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

In this paper, we present an improved quantization scheme for the redundancy data of a forward error correction (FEC) technique proposed for the transmission of code-excited linear prediction (CELP)-coded speech over erasure channels. The use of a FEC-based error protection scheme is motivated by the well-known fact that, after a frame erasure, the previous excitation is not available and a desynchronization between the encoder and the decoder long-term prediction (LTP) filters appears, causing an additional distortion which is propagated to subsequent frames. LTP synchronization can be recovered by means of a single-pulse representation of the previous excitation. No additional delay is introduced by this technique which only requires a small transmission bandwidth increase. In this paper, we focus on the efficient encoding of this pulse. Thus, an optimization procedure, which takes into account the overall synthesis error, is proposed in order to provide better pulse-position and pulse-amplitude quantization codebooks. Moreover, by extending the previous procedure, an efficient joint position-amplitude quantization can be obtained. Objective quality tests applied to our proposal show that, by means of the proposed codebooks, the number of bits required to represent the resynchronization pulse is effectively reduced. In addition, a discontinuous transmission mechanism is derived from the cost functional used during joint position-amplitude quantization, further reducing the bit-rate.

Code-excited-linear-predictive (CELP) coding has led to speech coding schemes with a good speech quality at 8 kb/s and a fair quality at data rates of about 4.8 kb/s. In the present work, the authors propose a novel method for improving the speech quality of a CELP codec at low bit rates by adaptation of the excitation codebook (ACELP). The adaptive cookbook consists of a set of basic excitation vectors which are adapted to an analysis speech frame by calculating optimum amplitude values. The amplitudes are quantized and encoded at a very low bit rate using a gain shape vector quantizer. The results of simulation experiments show that the mean squared error by CELP with an adaptive codebook is substantially lower than that of CELP with a fixed stochastic codebook. In informal listening tests the quality of the speech processed by the adaptive CELP scheme was superior and less speaker-dependent compared to a conventional CELP scheme. >

A 2.4-kbit/s analysis-by-synthesis speech codec based on a CELP (code-excited linear prediction) structure is presented. Several bit-rate reduction techniques, including the addition of a frame classifier and a frame-dependent codec structure, are used to improve the resulting speech quality. For voiced frames, the encoding is based entirely on the past excitation retrieved from an adaptive codebook using multitap gains with as many as seven taps. Unvoiced frames are encoded using only stochastic excitation. Finally, transition frames use both the adaptive and the stochastic codebooks. The results of subjective quality evaluation show that the open-loop classifier performs slightly better than the open-loop classifier, especially for female speakers. The best 2.4-kbit/s system uses the closed-loop classifier and a five-tap adaptive codebook for encoding voiced segments. This system achieve better subjective ratings than LPC-10e; however, it is still inferior to the Federal standard operating at double the bit rate, although for many speakers the difference was small. >

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.

The authors introduce two techniques for improving low-bit-rate CELP (code excited linear prediction) speech coders. The sparse-delta stochastic codebook is a recursive codebook design which can be searched using roughly 5% of the computational load required to search a full Gaussian codebook. Pitch sharpening is a class of algorithms which attempt to improve the perceptual quality of CELP by limiting the feedback of low-amplitude noiselike information to the adaptive codebook. Simulation results are presented for sparse-delta, ternary sparse-delta, and simplified-search sparse-delta coders, and for reduced-gain and sparse-adaptive-codebook pitch sharpening algorithms. >

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.

This paper presents a high-fidelity speech and audio codec operating at a sampling rate of 32 kHz and a bit rate of 64 kbit/s. Designed primarily for real-time speech communication systems with high port densities, this MDCT-based transform codec has a very low coding delay (8 ms frame size) and low codec complexity (less than 10 MIPS on a 16-bit fixed-point DSP). The codec achieves essentially transparent quality for speech, and very close to transparent quality for music. A novel frame erasure concealment algorithm makes this codec robust to frame erasures for both speech and music. Another novel feature allows the decoder to decode the bit stream directly into a 16 kHz or 8 kHz sampled signal, without the need to decode a 32 kHz signal first and then down-sample it to the target sampling rate. Other novel features include some speed-memory trade-off techniques to reduce the computational complexity.

