The method of a 1/sup st/ order long-term predictor (LTP) filter, using a sub-sample resolution delay, is extended to a multi-tap LTP filter, or, equivalently, the conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. Defining the delay with sub-sample resolution enables this novel multi-tap LTP filter to explicitly model delay values that have a fractional component. The filter coefficients, largely freed from implicitly modeling the effect of delays that have a fractional component, seek to maximize the prediction gain of the LTP filter by modeling the frequency dependent gain. This is in contrast to a conventional multitap LTP filter, which applies a single model to tackle the dual tasks of representing the non-integer valued delays and the frequency dependent gain. Experimental results are presented for narrowband and wideband speech. This technique is part of the 3GPP2 source-controlled variable-rate multimode wideband speech codec (VMR-WB) rate set 1 standard.

Codebook searches in analysis-by-synthesis speech coders typically involve minimization of a perceptually weighted squared error signal. Minimization of the error over multiple codebooks is often done in a sequential manner, resulting in the choice of overall excitation parameters being sub-optimal. In this paper, we propose a joint excitation parameter optimization framework in which the associated complexity is slightly greater than the traditional sequential optimization, but with significant quality improvement. Moreover, the framework allows joint optimization to be easily incorporated into existing pulse codebook systems with little or no impact on the codebook search algorithms. This technique is part of the 3GPP2 source-controlled variable-rate multimode wideband speech codec (VMR-WB) rate set 1 standard.

The bit rate for the speech and channel coder for the half-rate GSM channel is 11400 bps. A 5600 bps VSELF speech coder designed for this application is described. This speech coder is one of two candidates being considered by GSM channel for the half-rate channel. It employs a novel strategy for vector quantization of the reflection coefficients (rj), which combines high coding efficiency, low codebook search complexity, and low storage requirements. A computationally streamlined version of a zero-pole spectral noise weighting function is implemented. An adaptive pitch prefilter and an adaptive spectral postfilter are used to improve the speech coder's performance for both the tandemed and non tandemed cases.

Techniques for improving the performance of CELP (code excited linear prediction)-type speech coders while maintaining reasonable computational complexity are explored. A harmonic noise weighting function, which enhances the perceptual quality of the processed speech, is introduced. The combination of harmonic noise weighting and subsample pitch lag resolution significantly improves the coder performance for voiced speech. Strategies for reducing the speech coder's data rate, while maintaining speech quality, are presented. These include a method for efficient encoding of the long-term predictor lags, utilization of multiple gain vector quantizers, and a multimode definition of the speech coder frame. A 5.9-kb/s VSELP speech coder that incorporates these features is described. Complexity reduction techniques which allow the coder to be implemented using a single fixed-point DSP (digital signal processor) are discussed.>

Techniques for improving the performance of CELP (code excited linear prediction) type speech coders while maintaining reasonable computational complexity are explored. A harmonic noise weighting function which enhances the perceptual quality of the processed speech is introduced. The combination of harmonic noise weighting and subsample resolution pitch significantly improves the coder performance for voiced speech. A 6.9 kb/s VSELP speech coder which incorporates subsample resolution pitch and harmonic noise weighting is described. Complexity reduction techniques are discussed which allow the coder to be implemented using a single fixed point digital signal processor.>

The vector sum excited linear prediction speech coder is presented. It utilizes a codebook with a structure that allows for a very efficient search procedure. Other advantages of the VSELP codebook structure are discussed, and a detailed description of an 8-kb/s VSELP coder is given. This coder was selected by the Telecommunications Industry Association (TIA) as the standard for use in North American digital cellular telephone systems. The coder uses two VSELP excitation codebooks, a gain quantizer which is robust to channel errors, and a novel adaptive pre/postfilter arrangement.>

A speech modification method presented by Griffin and Lim [1] involves modifying the Short-Time Fourier Transform Magnitude function of the original speech, and using it as a reference to iteratively approximate a modified waveform. This method has been successfully applied to time-scale modification and noise reduction of speech [1-3]. In this paper, we discuss our experiments in applying the algorithm to the independent modification of formant and pitch information.