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检索条件"主题词=Linear predictive coding"

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A comparison between multi-channel audio equalization filters using warping

A comparison between multi-channel audio equalization filter...

引用

IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

作者： Bharitkar, S Kyriakakis, C Univ So Calif Integrated Media Syst Ctr Los Angeles CA 90089 USA

ISBN: (纸本)0780378504

Typically, room equalization techniques do not focus on designing filters that equalize the room responses at perceptually relevant frequencies. Thus, by performing Bark warping of the room responses and using lower order spectral models it is possible to design low order psycho-acoustically motivated equalization filters. In this paper, we compare the performance, through experiments, between the traditional RMS averaging filter (with and without warping to the Bark scale) and our pattern recognition based multiple listener equalization filter with warping [2]. It is shown that the our pattern recognition filter, using warping, outperforms the RMS averaging filter (with and without warping to the Bark scale).

关键词： Electronic mail Frequency response linear predictive coding linear systems Nonlinear filters Pattern recognition predictive models Psychoacoustic models Psychology Signal processing

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Matrix-based linear predictive compression of multi-channel surface EMG signals

Matrix-based linear predictive compression of multi-channel ...

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33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Carotti, Elias S. G. de Martin, J. C. Merletti, Roberto Farina, Dario Politecn Torino DAUIN Corso Duca Abruzzi 24 I-10129 Turin Italy Politecn Torino DELEN I-10129 Turin Italy Aalborg Univ Ctr Sensory Motor Interact Dept Hlth Sci & Tech Aalborg Denmark

ISBN: (纸本)9781424414833

We propose a linear predictive coding technique for multi-channel electromyographic (EMG) recordings. The signals are acquired using two-dimensional grid of electrodes which generate strongly correlated signals. Previous work only considered spectral redundancy across the signal matrix. In this paper we exploit the correlation present in the residual signals, i.e., the signals after the short term prediction. The proposed technique achieves a compression ratio of about 1 divided by 9, i.e., slightly better than spectral-only decorrelation methods, but with a strong increase of approximately 3.2 dB SNR in the quality of the reconstructed waveform.

关键词： data compression linear predictive coding electromyography

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A packet loss recovery method using packet arrived behind the playout time for CELP decoding

A packet loss recovery method using packet arrived behind th...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Serizawa, M Ito, H NEC Corp Ltd Multimedia Res Labs Kawasaki Kanagawa 2168555 Japan

ISBN: (纸本)0780374029

This paper proposes a packet loss recovery method using a packet arrived behind the playout time for CELP decoding. The proposed method recovers synchronization of the filter states between encoding and decoding in the period following packet loss. The recovery is performed by replacing the degraded filter states with ones adaptively calculated from the late arrival packet in decoding. When the proposed method is applied to the AMR speech decoder, it improves the segmental SNR by 0.2 to 1.8 dB at packet loss rates of I to 10 116, where all the packet losses occur due to their late arrival. The subjective test results show that five-grade mean opinion scores are improved by 0.3 and 0.2 at a packet loss rate of 5% at speech coding bitrates of 7.95 and 12.2 kbps, respectively.

关键词： speech coding decoding linear predictive coding synchronisation packet loss recovery method CELP decoding synchronization recovery filter states late arrival packet AMR speech decoder segmental SNR subjective test results five-grade mean opinion scores speech coding code excited linear predictive coding playout time VoIP applications 7.95 kbit/s 12.2 kbit/s

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ADPCM with adaptive pre- and post-filtering for delay-free audio coding

ADPCM with adaptive pre- and post-filtering for delay-free a...

引用

32nd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Holters, Martin Pabst, Oliver Zoelzer, Udo Helmut Schmidt Univ Univ Fed Armed Forces Dept Signal Proc & Commun Hamburg Germany

ISBN: (纸本)9781424407286

We propose an extension of ADPCM that includes adaptive pre- and post-filtering to achieve spectral shaping of the coding noise. The advantage of this coding scheme is that it allows a realization without algorithmic delay by making the filters backwards-adaptive. The measurements we present indicate that the addition of adaptive pre- and post-filtering to ADPCM results in a significant improvement in perceived audio quality. We therefore believe that the proposed system is a viable way to near-transparent lossy audio coding without algorithmic delay.

