Code Excited Linear Prediction Research Articles

Method and apparatus for speech encoding and decoding by sinusoidal analysis and waveform encoding with phase reproducibility

A speech encoding method and apparatus in which an input speech signal is divided in terms of blocks or frames as encoding units and encoded in terms of the encoding units, whereby explosive and fricative consonants can be impeccably reproduced, while there is an attenuation of the occurrence of foreign sounds being generated at a transient portion between voiced (V) and unvoiced (UV) portions, so that the speech with high clarity devoid of “stuffed” feeling may be produced. The encoding apparatus includes a first encoding unit for finding residuals of linear predictive coding (LPC) of an input speech signal for performing harmonic coding and a second encoding unit for encoding the input speech signal by waveform coding. The first encoding unit and the second encoding unit are used for encoding a voiced (V) portion and an unvoiced (UV) portion of the input signal, respectively. Code excited linear prediction (CELP) encoding employing vector quantization by a closed loop search of an optimum vector using an analysis-by-synthesis method is used for the second encoding unit. A corresponding decoding method and apparatus is also provided.

Bit rate reduction of mixed excitation linear prediction coder by Lempel-Ziv segment quantization

Bit rate reduction of mixed excitation linear prediction coder by Lempel-Ziv segment quantization

Encoding and decoding method and apparatus using rising-transition detection and notification

A decoding apparatus is provided. The decoding apparatus has a first decoding part for decoding a code word obtained by encoding an input signal using a Code-Excited Linear Prediction encoding method. A second decoding part decodes a code word obtained by encoding a signal with an encoding method other than the Code-Excited Linear Prediction encoding method. A rising-transition detection and notification part has a detection part that detects the existence of a rising-transition of amplitude of the input signal based on time variation of a gain of excitation vectors obtained by the first decoding part, and a notification part that notifies the second decoding part that the rising-transition of the amplitude exists.

Switching Search Method for Pulse Assignment in ITU-T G.729D

In this paper, the simplified search designs for the stochastic codebook of algebraic code excited linear prediction (ACELP) for ITU-T G.729D speech coder are proposed. By using two search rounds and limiting the search range, the computational complexity of the proposed approach is only 6.25% of the full search method recommended by G.729D. In addition, the computational complexity of proposed approach is only 59% of the global pulse replacement search method recommended by G.729.1. Simulation results show that the coded speech quality evaluated by using the standard subjective and objective quality measurements is with perceptually negligible degradation.

Decoder Initializing Technique for Improving Frame-Erasure Resilience of a CELP Speech Codec

The authors present and evaluate a technique for synchronizing the internal states of a code-excited-linear-prediction (CELP) encoder and decoder after the occurrence of frame erasure. The designed technique, called ldquoduplicated transmission (DT),rdquo uses some redundant information for realizing synchronization. The encoder performs encoding processes twice and sends two codes for each frame. One code is encoded by an encoder that is initialized. The code is used in cases where the previous frame is erased. An onset detector is combined with the DT technique to select the frames to which the DT should be applied. Subjective test results suggest that, by introducing DT selectively, the number of DT frames is reducible by about 80% without degrading the subjective quality. Results demonstrate that synchronization of the internal states is effective in cases of erasure of onset. The DT technique requires no additional algorithmic delay. For that reason, it would a better choice for particular applications for which the delay has a significant impact.

Codebook Based Digital Speech Compression

Speech compression has been an area of interest for many researchers, and their objectives remained to reduce the bit rates, also to maintain the speech quality. Adaptive Differential Pulse Code Modulation (ADPCM) was first such scheme that evolved out of PCM and in low bit rate source coding techniques; Code Excited Linear Prediction (CELP) was adopted as standard in the Department of Defense (DoD), USA. In this research a novel approach is developed and compression up to 50% is achieved with negligible loss of quality. This coding technique is developed to use a PCM coded speech, initially, and then substituting 3 to 5 Least Significant Bits (LSB) of every byte at the decoding end from a fixed codebook, randomly.