Performance study of stochastic speech coders

The authors describe an implementation of fast algorithms for codebook searching used in code-excited linear predictive (CELP) speech coding. They present details of the algorithms including (i) fast conversion of line spectrum pair parameters to linear predictive coding parameters, and (ii) fast searches of the parameters of adaptive and stochastic codebooks. The description can be readily used for CELP implementations, such as on (i) high-quality low-bit-rate speech transmission in point-to-point or store-and-forward mode, and (ii) efficient speech storage in speech recording and multimedia databases. The algorithms have been implemented on an IBM-PC AT equipped with a TMS320C30 module and a SUN Sparcstation 10 for the development of packet radio speech transmission and voice mail applications, respectively.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

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.

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.

In this paper, we derived a relationship between Linear Predictive Coding (LPC) coefficients and Immittance Spectral Pairs (ISP) pseudo-cepstrum. In Adaptive Multi-Rate Wideband(AMR-WB) speech codec, ISP parameters were used for quantisation and interpolation. This paper first described the representation of ISP pseudo-cepstrum. ISP pseudo-cepstrum was defined as the inverse z-transform of half natural logarithm of immittance function. References [2] and [3] proposed two kinds of new perceptual weighting filters based on ISP pseudo-cepstrum. Spectral tilt is more pronounced in wideband signals due to the wide dynamic range between low and high frequencies. The proposed filters are helpful for solving this problem. Then we gave the conversion method between ISP pseudo-cepstrum and polynomials cepstrum coefficients. Finally we derived the recursive relations between ISP pseudo-cepstrum and LPC coefficients. There are two methods of converting LPC coefficients to ISP pseudo-cepstrum. The one is directly from LPC coefficients to ISP pseudo-cepstrum. The other is from ISP parameters.

This paper proposes an inter-frame correlation based error concealment approach for hybrid CELP (Code Excited Linear Prediction) and transform codec's with both good speech and audio quality at moderate bit rates. The proposed scheme is designed to overcome the main challenge due to the diversified characteristics of input signals. The underlying idea is to employ the inter-frame correlation of previous neighborhood frames to circumvent the pitfalls of referring to the unrelated frames, and to enable effective prediction of ISF (Immittance Spectral Frequencies) spectrum coefficients of missing frames from the immediate relative history using linear regression approach. Objective and subjective evaluation results for the proposed approach, in comparison with existing technique of AVS-P10 (Audio Video coding of China Standard Part 10 -- Mobile Speech and Audio Codec), provide strong evidence for gains across a variety of speech and audio signals.

While VoIP (voice over IP) is gaining importance in comparison with other types of telephony, packet loss remains as the main source of degradation in VoIP systems. Traditional speech codecs, such as those based on the CELP (code excited linear prediction) paradigm, can achieve low bit-rates at the cost of introducing interframe dependencies. As a result, the effect of a packet loss burst is propagated to the frames correctly received after the burst. iLBC (internet low bit-rate codec) alleviates this problem by removing the interframe dependencies at the cost of a higher bit-rate. In this paper we propose a combination of iLBC with an ACELP (algebraic CELP) codec in which a variable number of ACELP-coded frames is inserted between every two iLBC-coded frames. The experimental results show that the combined codec can achieve a performance close to that of iLBC at different loss conditions but with a smaller bit-rate. Also, scalability is achieved by modifying the number of inserted ACELP-coded frames.

In this study, the author considers the problem about economic process mean selection of product with the quality loss and process adjustment cost. Assume that the quality characteristic is normally distributed with process mean (μ) and variance (σ 2). The product quality loss is measured by adopting the asymmetric quadratic and line quality loss function. The adjustment cost of process mean is assumed to be proportional to μ 2 and the adjustment cost of process variance is assumed to be proportional to 1/σ 2. One can obtain the optimal process mean by minimizing the expected total cost of product including the product quality loss and process adjustment cost. Finally, the numerical example and sensitivity analysis of parameters are provided for illustration.