关键词： audio coding linear predictive coding adaptive filters

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Multi-track codebook in low-rate CELP coding

Multi-track codebook in low-rate CELP coding

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IEEE International Symposium on Industrial Electronics

作者： Guerchi, D. Louzi, A. UAE Univ Coll Informat Technol Jimi 1POB 17555 Al Ain U Arab Emirates Ericsson Canada Montreal PQ H4P2N2 Canada

ISBN: (纸本)9781424404964

In this work, we propose a new technique to model the fixed codebook in low-rate code-excited linear prediction coders. The multi-track codebook presented in this paper reduces the coder bit-rate while maintaining a high speech quality. Unlike the algebraic codebook that is adopted in the 8 kb/s G.729 standard, the multi-track representation uses a pulse density of five pulses per 10 ms speech frame, and permits various pulse magnitudes. Four tracks of disjoint pulse positions sets are constructed and evaluated for each analysis frame. However, only the pulses of the optimal track are quantized and transmitted to the receiver. While the search of the best individual pulse positions is limited to one segment interval of 2 ms, the pulse magnitudes are optimized for the whole analysis frame. The excitation signal in the multi-track codebook is encoded with 2.2 kb/s. Informal test listening and objective test measures have revealed that at the same bit allocation, the multi-track codebook results in significantly higher objective and subjective coded speech quality than the algebraic codebook.

关键词： linear predictive coding speech coding testing coder bit-rate disjoint pulse positions high speech quality informal test listening low-rate CELP coding low-rate code-excited linear prediction coders multitrack codebook objective test pulse density

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A SOM-Based Analysis of Early Prosodic Acquisition of English by Brazilian Learners: Preliminary Results

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8th International Workshop on Self-Organizing Maps (WSOM)

作者： Silva, Ana Cristina C. Macedo, Ana Cristina P. Barreto, Guilherme A. Univ Estadual Piaui Dept Linguist Corrente Brazil Univ Fed Ceara Dept Linguist Fortaleza Ceara Brazil Univ Fed Ceara Dept Teleinformat Engn Fortaleza Ceara Brazil

ISBN: (纸本)9783642215667

In this paper the SOM is used in an exploratory analysis of transfer phenomena from first language (L1) to the second language (L2) related to word/lexical stress. The basic hypothesis tested is whether the parameterization of the speech signal of the learner's utterances by standard signal processing techniques, such as linear predictive coding (LPC), used to encode the input of the network results in efficient categorization of speakers by the SOM. Preliminary results indicates that the combination LPC+SOM is indeed able to produce well-defined clusters of speakers that possess similarities regarding the transfer of stress patterns among Brazilian students in learning English as a foreign language.

关键词： Self-organizing map word/lexical stress linear predictive coding U-matrix

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Energy-Aware Compress and Forward systems for Wireless Monitoring of Time-Varying Fields

Energy-Aware Compress and Forward systems for Wireless Monit...

引用

21st Annual IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC)

作者： Savazzi, S. Molteni, D. Spagnolini, U. Politecn Milan Dipartimento Elettron & Informaz Milan Italy

ISBN: (纸本)9781424480166

Dense monitoring of time-varying 2D field by Wireless Sensor Networks (WSN) needs to define new strategies to aggregate and encode data according to 2D sensor deployments. In this paper optimal design of communication protocol for WSN is conveniently cast for a set of linear sensor networks (sensor arrays) synchronously monitoring a correlated 2D time-varying field (e. g., for tracking time variations). Sensors aggregate data hop-by-hop by compressing the new samples gathered from the sensor board and forwarding towards the next sensor in the route. Compression is based on a linear predictive encoding to exploit the correlation properties of the field. Cross-layer design of source, channel coding and medium access control are jointly analyzed for optimal resource allocation and interference management to minimize energy consumption. Seismic monitoring application is analyzed to corroborate the design approach.

