The conversion between Immittance Spectral Pairs pseudo-cepstrum and Linear Predictive Coding coefficients

음성코딩 시 성도는 Linear Predictive Coding (LPC) 계수를 이용해서 모델링 한다. 일반적으로 LPC 계수는 양자화와 선형보간 관점에서 유리한 Line Spectral Frequency (LSF) 파라미터로 변경하여 사용한다. 10차 이상의 다차원 LSF 데이터를 벡터 양자화를 이용하여 직접 코딩하게 되면 벡터 내 상관관계 (intra-frame correlation)를 모두 이용할 수 있으므로 rate-distortion 관점에서는 높은 효율을 기대할 수 있다. 하지만, 계산량과 메모리 요구량이 높아져서 실제 코딩 시스템에서는 사용할 수 없게 되므로, 차원을 나누어 압축하는 Split Vector Quantization (SVQ)이 이용된다. 또한, LSF 데이터는 과거 벡터와의 벡터 간 상관관계 (inter-frame correlation)가 높으므로, 이를 이용한 Predictive Split Vector Quantization (PSVQ)이 사용되고 있다. PSVQ는 SVQ 보다 높은 rate-distortion 성능을 보인다. 본 논문에서는 음성 저장 장치를 위한 최적의 PSVQ를 구현하기 위해서 다수의 과거 프레임 정보와의 벡터 간상관관계 (inter-frame correlation)를 고려한 Multi-Frame AR-model 기반 SVQ (MF-AR-SVQ)를 제안하였다. 기존 PSVQ와 비교해 보았을 때, MF-AR-SVQ는 계산량과 메모리 요구량의 큰 증가 없이, 평균 spectral distortion 관점에서 약 1비트의 성능 향상을 보였다. For speech coding, a vocal tract is modeled using Linear Predictive Coding (LPC) coefficients. The LPC coefficients are typically transformed to Line Spectral Frequency (LSF) parameters which are advantageous for linear interpolation and quantization. If multidimensional LSF data are quantized directly using Vector-Quantization (VQ), high rate-distortion performance can be obtained by fully utilizing intra-frame correlation. In practice, since this direct VQ system cannot be used due to high computational complexity and memory requirement, Split VQ (SVQ) is used where a multidimensional vector is split into multilple sub-vectors for quantization. The LSF parameters also have high inter-frame correlation, and thus Predictive SVQ (PSVQ) is utilized. PSVQ provides better rate-distortion performance than SVQ. In this paper, to implement the optimal predictors in PSVQ for voice storage devices, we propose Multi-Frame AR-model based SVQ (MF-AR-SVQ) that considers the inter-frame correlations with multiple previous frames. Compared with conventional PSVQ, the proposed MF-AR-SVQ provides 1 bit gain in terms of spectral distortion without significant increase in complexity and memory requirement.

The aim of the study was to inspect the acoustic properties and sleep characteristics of a preapneic snoring sound. The feasibility of forecasting upcoming respiratory events by snoring sound was also investigated. Participants with habitual snoring or a heavy breathing sound during sleep were recruited consecutively. Polysomnography was conducted, and snoring-related breathing sound was recorded simultaneously. Acoustic features and sleep features were extracted from 30-second samples, and a machine learning algorithm was used to establish 2 prediction models. A total of 74 eligible participants were included. Model 1, tested by 5-fold cross-validation, achieved an accuracy of 0.92 and an area under the curve of 0.94 for respiratory event prediction. Model 2, with acoustic features and sleep information tested by Leave-One-Out cross-validation, had an accuracy of 0.78 and an area under the curve of 0.80. Sleep position was found to be the most important among all sleep features contributing to the performance of the 2 models. Preapneic sound presented unique acoustic characteristics, and snoring-related breathing sound could be deployed as a real-time apneic event predictor. The models, combined with sleep information, serve as a promising tool for an early warning system to forecast apneic events. Wang B, Yi X, Gao J, etal. Real-time prediction of upcoming respiratory events via machine learning using snoring sound signal. J Clin Sleep Med. 2021;17(9):1777-1784.

