James P. Ashley

Low Complexity Factorial Pulse Coding of MDCT Coefficients using Approximation of Combinatorial Functions

Factorial pulse coding, a method which is known to efficiently code an information signal using unit magnitude pulses, involves computation of combinatorial functions. These computations are highly complex as they require many multiply and divide operations on multi-precision numbers, especially when the length of a signal is large or many unit magnitude pulses are used for coding. In this paper, we propose a very low complexity method for approximation of these combinatorial functions. The approximate functions satisfy a property which preserves unique decode-ability of the factorial packing encoding/decoding algorithm. The low complexity computation enables use of factorial packing in encoding/decoding of 144 MDCT coefficients using 28 unit magnitude pulses for the audio coding mode of the EVRC-WB speech coding standard without affecting the number of bits required for coding.

Background noise suppression for speech enhancement and coding

A background noise suppression system developed by Motorola is included as a feature in IS-127, the TIA/EIA standard for the enhanced variable rate codec (EVRC) to be used in CDMA based telephone systems. We describe the algorithm used in this system and its implementation. We then present subjective listening test results showing the advantages of using such a system as a prefilter to a speech coder.

Closed Loop Dynamic Bit Allocation for Excitation Parameters in Analysis-by-Synthesis Speech Codec

A method for dynamically allocating bits for the adaptive and fixed codebook of an analysis-by-synthesis speech codec is proposed. The bit allocation is based on the closed loop weighted mean squared error. The different bit allocations identify various codebook configurations used by the adaptive and fixed codebooks of the codec. Unlike open loop approaches where the decision on a codebook configuration is made once per frame, in the closed loop approach the decision on which codebook configuration should be used is made in each subframe. A variable length code is used for coding the codebook configuration. The factorial packing codebook is used as the fixed codebook. The technique is a part of 8.5 kbps mode of EVRC-WB speech coding standard.

Joint optimization of excitation parameters in analysis-by-synthesis speech coders having multi-tap long term predictor

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.

Structural issues in cascade-form adaptive IIR filters

Adaptive IIR filters implemented in cascade-form are attractive due to the ease with which their stability may be monitored. Four cascade-form structures are compared for use in adaptive filtering with respect to complexity of implementation, error surface geometry, and adaptation speed. The four structures include a cascade of second order pole/zero sections, a cascade of second order all-pole sections followed by a tapped delay line, and two new structures. The latter pair includes a tapped cascade, which is a cascade of second order all-pole sections whose output is constructed as a weighted combination of signals tapped from the cascade. The second new structure is a modification of the tapped cascade that yields orthogonal signals at the taps of the cascade. It is shown that the tapped cascade provides the best overall performance in the respects noted above.

Coding pulse sequences using a combination of factorial pulse coding and arithmetic coding

A Factorial Pulse Coding approach using combinatorial functions for coding of pulse sequences is presented. An arithmetic coding approach for coding of pulse sequences is also described. Factorial pulse coding and arithmetic coding are compared. A method of combining the two approaches is proposed. The proposed combining method works by extending the pulse sequence by one bit whose probability is found from the arithmetic coding bounds.

A technique of multi-tap long term predictor (LTP) filter using sub-sample resolution delay [speech coding applications]

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.