关键词： access protocols channel coding channel coding communication protocol dense monitoring energy-aware compress and forward systems linear predictive coding linear predictive encoding linear sensor networks medium access control seismic monitoring application sensor arrays sensor arrays source coding source coding time-varying 2D field time-varying fields wireless monitoring wireless sensor networks wireless sensor networks

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Energy-aware data compression for wireless sensor networks

Energy-aware data compression for wireless sensor networks

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32nd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Puthenpurayil, Sebastian Gu, Ruirui Bhattacharyya, Shuvra S. Univ Maryland Inst Adv Comp Studies Dept Elect & Comp Engn College Pk MD 20742 USA

ISBN: (纸本)1424407281

Data compression techniques have extensive applications in power-con strained digital communication systems, such as in the rapidly-developing domain of wireless sensor network applications. This paper explores energy consumption tradeoffs associated with data compression, particularly in the context of lossless compression for acoustic signals. Such signal processing is relevant in a variety of sensor network applications, including surveillance and monitoring. Applying data compression in a sensor node generally reduces the energy consumption of the transceiver at the expense of additional energy expended in the embedded processor due to the computational cost of compression. This paper introduces a methodology for comparing data compression algorithms in sensor networks based on the figure of merit DIE, where D is the amount of data (before compression) that can be transmitted under a given energy budget E for computation and communication. We develop experiments to evaluate, using this figure of merit, different variants of linear predictive coding. We also demonstrate how different models of computation applied to the embedded software design lead to different degrees of processing efficiency, and thereby have significant effect on the targeted figure of merit.

关键词： DSP software lossless data compression linear predictive coding low power design wireless sensor networks

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Doppler-variant modeling of the vocal tract

Doppler-variant modeling of the vocal tract

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33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Heim, Axel Sorger, Uli Hug, Florian Univ Ulm Inst Telecommun & Appl Informat Theory D-89069 Ulm Germany Univ Luxembourg Comp Sci & Commun Luxembourg Luxembourg

ISBN: (纸本)9781424414833

A common technique to deploy linear prediction to non-stationary signals is time segmentation and local analysis. Variations of a process within such a segment cause inaccuracies. In this paper, we model the temporal changes of linear prediction coefficients (LPCs) as a Fourier series. We obtain a compact description of the vocal tract model limited by the predictor order and the maximum Doppler frequency. Filter stability is guaranteed by all-pass filtering, deploying the human ear's insensitivity to absolute phase. The periodicity constraint induced by the Fourier series is counteracted by oversampling in the Doppler domain. With this approach, the number of coefficients required for the vocal tract modeling is significantly reduced compared to a LPC system with block-wise adaptation while exceeding its prediction gain. As a by-product it is found that the Doppler frequency of the vocal tract is in the order of 10 Hz. A generalization of the algorithm to an auto-regressive moving average model with time-correlated filter coefficients is straight forward.

关键词： speech coding linear predictive coding Fourier series

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ENVELOPE MODELING FOR SPEECH AND AUDIO PROCESSING USING DISTRIBUTION QUANTIZATION 23

ENVELOPE MODELING FOR SPEECH AND AUDIO PROCESSING USING DIST...

引用

23rd European Signal Processing Conference (EUSIPCO)

作者： Jaehnel, Tobias Baeckstroem, Tom Schubert, Benjamin Univ Erlangen Nurnberg Int Audio Labs Erlangen D-91054 Erlangen Germany Fraunhofer IIS D-91058 Erlangen Germany

ISBN: (纸本)9780992862633

Envelope models are common in speech and audio processing: for example, linear prediction is used for modeling the spectral envelope of speech, whereas audio coders use scale factor bands for perceptual masking models. In this work we introduce an envelope model called distribution quantizer (DQ), with the objective of combining the accuracy of linear prediction and the flexibility of scale factor bands. We evaluate the performance of envelope models with respect to their ability to reduce entropy as well as their correlation to the original signal magnitude. The experiments show that in terms of entropy, distribution quantization and linear prediction arc comparable, whereas for correlation, distribution quantization is better. Furthermore the coefficients of distribution quantization are independent and thus more flexible and easier to quantize than linear predictive coefficients.

关键词： Speech coding linear predictive coding signal modeling

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