Fast Recovery for a CELP-Like Speech Codec After a Frame Erasure

The adaptive codebook used in code-excited linear prediction (CELP)-like speech codecs is very effective for modeling the quasi-periodic component of the excitation signal but, unfortunately, introduces a strong interframe dependency that renders the decoder vulnerable to frame erasures. For voiced speech, the error affects not only the erased frame but also all the subsequent frames. In this paper, a technique to improve the recovery after a frame erasure is proposed. The technique consists in a constrained excitation search at the encoder and a resynchronization procedure at the decoder. The constraint aims at reducing the contribution of the adaptive codebook by making the innovation codebook partially model the pitch excitation. Further, for highly voiced frames, the pitch-related information contained in the innovation excitation is exploited at the decoder to speed up the resynchronization of the adaptive codebook after a frame erasure. When applied to the adaptive multirate wideband (AMR-WB) codec, the method brings a significant improvement in the case of frame erasures, at the cost of a minor quality loss compared to the standard codec at the same bit rate. The method does not need additional delay and has the advantage of maintaining full interoperability between the standard codec and its modified version.

Soft Reconstruction of Speech in the Presence of Noise and Packet Loss

Exploiting the residual redundancy in a source coder output stream during the decoding process has been proven to be a bandwidth efficient way to combat the noisy channel degradations. In this paper, we consider soft reconstruction of speech spectrum, in GSM adaptive multirate and IS-641 vocoders, transmitted over a channel disturbed with noise and/or packet loss. Several schemes are presented which exploit different levels of intraframe and interframe residual redundancy for improved source decoding at the receiver. A packetization strategy is proposed which is matched to the presented error concealment units. For decoders that exploit the residual redundancy, extensive complexity has been a serious concern, especially as the quantizer bitrate increases . In this paper, a novel method is presented to construct reduced complexity algorithms. The proposed methodology is based on the classification of the signal domain and efficient approximation of the residual redundancy or the a priori transition probabilities. The presented schemes provide high quality error concealment solutions for code excited linear prediction (CELP) coders

Compression of surface EMG signals with algebraic code excited linear prediction

Despite the interest in long timescale recordings of surface electromyographic (EMG) signals, only a few studies have focused on EMG compression. In this paper we investigate a lossy coding technique for surface EMG signals that is based on the algebraic code excited linear prediction (ACELP) paradigm, widely used for speech signal coding. The algorithm was adapted to the EMG characteristics and tested on both simulated and experimental signals. The coding parameters selected led to a compression ratio of 87.3%. For simulated signals, the mean square error in signal reconstruction and the percentage error in average rectified value after compression were 11.2% and 4.90%, respectively. For experimental signals, they were 6.74% and 3.11%. The mean power spectral frequency and third-order power spectral moment were estimated with relative errors smaller than 1.23% and 8.50% for simulated signals, and 3.74% and 5.95% for experimental signals. It was concluded that the proposed coding scheme could be effectively used for high rate and low distortion compression of surface EMG signals. Moreover, the method is characterized by moderate complexity (approximately 20 million instructions/s) and an algorithmic delay smaller than 160 samples (∼160 ms).

Efficient algebraic code-excited linear predictive codebook search

To reduce the computational complexity of algebraic code-excited linear prediction (ACELP) coders, an efficient codebook search mechanism based on a simplified correlation matrix (SCM) of the vocal impulse response is proposed. In the proposed approach, the statistical characteristics of the vocal impulse response are identified such that only a small proportion of the total number of correlation coefficients in the correlation matrix need be calculated before the ACELP search procedure is carried out. Furthermore, the proposed joint scheme, by combining the SCM method and a pulse position prediction scheme, not only decreases the arithmetic complexity in the pre-computing autocorrelation matrix but also reduces the number of pulse position combinations. The simulation and experimental results show that the proposed method provides an effective reduction in the computational load of the ACELP codebook search procedure with no discernible degradation of the speech quality.

