LSF Parameters Research Articles

Fast Adaptive Reconstruction Algorithm for Line Spectrum Pair Parameters Based on Compressed Sensing

Line Spectrum Pair Parameters (LSF) are important parameters in low rate speech coding. Implementing LSF transparent quantization with as few bits as possible is a research hotspot in the field of low-rate speech coding. Based on the theory of compressed sensing, this paper proposes a new adaptive reconstruction algorithm. At the encoding end, the algorithm performs frame type discrimination on the LSF of single frame or superframe after voice activation detection and then applies the compressed sensing method to observe and quantize the LSF parameters of the single frame or superframe. At the decoding end, parameters are adaptively selected according to every single frame or superframe type, so that the number of bits required to reconstruct the LSF is reduced. The experimental results show that the proposed algorithm can reduce the number of bits required for LSF reconstruction and improve the reconstruction quality under the appropriate codebook storage and search complexity.

LSF Vector Quantizer for 2.4kb/s Codec Based on Speech Unvoiced/Voiced Classification

In order to reduce bit-rate and still maintain fine distortion performance,this paper proposed a method of LSF quantization based on speech unvoiced/voiced classification. This method use differential LSF parameters from unvoiced/voiced database to train codebook. And using this method can suppress the quantization error propagation caused by directly vector quantization of LSF parameters. Experimental results show that using this method to quantify LSF will have a better quality while allocating the same number of bits.

Research on EEDSVQ of LSF Parameters Based on Voiced and Unvoiced Classification

Research on EEDSVQ of LSF Parameters Based on Voiced and Unvoiced Classification

A NEW DISTORTION MEASURE FOR PARAMETER QUANTIZATION BASED ON MELP

Parameter quantization is very important for the synthetic speech quality of the vocoder. A new distortion measure for pitch as well as lsf quantization in ultra low bit rate Vocoder, whose parameters for several consecutive frames are grouped into a vector and jointly quantized to obtain high coding efficiency, is proposed based on mixed excitation linear prediction(MELP) vocoder. The product of sum of band pass voicing coefficients and gain parameter is used to denote the weighting factor of pitch as well as lsf parameters of current speech frame in the consecutive frames using weighted squared Euclidean distance measure to search the vector codebook. Comparing with the traditional method for a constant weighting factor by distinguishing Voiced/Unvoiced(UV) pattern of each speech frame, objective test results show that the quantization distortion of pitch is reduced by 3.3% and the mean opinion score (MOS) is increased by almost 0.1(3.5%).

Adaptive Long-Term Coding of LSF Parameters Trajectories for Large-Delay/Very- to Ultra-Low Bit-Rate Speech Coding

This paper presents a model-based method for coding the LSF parameters of LPC speech coders on a basis, that is, beyond the usual 20-30 ms frame duration. The objective is to provide efficient LSF quantization for a speech coder with large delay but very- to ultra-low bit-rate (i.e., below 1 kb/s). To do this, speech is first segmented into voiced/unvoiced segments. A Discrete Cosine model of the time trajectory of the LSF vectors is then applied to each segment to capture the LSF interframe correlation over the whole segment. Bi-directional transformation from the model coefficients to a reduced set of LSF vectors enables both efficient sparse coding (using here multistage vector quantizers) and the generation of interpolated LSF vectors at the decoder. The proposed method provides up to 50% gain in bit-rate over frame-by-frame quantization while preserving signal quality and competes favorably with 2D-transform coding for the lower range of tested bit rates. Moreover, the implicit time-interpolation nature of the long-term coding process provides this technique a high potential for use in speech synthesis systems.

Block Constrained Trellis Coded Vector Quantization of LSF Parameters for Wideband Speech Codecs

In this paper, block constrained trellis coded vector quantization (BC-TCVQ) is presented for quantizing the line spectrum frequency parameters of the wideband speech codec. Both a predictive structure and a safety-net concept are combined into BC-TCVQ to develop the predictive BC-TCVQ. The performance of this quantization is compared with that of the linear predictive coding vector quantizer used in the AMR-WB codec, demonstrating reductions in spectral distortion.

Atributos eficientes em reconhecimento automático de voz distribuído

Este artigo descreve e examina atributos eficientes para sistemas de reconhecimento automático de voz (RAV) distribuídos. Os atributos são extraídos a partir de transformações dos parâmetros dos codificadores utilizados em redes móveis celulares e em redes IP que são as principais aplicações visadas em reconhecimento de voz distribuído. Em particular, são analisados atributos obtidos de parâmetros LPC e LSF, em intervalos de 10 ms e 20 ms, utilizando um reconhecedor independente do locutor baseado em HMM. É examinado também o melhor domínio para se realizar a interpolação dos parâmetros em reconhecimento automático de voz distribuído.

