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检索条件"主题词=Linear Predictive coding"

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A pseudo-cepstrum based short-term postfilter

A pseudo-cepstrum based short-term postfilter

引用

IEEE Workshop on Speech coding Proceedings

作者： Hong Kook Kim Hong-Goo Kang AT and T Research Laboratories Florham Park NJ USA

We propose an adaptive short-term postfilter for speech coders by incorporating the properties of the pseudo-cepstrum. Since the proposed postfilter implicitly has a characteristic of tilt compensation, it does not require an additional tilt compensation filter as conventional techniques. We derive a relationship between the parameters of the proposed postfilter based on a minimum phase distortion criterion, and show a simple tuning procedure for the parameters. It is also shown that the postfilter can be implemented with a lower order. By applying this postfilter to several international speech coding standards, we reduce the complexity of the speech coders while obtaining comparable performance to conventional approaches.

关键词： linear predictive coding Polynomials Speech coding Frequency Nonlinear filters Cepstrum Cepstral analysis Code standards Power harmonic filters Signal synthesis

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Toward naturalness in narrow-band speech compression

Toward naturalness in narrow-band speech compression

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： S. Ghaemmaghami Electronic Research Center Sharif University of Technology Iran

The paper addresses a new mixed model for characterizing LPC excitation on a 3-band basis through analyzing harmonic structure of the residual signal. In addition, a sub-frame based analysis is developed for detecting both aperiodic pulses and noisy signals, which plays a major role in reduction of perceptual errors introduced by some certain consonants. Preliminary results show that near natural speech is achieved at 1050 bps, allocated to the excitation parameters, suggesting superiority of the proposed coding scheme to the MELP-2400 coding standard, in the sense of perceptual quality of reconstructed speech.

关键词： Narrowband linear predictive coding Speech coding Signal analysis Acoustic noise Signal processing Spectral analysis Frequency estimation Power harmonic filters Noise reduction

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A weighted autocorrelation method for pitch extraction of noisy speech

A weighted autocorrelation method for pitch extraction of no...

引用

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

作者： H. Kobayashi T. Shimamura Department of Information and Computer Sciences Saitama University Urawa Japan

Pitch period (or fundamental frequency) extraction plays an important role on speech processing and has a wide spread of applications in systems associated with speech. Many pitch extraction methods have been proposed so far, but improvement in noisy environments is still a remaining subject. In this paper, we propose a modified version of the autocorrelation method which is well known to be robust against noise. Utilizing that the difference function (amplitude difference function) has similar characteristics with the autocorrelation function, the autocorrelation function is weighted by the reciprocal of the difference function. By simulation experiments based on continuous speech, it is shown that the proposed pitch extraction method behaves more robustly than the conventional methods against additive noise, and especially it is very effective at low signal-to-noise ratio.

关键词： Autocorrelation Working environment noise Data mining Speech enhancement linear predictive coding Speech processing Frequency Noise robustness Application software Signal to noise ratio

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MPEG-4 HVXC encoder realtime implementation on floating point DSP

MPEG-4 HVXC encoder realtime implementation on floating poin...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Kyeongok Kang Jin-Woo Hong Jinwoong Kim Dae Gwon Jeong Electronics and Telecommnications Research Institute Taejon South Korea Hankuk Aviation University Koyang South Korea

ISBN: (纸本)0780354826

In this paper, we describe the real-time implementation effort of MPEG-4 audio HVXC (Harmonic Vector eXcitation coding) encoder algorithm for very low bitrates, which has target applications from mobile communications to Internet telephony, on current high performance floating point DSP. Using the C-language and assembly-language level optimization for time-critical functional codes, utilizing the internal program memory of the DSP as the program cache, and further utilizing the internal data memory operation and DMA functionality we could get a goal of realtime operation of HVXC encoder both at 2 kbit/s and at 4 kbit/s.

关键词： MPEG 4 Standard Digital signal processing Bit rate linear predictive coding Decoding Speech coding Distortion measurement Internet telephony Vectors Mobile communication

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Real-time implementation of a 12.8 kbit/s LD-CELP speech codec

Real-time implementation of a 12.8 kbit/s LD-CELP speech cod...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： Zhang Xueying College of Information Engineering Taiyuan University of Technology Taiyuan China

The paper describes a 12.8 kbit/s LD-CELP speech coding algorithm on the basis of the G.728 speech codec, and presents the concept of codeword use frequency (CUF). The experimental results show that the method can decrease effectively the complexity of vector quantization codebooks and give better synthesis speech quality than other similar methods. The algorithm is implemented in duplex real-time on a high-speed DSP system with two TMS320C31 chips.

关键词： Speech codecs Speech synthesis Speech coding Filters Speech analysis Vector quantization Digital signal processing chips Delay linear predictive coding Frequency

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Multivariate-state hidden Markov models for simultaneous transcription of phones and formants

Multivariate-state hidden Markov models for simultaneous tra...

引用

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

作者： M. Hasegawa-Johnson Department of ECE University of Illinois Urbana-Champaign IL USA

A multivariate-state HMM-an HMM with a vector state variable-can be used to find jointly optimal phonetic and formant transcriptions of an utterance. The complexity of searching a multivariate state space using the Baum-Welch algorithm is substantial, but may be significantly reduced if the formant frequencies are assumed to be conditionally independent given knowledge of the phone. Operating with a known phonetic transcription, the multivariate-state model can provide a maximum a posteriori formant trajectory, complete with confidence limits on each of the formant frequency measurements. The model can also be used as a phonetic classifier by adding the probabilities of all possible formant trajectories. A test system is described which requires only nine trainable parameters per formant per phonetic state: five parameters to model formant transitions, and four to model spectral observations. Further simplifications were achieved through parameter tying.