Multiband Vector Quantization Based on Inner Product for Wideband Speech Coding

This paper describes a multiband vector quantization (VQ) technique based on inner product for wideband speech coding at 16 kb/s. Our approach consists of splitting the input speech into two separate bands and then applying an independent coding scheme for each band. A code excited linear prediction (CELP) coder is used in the lower band while a transform based coding strategy is applied in the higher band. The spectral components in the higher frequency band are represented by a set of modulated lapped transform (MLT) coefficients. The higher frequency band is divided into three subbands, and the MLT coefficients construct a vector for each subband. Specifically, for the VQ of these vectors, an inner product-based distance measure is proposed as a new strategy. The proposed 16 kb/s coder with the inner-product based distortion measure achieves better performance than the 48 kb/s ITU-T G.722 in subjective quality tests.

Signal modification method for variable bit rate wide-band speech coding

This paper introduces a novel signal modification method for wide-band code-excited linear prediction (CELP) speech codecs to improve pitch prediction at low bit rates. The method is enabled only in stable voiced speech frames, and preserves the original time scale at the end of each frame. This feature helps to avoid artifacts and simplifies an encoder implementation. The signal modification includes a classification algorithm as an integral part. The classification algorithm detects the frames most suitable for signal modification and low bit rate coding, and can be employed in a rate selection module of variable bit rate (VBR) codecs. In this paper, the signal modification method is applied in an experimental VBR wide-band speech codec derived from the 3GPP adaptive multirate wideband (AMR-WB) standard (ITU-T Recommendation G.722.2). The codec fulfills the system requirements of IS-95/CDMA2000 Rate Set II, operating at source coding bit rates 12.65, 6.2, and 1.0 kb/s. The signal modification is used in the 6.2 kb/s mode dedicated for voiced speech frames. Listening test results demonstrate the good performance of the proposed method. The signal modification method is used in the Nokia/VoiceAge codec that was declared in April 2003 as the winner of the selection phase in the 3GPP2 CDMA2000 wide-band speech codec standardization.

Optimized trellis coded vector quantization of LSF parameters, application to the 4.8 kbps FS1016 speech coder

Speech coders operating at low bit rates necessitate efficient encoding of the linear predictive coding (LPC) coefficients. Line spectral frequencies (LSF) parameters are currently one of the most efficient choices of transmission parameters for the LPC coefficients. In this paper, an optimized trellis coded vector quantization (TCVQ) scheme for encoding the LSF parameters is developed. When the selection of a proper distortion measure is the most important issue in the design and operation of the encoder, an appropriate weighted distance measure has been used during the TCVQ construction process. Using this distance, we will show that the LSF TCVQ encoder performs better than the encoder conceived with the unweighted distance and a reduction of about 1–2 bits/frame is obtained while maintaining the same performance. We further applied the TCVQ encoder system for encoding the LSF parameters of the US federal standard (FS1016) 4.8 kbps code excited linear prediction (CELP) speech coder. At lower bit rates, our objective and subjective evaluation results show that the incorporated LSF TCVQ encoder performs better than the 34 bits/frame LSF scalar quantizer used originally in the FS1016 coder. The subjective tests reveal also that the 27 bits/frame scheme produces equivalent perceptual quality to that when the LSF parameters are unquantized.

Voice Activity Detection Algorithm Based on Radial Basis Function Network

This paper proposes a Voice Activity Detection (VAD) algorithm using Radial Basis Function (RBF) network. The k-means clustering and Least Mean Square (LMS) algorithm are used to update the RBF network to the underlying speech condition. The inputs for RBF are the three parameters a Code Excited Linear Prediction (CELP) coder, which works stably under various background noise levels. Adaptive hangover threshold applies in RBF-VAD for reducing error, because threshold value has trade off effect in VAD decision. The experimental results show that the proposed VAD algorithm achieves better performance than G.729 Annex B at any noise level.