This paper presents a 3D model descriptor based on the linear prediction coding (LPC) coefficients to retrieve 3D objects. In this method, the 3D object is projected on the lateral surface of a cylinder parallel to one of its principal axes and centered at the centroid of the object. To improve efficiency, cylindrical projection is performed on three cylinders parallel to principal axes. Besides, the object is projected on a sphere centered at the centroid of the 3D object. The surface of object is converted to two-dimensional shapes as a result of performing the projections which preserves the geometric features of the object. Then, LPC coefficients are extracted from the two-dimensional projected shapes. These coefficients estimate the parameters of correlated signal, efficiently. Since the cylindrical and spherical projections of the 3D model are correlated surfaces, LPC coefficients describe the surfaces as well. The rotation normalization is performed employing the principal component analysis. Furthermore, the retrieval performance is enhanced employing SVM-OSS similarity measure which efficiently compares two model feature vectors. We implement an experimental comparison on PSB database of 3D models in order to demonstrate the performance of the proposed descriptor. Experimental results show the effectiveness of the proposed descriptor compared to current methods.

The importance of packet-based speech transmissions has grown since it offers cheaper and efficient communications. However, frame erasures are a common hurdle in these networks and concealment techniques are necessary to ensure a minimum quality of service. In this paper, we propose a mitigation technique focused on the reconstruction of the linear prediction coding (LPC) coefficients and the excitation signal of the lost frame by using a replacement technique. These replacements are obtained by means of a minimum mean square error estimation based on a source model of the speech parameters (LPC coefficients and the excitation signal). As this approach critically relies on the quantization and representation of the excitation signal, we explore the Haar wavelet transform as a novel approach to represent the excitation signal for error mitigation. Thus, this paper describes how optimal codebook and estimates can be computed in a Haar transformed domain. As a result, the excitation signal of a frame can be decomposed in several partitions where each one is independently reconstructed. Objective and subjective tests are conducted in order to assess the quality of the concealed speech signal resulting from our proposal. Both evaluations confirm noticeable improvements over the default mitigation method included in the two tested standard codecs, adaptive multirate, and Internet low bitrate codec.

Linear Predictive Coding coefficients are of the main extraction feature in digital forensic. In this paper, we perform several experiments focusing on the problems of environments recognition from audio particularly for forensic application. We investigated the effect of temporal Linear Predictive Coding coefficient as feature extraction on environment sound recognition to compute the Linear Predictive Coding coefficient for each frame for all files. The performance is evaluated against varying number of training sounds and samples per training file and compare with Zero Crossing feature and Moving Picture Experts Group-7 low level description feature. We use K-Nearest Neighbors as classifier feature to detect which the environment for any audio testing file. Experimental results show that higher recognition accuracy is achieved by increasing the number of training files and by decreasing the number of samples per training file.

Speech coders operating at low bit rates necessitate efficient encoding of the linear predictive coding (LPC) coefficients. In this work, we developed an optimized switched split vector quantization (SSVQ) scheme applied for encoding the LPC coefficients represented by the line spectral frequencies (LSF) parameters. It will be shown that the SSVQ provides better performance in terms of bit-rate, spectral distortion and computational complexity than the traditional split vector quantizer. We further applied our SSVQ encoding system, called LSF-SSVQ encoder, to quantize the LSF parameters of the 2.4 Kbits/s normalised speech coder MELP operating over an ideal noiseless channel.

Experimental results are reported in the area of speech modification for the purpose of correcting vocal tract resonance disorders. The linear predictive coding (LPC) coefficients and residual corresponding to an input speech signal are calculated, and a combination of nonlinear frequency warping and amplitude scaling is applied to produce modified LPC coefficients and then synthesize an output speech signal from these modified coefficients and the residual signal. The spectral envelope modification parameters are designed to adapt to the first two formant frequencies of the input speech. Cubic polynomial warping functions and amplitude scaling factors are found at vowel formant locations and are interpolated from these points over F1, F2 space. The results are compared with those obtained from frame-by-frame nonlinear spectral warping using a dynamic programming algorithm. >

Linear predictive coding (LPC) coefficients are widely used in a various speech coding method. Line spectrum pairs (LSP), as alternate representation of LPC presents some useful properties for quantization and transmission. This work investigates the computing algorithm of LSP and proposes an efficient method to isolate the LSP in pairs of roots in order to reduce the required computational complexity for root finding. A combined LSP computation and quantization are proposed and various correction techniques are investigated, in order to preserve the LSP order properties after quantization.

