Error protection for a 4.8 kbps VQ based CELP coder

A testbench which enables a software implementation of different code-excited linear predictive (CELP) coder configurations in a simple and modular way, is presented as a research and development tool. It permits a flexible simulation process while using the actual hardware and software resources which constitute a real system. Its behaviour is consequently close to that of the final system envisaged. This testbench has been developed using the TMS320C30 digital signal processor and has been programmed in a modular way in C language. A CELP coder using two different codeword generation methods (the LBG algorithm and the Fuzzy-LVQ partitioning algorithm) have been tested using the proposed testbench.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

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.

In the last few years, Code-Excited Linear Predictive (CELP) coding has emerged as the most prominent technique for digital speech communication at rates of 8 Kb/s and below, and it is now considered the best candidate coder for digital mobile telephony and secure speech communication. While the CELP coder is able to provide fairly good-quality speech at 8 Kb/s, its performance at 4.8 Kb/s is yet unsatisfactory for many applications. The novelty in the CELP coding concept, namely, the stochastic excitation of a linear filter, also constitutes a weakness of this method: the excitation contains a noisy component which does not contribute to the speech synthesis process and can not be completely removed by the filter. It is a common opinion among speech communication researchers that new forms of excitations need to be studied in order to improve the CELP performance at low bit rates.

Although the speech quality of the code excited linear prediction (CELP) coder at 4800bit/s is relatively good, it is still perceived as rough or noisy. In the letter we propose a residual shaping method that produces, without signal distortion, quality comparable to that obtained by adaptive postfiltering. The proposed method is particularly effective in the multiple tandeming environment.

By reconciling differences between the estimator and the filter of a code excited linear predictive (CELP) voice coder, higher quality is achieved in the output speech. The pulse amplitudes and pitch tap gain are solved for simultaneously to minimize the estimator bias in the CELP excitation. Increased signal to noise ratio is accomplished by modifying the pitch predictor such that the pitch synthesis filter accurately reflects the estimation procedure used to find the pitch tap gain, and by improving the excitation analysis technique such that the pitch predictor tap gain and codeword gain are solved for simultaneously, rather than sequentially. These modifications do not result in an increased transmission rate or significant increase in complexity of the CELP coding algorithm.

We present a study of a channel optimized matrix quantizer (COMQ) applied to quantize LSP (line spectral pair) parameters in a CELP (code excited linear prediction) coder. A modification of the DoD FS-1016 standard (Campbell et al., 1989) is used for this purpose. A Gaussian channel is considered as the channel through which information is sent and BPSK (binary phase shift-keying) is used as modulation technique. To measure the performance of this coder several channel signal to noise ratios are considered. Also, the performance of the FS-1016 standard coder is computed for comparison purposes.

Remote retrieval of compressed voicemail data over a telephone line is one of several emerging applications of speech coding. Using a notebook computer equipped with a modem, a user can remotely access a networked desktop unit located at their office or home to retrieve various types of information such as email, FAX, electronic documents as well as voicemail. By compressing the speech data, we reduce the amount of time needed for the transfer of voice data over the telephone line. We have developed a high-quality variable-rate CELP (code-excited linear prediction) coder which can be used in such an application. The coder operates at an average rate of 3 kb/s assuming 80% speech activity and uses several new techniques including a mode decision mechanism based on the use of a peakiness measure and a novel excitation search method called gain-matched analysis-by-synthesis. The coder gives quality comparable to the Microsoft GSM 6.10 Audio Codec at 13 kb/s.

Electronic speech coding had its origin in 1928 when Homer Dudley proposed to send speech signals over a new transatlantic telegraph cable with the (then) enormous bandwidth of 100 Hz. Dudley argued that speech was generated by slowly moving articulators and should therefore require only some 100-Hz total bandwidth for transmission. Since extraction of these parameters proved difficult, Dudley suggested, as an alternative, the transmission of the likewise slowly changing spectral information. The result was the channel vocoder, which (together with formant and voice-excited vocoders) dominated speech coding and that of musical sounds of single-pitch instruments until the mid 1960s. Unhappy with the vocoder's “electronic accent.” Bishnu Atal and the author, in 1967, advanced the idea of linear predictive coding (LPC), motivated by the desire to replace rigid vocoderlike coding by a more flexible method in which a “prediction residual” could represent deviations of real speech signals from an assumed all-pole model. In 1972, subjective error criteria were introduced, based on auditory models of masking and the possibility of “hiding” quantizing noise under the speech signal. The resulting code-excited linear predictive (CELP) coders based on statistical models have been operated at 1/4 bit per sample for the residual at virtually no audible degradation. With powerful digital signal processing chips, CELP has now become the method of choice for synthetic-voice applications. For musical sounds, masking models and linear predictive coding, which is equivalent to prewhitening of the spectrum, lead to a substantial reduction in the required number of bits per sample.

