prediction matrices as well as to the quantizer incorporating diagonal matrices is given. The effectiveness of the proposed approach is shown and the trade-off between complexity and quality of the quantizer is analyzed.

关键词： Vector quantization Sparse matrices Frequency Covariance matrix Speech coding linear predictive coding Bit rate Computational complexity Image reconstruction Signal processing

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Time scale modification of noises using a spectral and statistical model

Time scale modification of noises using a spectral and stati...

引用

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

作者： P. Hanna M. Desainte-Catherine SCRIME-LaBRI Université de Bordeaux 1 Talence France SCRIME Université de Bordeaux 1 Talence France

Some natural sounds, such as speech parts can essentially be considered as noises. For instance, models suppose noisy parts of sounds as weak parts and apply basic approximations. But transformations such as time stretching do not preserve the noisy characteristics of sounds. Moreover, we show that those transformations introduce artificial intensity variations. In this paper we propose a spectral model for noise modeling which takes into account the statistical properties of such sounds. The analysis is based on the classical spectral models. The synthesis consists of randomly defining sinusoidal components. These components are then added using the adapted overlap-add method to keep statistical moments constant. Time scaling operations using this approach are described. Experiments on artificial sounds (filtered white noises) as well as natural sounds such as consonants and whispered vowels, show impressive enhancement in quality. Infinite time stretching transformations of such noises can be perfectly performed.

关键词： Acoustic noise Stochastic resonance White noise Stochastic processes Speech Signal synthesis linear predictive coding Frequency Vocoders Instruments

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The influence of reverberation on multichannel equalization: an experimental comparison between methods

The influence of reverberation on multichannel equalization:...

引用

Asilomar Conference on Signals, Systems & Computers

作者： S. Bharitkar C. Kyriakakis Integrated Media Systems Center University of Southern California Los Angeles CA USA

Room equalization is important for delivering high-quality audio in multiple listener environments and for improving speech recognition rates. Lower order equalization filters can be designed at perceptually relevant frequencies through warping. However, one of the major factors that affects multi-channel equalization performance is the reverberation of the room. In this paper, we compare the equalization performance of our method (S. Bharitkar et al., Nov. 2002) to the industry standard root-mean-square (RMS) method, through the image method. It is shown that our method outperforms the RMS method in terms of maintaining a lower spectral deviation, across multiple listener positions, when the reverberation time is increased.

关键词： Reverberation Nonlinear filters Frequency response linear predictive coding Speech processing Speech recognition Multilevel systems linear systems Transfer functions Acoustic reflection

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Spectrogram-based formant tracking via particle filters

Spectrogram-based formant tracking via particle filters

引用

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

作者： Yu Shi E. Chang Microsoft Research Asia China

The paper presents a particle-filtering method for estimating formant frequencies of speech signals from spectrograms. First, frequency bands corresponding to the analyzed formants are extracted via a two-step dynamic programming based algorithm. A particle-filtering method is then used to locate accurately formants in every formant area based on the posterior PDF described by a set of support points with associated weights. Formant trajectories of voiced frames of a group of 81 utterances were manually tracked and labeled, partly for model training and partly for algorithm evaluation. In the experiments, the proposed method obtains average estimation errors of 72, 115, and 113 Hz for the first three formants, respectively, whereas the LPC based method induces 118, 172, and 250 Hz deviations. The experimental results show that the formants estimated by the proposed method are quite reliable and the trajectories are more accurate than LPC.

关键词： Particle tracking Particle filters Frequency estimation linear predictive coding Speech Spectrogram Algorithm design and analysis Dynamic programming Heuristic algorithms Trajectory

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CELP based speech coding with fine granularity scalability

CELP based speech coding with fine granularity scalability

引用

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

作者： Fang-Chu Chen I-Hsien Lee Computer and Communications Research Laboratories Industrial Technology and Research Institute Hsinchu Taiwan

ISBN: (纸本)0780376633

General audio and video with fine granularity scalability (FGS) has become favored in next generation multimedia coding standards due to its high flexibility in channel rate adaption. However, the FGS phenomenon has not yet been fitted into existing speech codecs. We introduce the FGS feature to the code excited linear prediction (CELP) based speech coding algorithm by adjusting the amount of transmitted fixed excitation information. We further improve the algorithm by relaxing the constraints and re-ordering the sequence of pulses. To achieve this target, we need to make modifications to the conventional coding algorithm, but the computation overhead is little and affected modules are few. As a consequence, developers can, in a short time, easily migrate their existing codec to one with the FGS advantage.

关键词： Speech coding Scalability Streaming media Decoding Speech codecs Bandwidth MPEG 4 Standard linear predictive coding Communication industry Computer industry

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Recognising moving hand shapes

Recognising moving hand shapes

引用

International Conference on Image Analysis and Processing

作者： E.-J. Holden R. Owens School of Computer Science & Software Engineering University of Western Australia Crawley WA Australia

The paper presents a new hand shape representation technique that characterises the finger-only topology of the hand, by adapting an existing technique from speech signal processing. From a moving hand sequence, the tracking algorithm determines the centre of the largest convex subset of the hand, using a combination of pattern matching and condensation algorithms. A hand shape feature represents the topological formation of the finger-only regions of the hand using a linear predictive coding parameter set called cepstral coefficients. Experimental results demonstrate the effectiveness of detecting the shape feature from motion sequences.

关键词： Shape Topology Signal processing algorithms linear predictive coding Cepstral analysis Feature extraction Pattern matching Speech recognition Autocorrelation Pixel

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A modified group vector quantization algorithm for speaker identification

A modified group vector quantization algorithm for speaker i...

引用

Midwest Symposium on Circuits and Systems (MWSCAS)

作者： M.F. Abu El-Yazeed N.S. Abdel Kader M.M. El-Henawy Department of Electronics & Comm. Faculty of Engineering Cairo University Giza Egypt National Telecommunication Institute Cairo Egypt

In this paper, a modification to the group vector quantization (GVQ) discriminative training algorithm is proposed to train VQ codebooks for closed set speaker identification. The proposed algorithm, referred to as modified GVQ (MGVQ), shifts the decision surfaces between speakers smoothly toward the Bayes limits. This is achieved by varying the learning rate during training iterations. The proposed MGVQ algorithm achieves higher speaker identification rate compared to the standard GVQ

关键词： Vector quantization Hidden Markov models Feature extraction Speech Spatial databases Iterative algorithms Testing linear predictive coding Cepstral analysis Clustering algorithms

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