This paper deals with implementing a real-time gunshot detection algorithm on the digital signal processor TMS320C6713. The developed algorithm uses 3 linear predictive coding coefficients, energy in 3 spectral bands, and frame correlation. The audio input signal is continuously processed and signal frames considered to contain a gunshot are shortly signalized by an LED indicator. Experimental results achieved 82% correctly detected gunshots and 3% false alarms. The system should increase protection of wild elephants in Africa.

This paper presents a fast efficient codeword search algorithm for vector quantization (VQ) based on the Adaptive Multi-Rate Wideband (AMR-WB) speech codec. AMR-WB is the speech codec standardized by the 3GPP for the GSM and WCDMA 3G system. It is also selected as the ITU-T G.722.2 standards. In AMR-WB speech codec VQ codeword search section, full search algorithm is used to find matching codeword .This requires a huge amount of computation. For large-size codebooks and high-dimensional vector of AMR-WB, computational complexity is significantly higher.The proposed fast codeword search method selects the codeword ,which has the least absolute error component in VQ codebook, as the initial matching codeword. It can help to reduce computational complexity, whereas the reconstructed speech quality almost has no dropping. The article introduces the basic principle of the fast codeword search algorithm in detail. Simulation results show that the algorithm based on AMR-WB speech codec, can reduce the time of vector quantization codebook search and the w-PESQ of nine modes is no significant dropping.

A class of reverberant speech enhancement techniques involve processing of the linear prediction residual signal following Linear Predictive Coding (LPC). These approaches are based on the assumption that reverberation is mainly confined to the prediction residual and affects the LPC coefficients to a lesser extent. This chapter begins with a study on the effects of reverberation on the LPC parameters where mathematical tools from statistical room acoustics are used in the analysis. Consequently, a general framework for dereverberation using LPC is formulated and several existing methods utilizing this approach are reviewed. Finally, a specific method for processing a reverberant prediction residual is presented in detail. This method uses a combination of spatial averaging and larynx cycle-based temporal averaging. Experiments with a microphone array in a small office demonstrate the dereverberation and noise suppression of the spatiotemporal averaging method, showing up to a 5 dB improvement in segmental SRR and 0.33 in the normalized Bark spectral distortion score.

LPC parameters substitution for speech information hiding

The adaptive multi-rate wideband (AMR-WB) speech codec is widely used in modern mobile communication systems for high speech quality in handheld devices. Nonetheless, a major disadvantage is that vector quantization (VQ) of immittance spectral frequency (ISF) coefficients takes a considerable computational load in the AMR-WB coding. Accordingly, a binary search space-structured VQ (BSS-VQ) algorithm is adopted to efficiently reduce the complexity of ISF quantization in AMR-WB. This search algorithm is done through a fast locating technique combined with lookup tables, such that an input vector is efficiently assigned to a subspace where relatively few codeword searches are required to be executed. In terms of overall search performance, this work is experimentally validated as a superior search algorithm relative to a multiple triangular inequality elimination (MTIE), a TIE with dynamic and intersection mechanisms (DI-TIE), and an equal-average equal-variance equal-norm nearest neighbor search (EEENNS) approach. With a full search algorithm as a benchmark for overall search load comparison, this work provides an 87% search load reduction at a threshold of quantization accuracy of 0.96, a figure far beyond 55% in the MTIE, 76% in the EEENNS approach, and 83% in the DI-TIE approach.

Chapter 10 - Parameterization techniques for automatic speech recognition system

Linear predictive coding (LPC) parameters are widely used in various speech coding applications for representing the short-time spectral envelope information of speech [1]. For low bit rate speech coding applications, it is important to quantize these parameters using as few bits as possible. Considerable workhas been done in the past to develop both scalar and vector quantization procedures to quantize the LPC parameters [2, 3,4]. Scalar quantizers quantize each of the LPC parameters independently, while vector quantizers consider the entire set of LPC parameters as an entity and allow for direct minimization of quantization distortion. Because of this, the vector quantizers result in smaller distortion than the scalar quantizers at any given bit rate. The vector quantizers, however, have one major problem; their computational complexity is high. In our earlier paper [3], we have reported on a vector quantizer where the LPC parameter vector is split in the line spectral frequency (LSF) domain to overcome this complexity problem. We have shown that this quantizer can quantize the LPC parameters at 24 bits/frame with an average spectral distortion of 1 dB, less than 2% frames having spectral distortionl in the range 2-4 dB and no frame having spectral distortion greater than 4 dB.