The behavior of several windows in relation to time delay and spectral estimation is analyzed in its application to CELP (code excited linear prediction) coders, with emphasis on time delay considerations. From the data, the importance of asymmetric windows in such a context becomes evident. In order to obtain objective conclusions, a perceptually weighted residual energy was used as the comparison criterion. Experimental data show that an appropriately chosen window can reduce the time delay by up to 50% of the analysis frame. Typical reductions in delay will be about 6-8 ms. It is observed that this delay reduction is obtained with no cost in complexity or speech quality, and that in many algorithms it can be implemented in the coder without affecting compatibility with previous versions of the same algorithm. As an example of a specific implementation, an asymmetric window has been introduced in the Federal Standard 1016 CELP algorithm (4.8 kbit/s CELP). It has been shown to reduce the implicit delay of that algorithm by 5.6 ms with no loss in quality. >

The research reported herein is concerned with the development of high quality speech coding algorithms at bit rates between 4 and 8 kb/s. In the literature, analysis by synthesis coding based on a weighted mean squared error distortion criteria has been demonstrated to work well at rates as low as 4-16 kb/s. In this research, improved search techniques coupled with improved codebook design methodologies are proposed to improve the performance of conventional code-excited linear predictive coders. Improved methods for quantizing the short term filter are developed by applying a tree search algorithm and joint codebook design to multi-stage vector quantization. Joint codebook design procedures are developed to design locally optimal multi-stage codebooks. Weighting during centroid computation is introduced in order to improve the outlier performance of the multi-stage vector quantizer. Multi-stage vector quantization is shown to be both robust against input characteristics and in the presence of channel errors. Spectral distortions of approximately 1 dB are obtained at rates between 22 and 28 bits/frame for various multi-stage configurations. Structured codebook design procedures for the excitation in code-excited linear predictive coders are compared to general codebook design procedures. It is determined that little is lost in using significant structure in the excitation codebooks, while greatly reducing the search complexity. Sparse multi-stage configurations are proposed as a means of reducing computational complexity and memory size while offering both a variable rate and variable complexity. Improved search procedures applied to code-excited linear prediction which attempt to jointly optimize the short term filter, the adaptive codebook, and the excitation are proposed. Although the computational complexity is rather large, segmental signal to noise ratio improvements between 1 and 2 dB are realized in practice.

This letter proposes a 4-kb/s multimode code-excited linear prediction (CELP) coder with pitch synchronous extended excitation. Three modes are used for the short-term excitation, namely algebraic, extended, or stochastic excitations, together with an adaptive codebook for the long-term excitation. Comparisons with the FS-1016 and ITU-T G.723.1 coders show a performance level between these standards.

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

A systematic study of code excited linear predictive (CELP) coders is presented. The design of an optimized version complying with the specifications for a GSM half-rate coder is described. This study is divided into three parts: election of an unquantized configuration, assuming unquantized scale factors and filter coefficients, fine tuning of the parameter quantization, and adaptation to the GSM specifications. A strong emphasis is placed on efficient codebook search procedures, and two approaches are briefly described: the truncated autocorrelation approach and the unity-magnitude approach. The final coder version achieves a very good combination of speech quality and implementation complexity.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Code-excited linear prediction (CELP) is one of the most commonly used approaches for speech compression. The CELP coder, in particular the ACELP coder, relies on extracting an appropriate excitation sequence that is passed through a long-term and short-term linear prediction filter to synthesize the speech. One limitation of the analysis-by-synthesis search methods employed in the ACELP coder is that the positions of the non-zero entries in the excitation sequence and their gains are limited. Increasing the richness of the excitation to improve the speech quality is not only accompanied by the usual increase in the bit rate but also by a significant increase in search complexity. We propose dealing with this scalability issue by using tools from the compressed sensing (CS) domain. An analysis by synthesis coder using a CS based excitation sequence framework is developed and the encoder is shown to scale gracefully with CS dimension and offers a mechanism for quality to bit rate trade off.

A speech coding and decoding system, the system is operated under a known code-excited linear prediction (CELP) coding method. The CELP coding is achieved by selecting an optimum pitch vector P from an adaptive codebook and the corresponding first gain, and at the same time, selecting an optimum code vector from a stochastic codebook and the corresponding second gain. The system of the present invention is featured by a weighted orthogonalization transforming unit introduced therein. The perceptually weighted code vector AC is not used as is, as usual, but after the transformation thereof into a perceptually weighted code vector AC' by the above unit; the vector AC' being made orthogonal to the optimum perceptually weighted pitch vector AP.