关键词： Hidden Markov models State-space methods Viterbi algorithm Equations System testing Speech recognition Frequency measurement Vector quantization linear predictive coding Costs

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Improving the performance of ITU-T G.729A for VoIP

Improving the performance of ITU-T G.729A for VoIP

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： C. Montminy T. Aboulnasr School of Information Technology and Engineering University of Ottawa Ottawa ONT Canada Canada

The authors propose a novel method for improving the performance of ITU-T G.729A for Voice-over-IP (VoIP) applications. One of the most important features for a speech compression algorithm that is to be used for VoIP is that it be resistant to packet loss. Because of the memory that resides in the G.729A algorithm, a packet loss will not only cause degradations during the period of the loss but also following the packet loss due to the differing states of the encoder and decoder. The G.729A algorithm deals quite well with the packet loss period through error concealment, but it does not deal with the state error that follows the packet loss. The paper introduces a new scheme called recovery by reinitialization (RbR) that reduces this state error at minimal cost.

关键词： Internet telephony Compression algorithms Speech synthesis Delay Jitter predictive models Nonlinear filters Speech enhancement Costs linear predictive coding

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Design and performance of a 4.0 kbit/s speech coder based on frequency-domain interpolation

Design and performance of a 4.0 kbit/s speech coder based on...

引用

IEEE Workshop on Speech coding Proceedings

作者： U. Bhaskar S. Nandkumar K. Swaminathan Hughes Network Systems Germantown MD USA

The 4.0 kbit/s speech codec described is based on a frequency domain interpolative (FDI) coding technique, which belongs to the class of prototype waveform interpolation (PWI) coding techniques. The codec also has an integrated voice activity detector (VAD) and a noise reduction capability. The input signal is subjected to LPC analysis and the prediction residual is separated into a slowly evolving waveform (SEW) and a rapidly evolving waveform (REW) component. The SEW magnitude component is quantized using a hierarchical predictive vector quantization approach. The REW magnitude is quantized using a gain and a sub-band based shape. The SEW and REW phases are derived at the decoder using a phase model, based on a transmitted measure of voice periodicity. The spectral (LSP) parameters are quantized using a combination of scalar and vector quantizers. The 4.0 kbits/s coder has an algorithmic delay of 60 ms and an estimated floating point complexity of 21.5 MIPS. The performance of this coder has been evaluated using in-house MOS tests under various conditions such as background noise, channel errors, self-tandem, and DTX mode of operation, and has been shown to be statistically equivalent to ITU-T G.729 8 kbps codec across all conditions tested.

关键词： Delay estimation Automatic testing Speech codecs Frequency domain analysis Fault detection Prototypes Interpolation Detectors Noise reduction linear predictive coding

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Low-rate quantization of spectral information in a 4 kb/s pitch-synchronous CELP coder

Low-rate quantization of spectral information in a 4 kb/s pi...

引用

IEEE Workshop on Speech coding Proceedings

作者： D. Guerchi P. Mermelstein INRS-Télécommunications Université du Quàbec Montreal QUE Canada

A new efficient algorithm for quantizing the spectral information for a pitch-synchronous CELP (PSCELP) speech coder is proposed. LPC analysis in the PSCELP is carried out once per pitch period. Direct quantization of the pitch synchronous LSF vectors would lead to a variable-rate codec, which is inconsistent with the objective of achieving a fixed-rate speech coder operating at 4 kb/s. Hence, a linear trajectory of LSF vectors is selected which can be encoded by one LSF vector each 20 ms. This conversion exploits the high correlation between successive pitch periods of the LSF parameters to achieve joint quantization. A coding rate of 1.2 kb/s is achieved for the LSF information with no noticeable degradation. The proposed algorithm employs linear interpolation at the decoder to recover the spectral parameters for the individual pitch periods used in the pitch-synchronous reconstruction of the speech signal. The comparison simulation results show that this algorithm produces comparable performance to that of LSF's linear interpolation quantization in a time-synchronous CELP coder.

关键词： Quantization linear predictive coding Speech analysis Vectors Degradation Interpolation Bit rate Speech codecs Decoding Code standards

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The high-speed LSF transformation algorithm in CELP vocoder

The high-speed LSF transformation algorithm in CELP vocoder

引用

IEEE Region 10 International Conference TENCON

作者： DuckSu Na SoYeon Min MyungJin Bae Department of Information & Telecommunication Engineering Soongsil University Seoul South Korea

We propose a computation reduction method of the real root method that is mainly used in the CELP (code excited linear prediction) vocoder. The real root method is that if polynomial equations have real roots, we are able to find those and transform them into the LSF. However, this method takes much time to compute, because the root searching is processed sequentially in the frequency region. But, the important characteristic of the LSF is that most of the coefficients occur in a specific frequency region. So, the searching frequency region is ordered by each coefficient's distribution, and the coefficients are searched in the ordered frequency region. The transformation time can be reduced further by this method than by the sequential searching method in the frequency region. When we compare this proposed method with the conventional real root method, the experimental result is that the searching time was reduced by about 46% in average.

关键词： Vocoders linear predictive coding Frequency Speech analysis Polynomials Equations Speech synthesis Vector quantization Acoustic propagation Lips

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