A Fast Encoding Technique for Vector Quantization of LSF Parameters

A fast encoding technique is described for vector quantization (VQ) of line spectral frequency parameters. A reduction in VQ encoding complexity is achieved by using a preliminary test that reduces the necessary codebook search range. The test is performed based on two criteria. One criterion uses the distance between a specific single element of the input vector and the corresponding element of the codevectors in the codebook, The other criterion makes use of the ordering property of LSF parameters. The fast encoding technique is implemented in the enhanced variable rate codec (EVRC) encoding algorithm. Simulation results show that the average searching range of the codebook can be reduced by 44.50% for the EVRC without degradation of spectral distortion (SD).

Strategies to improve the performance of very low bit rate speech coders and application to a variable rate 1.2 kb/s codec

This paper presents several strategies to improve the performance of very low bit rate speech coders and describes a speech codec that incorporates these strategies and operates at an average bit rate of 1.2 kb/s. The encoding algorithm is based on several improvements in a mixed multiband excitation (MMBE) linear predictive coding (LPC) structure. A switched-predictive vector quantiser technique that outperforms previously reported schemes is adopted to encode the LSF parameters. Spectral and sound specific low rate models are used in order to achieve high quality speech at low rates. An MMBE approach with three sub-bands is employed to encode voiced frames, while fricatives and stops modelling and synthesis techniques are used for unvoiced frames. This strategy is shown to provide good quality synthesised speech, at a bit rate of only 0.4 kb/s for unvoiced frames. To reduce coding noise and improve decoded speech, spectral envelope restoration combined with noise reduction (SERNR) postfilter is used. The contributions of the techniques described in this paper are separately assessed and then combined in the design of a low bit rate codec that is evaluated against the North American Mixed Excitation Linear Prediction (MELP) coder. The performance assessment is carried out in terms of the spectral distortion of LSF quantisation, mean opinion score (MOS), A/B comparison tests and the ITU-T P.862 perceptual evaluation of speech quality (PESQ) standard. Assessment results show that the improved methods for LSF quantisation, sound specific modelling and synthesis and the new postfiltering approach can significantly outperform previously reported techniques. Further results also indicate that a system combining the proposed improvements and operating at 1.2 kb/s, is comparable (slightly outperforming) a MELP coder operating at 2.4 kb/s. For tandem connection situations, the proposed system is clearly superior to the MELP coder.

Safety-net pyramid VQ of LSF parameters for widebandspeech codecs

The authors propose a predictive pyramid VQ (PPVQ) that uses the safety-net concept to quantise the LSF parameters of wideband speech. A first-order auto-regressive (AR) predictor is used to predict the LSF parameters. To quantise the prediction error signal effectively, a pyramid VQ (PVQ) is used while a memoryless PVQ is used to encode low-correlation frames. By combining the PPVQ and the memoryless PVQ, denoted ‘safety-net pyramid VQ’, the advantages of both quantisation methods can be utilised. The average SD value of a safety-net pyramid VQ is 0.35 dB less than that of a predictive split VQ (PSVQ). The safety-net pyramid VQ also requires low complexity and does not require any memory for the codebook storage.

A fast-search algorithm for the predictive split VQ of LPC parameters

A fast-search method is introduced to reduce the searching range of a codebook using the ordering property of LSF parameters. We apply the proposed fast-search method to the G.723.1 coder with the predictive split VQ structure. Simulation results indicate that using this fast-search method, the average searching range of the codebook can be reduced by 20.1% without degradation of the spectral distortion (SD) and additional increases of memories. The reduction of 20.1% in the average searching range results in the decrease of 19.1%, 20.1%, 19.4%, and 12.2% in the number of additions, subtractions, multiplies, and comparisons, respectively.

Predictive pyramid vector quantisation ofLSF parameters

The authors propose predictive pyramid vector quantisation (PPVQ) to quantise LSF parameters. For prediction of the LSF parameters, a 4th order moving average (MA) predictor is used. To quantise the prediction error signal effectively, pyramid VQ (PVQ) is considered. The performance of PPVQ has an average spectral distortion (SD) of 1.06 dB at 22bit/frame.

Efficient adaptive vector quantization of LPC parameters

This correspondence presents a new two-stage adaptive vector quantizer of LSF parameters in LPC speech coding. The first codebook is adapted by a partition-delete operation, whereas the code-vectors of the second codebook remain unchanged. The objective and subjective evaluations show that the proposed scheme offers transparent quantization with 22 b/frame.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

An LSF quantizer for the North-American half-rate speech coder

The design of the half-rate coder for the North-American cellular communication standard poses a challenging problem. The desired speech quality is that of the full-rate coder, but the total bit rate specified for the half-rate is only 6.5 kb/s. Since the mobile communication channel is very noisy, the use of error-correcting codes is necessary, which leaves only about 4 kb/s for the actual speech information. This very restricted bit budget requires that fewer than 30 b be allotted to the quantization of the LSF parameters, which precludes the use of scalar quantization. In this work, a quantizer design procedure which is tailored for the transmission frame structure of the North-American (TIA) cellular communication standard is presented. It is basically a split vector quantizer which makes use of interframe prediction to lower the number of bits required for quantization. Since the prediction is performed only on every other frame, the error propagation is limited. The quantization technique yields good performance in the 26-27 b/frame range, and the complexity of its implementation is less than 0.5 Mip.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>