Watermarking Combined with CELP Speech Coding for Authentication

This letter presents a speech watermarking scheme that is combined with CELP (Code Excited Linear Prediction) speech coding for speech authentication. The excitation codebook of CELP is partitioned into three parts and labeled '0', '1' and 'any' according to the private key. Watermark embedding process chooses the codebook whose label is the same as the watermark bit and combines it with the codebook labeled 'any' for CELP coding. A statistical method is employed to detect the watermark, and the watermark length for authentication and detection threshold are determined by false alarm probability and missed detection probability. The new codebook partition technique produces less distortion, and the statistical detection method guarantees that the error probability can be controlledunder prescribed level.

Efficient two-stage vector quantization speech coder using wavelet coefficients of excitation signals

An improved backward prediction coder featuring two-stage vector quantization (VQ) of shape codevectors is presented. Efficient two-stage VQ is achieved using the wavelet coefficients of excitation signals; i.e., wavelet coefficients are calculated by applying a discrete wavelet transform to excitation signals, and the results are divided into an approximation group and a detail group. The data lengths of both approximation and detail coefficients are half that of conventional two-stage VQ systems. Simulation results show that the proposed coder achieves a better weighted signal-to-noise ratio (WSNR) than conventional coders and, in terms of reconstructed speech quality, ranks between the FS-1016 Code Excited Linear Prediction (CELP) coder and the Vector Sum Excited Linear Predictive Coding (VSELP) coder.

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.

A Noise Reduction Preprocessor for Mobile Voice Communication

We describe a speech enhancement algorithm which leads to significant quality and intelligibility improvements when used as a preprocessor to a low bit rate speech coder. This algorithm was developed in conjunction with the mixed excitation linear prediction (MELP) coder which, by itself, is highly susceptible to environmental noise. The paper presents novel as well as known speech and noise estimation techniques and combines them into a highly effective speech enhancement system. The algorithm is based on short-time spectral amplitude estimation, soft-decision gain modification, tracking of the a priori probability of speech absence, and minimum statistics noise power estimation. Special emphasis is placed on enhancing the performance of the preprocessor in nonstationary noise environments.

KLT-Based Adaptive Classified VQ of the Speech Signal

Compared to scalar quantization (SQ), vector quantization (VQ) has memory, space-filling, and shape advantages. If the signal statistics are known, direct vector quantization (DVQ) according to these statistics provides the highest coding efficiency, but requires unmanageable storage requirements if the statistics are time varying. In code-excited linear predictive (CELP) coding, a single compromise codebook is trained in the excitation-domain and the space-filling and shape advantages of VQ are utilized in a nonoptimal, average sense. In this paper, we propose Karhunen-Loe/spl grave/ve transform (KLT)-based adaptive classified VQ (CVQ), where the space-filling advantage can be utilized since the Voronoi-region shape is not affected by the KLT. The memory and shape advantages can be also used, since each codebook is designed based on a narrow class of KLT-domain statistics. We further improve basic KLT-CVQ with companding. The companding utilizes the shape advantage of VQ more efficiently. Our experiments show that KLT-CVQ provides a higher SNR than basic CELP coding, and has a computational complexity similar to DVQ and much lower than CELP. With companding, even single-class KLT-CVQ outperforms CELP, both in terms of SNR and codebook search complexity.

The 2003 Benjamin Franklin medal in electrical engineering presented to Bishnu S. Atal

Dr. Atal was chosen for his fundamental contributions to speech coding in digital cellular telephony—specifically for pioneering work in linear predictive coding (LPC) and use of analysis-by-synthesis leading to multi-pulse excited linear predictive coding and code excited linear predictive coding (CELP). These are the basis of time-division and code-division cellular systems (TDMA & CDMA), the Nextel system, and the global system for mobile communications (GSM).