Mixed Excitation Linear Prediction Research Articles

Channel and source considerations of a bit-rate reduction technique for a possible wireless communications system's performance enhancement

In wireless commercial and military communications systems, where bandwidth is at a premium, robust low-bit-rate speech coders are essential. They operate at fix bit rates and those bit rates cannot be altered without major modifications in the vocoder design. A novel approach to vocoders, in order to reduce the bit rate required to transmit speech signal, is proposed. While traditional low-bit-rate vocoders code original input speech, the proposed procedure operates on the time-scale modified signal. The proposed method offers any bit rate from 2400 b/s to downwards without modifying the principle vocoder structure, which is the new NATO standard, Stanag 4591, Mixed Excitation Linear Prediction (MELP) vocoder. We consider the application of transmitting MELP-encoded speech over noisy communication channels by applying different modulation techniques, after time-scale compression is applied. Three different time-scale modification algorithms have been evaluated and waveform similarity overlap and add (WSOLA) algorithm has been selected for time-scale modification purposes. Computer simulation results, both source and channel, are presented in terms of objective speech quality metrics and informal subjective listening tests. Design parameters such as codec complexity and delay are also investigated. Simulation results lead to a possible wireless communications system, whose performance might be enhanced by using the spared bits offered by the procedure.

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.

A low bit rate speech codec using mixed excitation linear prediction for private mobile radio

AbstractThe authors describe a low bit rate speech codec developed for the next generation of narrow‐band (3 kHz/voice channel) private mobile radio. The speech encoding speed under the proposed speech codec is 1.6 kbit/s, and the speed after communications path encoding is 3.2 kbit/s. The authors propose a method using the speech spectrum envelope information to determine the mixing ratio for pulses and noise in each frequency band used to generate the mixed excitation signal in the Mixed Excitation Linear Prediction (MELP) vocodor model used as a speech encoding method. For the communications path encoding method, the authors assume a burst error of 7% for the worst communications environments in private mobile radio. In order to improve resistance to transmission path errors, the authors create soft decision demodulation data while varying the quantization step width until the errors are no longer detected or a designated number of repetitions is reached for the frame determined as having errors based on the results of error detection during communications path decoding. They then propose a method to repeat soft decision Viterbi decoding. The authors perform subjective evaluation experiments, and confirm the effectiveness of the proposed speech codec in a private mobile communications environment. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 2, 87(6): 69–81, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjb.20096

An intelligibility enhancement for the mixed excitation linear prediction speech coder

We recently proposed a technique to increase the robustness of perceptually important acoustic cues in speech, based on modification of the phoneme timing structure. We now apply the technique as a preprocessor to improve the intelligibility of the mixed excitation linear prediction (MELP) coder. The enhancement strategy, however, requires adaptations to minimize the introduction of unnatural speech characteristics that are introduced and intensified by parametric coding schemes. The enhanced 2.4 kb/s fixed-point MELP coder achieves a level of speech intelligibility comparable to the 8 kb/s G.729 Annex A coder.

Hybrid symbol- and bit-based source-controlled channel decoding for MELP speech parameters

A joint source channel coding (JSCC) scheme which exploits bit-level correlation as well as symbol-level correlation efficiently in a source-controlled channel decoding (SCCD) process is proposed and applied to the mixed-excitation linear prediction (MELP) parameters of speech. A modified BCJR algorithm is also proposed for use in the SCCD algorithm. Simulation results show that our proposed scheme performs better than other redundancy-based JSCC schemes such as bit-based SCCD, soft-bit speech decoding (SBSD) and iterative source-channel decoding.

Robust transmission of melp-compressed speech: an illustrative example of joint source-channel decoding

A mixed-excitation linear predictive (MELP) speech coder was selected as the US federal standard for 2400 b/s speech compression. This paper examines the quality of MELP-compressed speech when transmitted over noisy communication channels in conjunction with a variety of error-control schemes. The focus is on channel decoders that exploit the inherent in the MELP bitstream. This residual redundancy, which is manifested by the correlation in time and the nonuniform distribution of various MELP parameters, can be quantified by modeling the parameters as one-step Markov chains and computing the entropy rate of the Markov chains based on the relative frequencies of transitions. Moreover, this residual redundancy can be exploited by an appropriately tuned channel decoder to provide substantial coding gain when compared with decoders that do not exploit it. Channel coding schemes include conventional binary convolutional codes and iteratively-decoded parallel concatenated convolutional (turbo) codes.

Channel and source considerations of a MELP derived vocoder operating at reduced bit rates

Robust low bit rate speech coders are essential in commercial and military communication systems. They operate at fixed bit rates and those bit rates cannot be altered without major modifications in the vocoder design. In this paper we introduce a scaled speech coder, which operates on time-scale modified input speech. The proposed method offers any bit rate from 2400 b/s to downwards without modifying the principle vocoder structure, which is the mixed excitation linear prediction (MELP) vocoder. We consider the application of transmitting MELP-encoded speech over noisy communication channels after time scale compression is applied. Computer simulation results, both source and channel, are presented in terms of objective speech quality metrics and informal subjective listening tests. A statistical tool called bootstrap is also used to determine the accuracy of these test results. Design parameters such as codec complexity and delay are also investigated.

Simplification of soft-bit speech decoding and application to MELP encoded speech

A simplified soft-bit speech decoding (SBSD) algorithm is proposed and is used in decoding the encoded speech parameters from the mixed excitation linear prediction (MELP) coder. Simulation results show that the simplified algorithm can reduce significantly the decoding complexity and memory requirement of SBSD at a slight penalty of a marginal drop in performance.

Reconstruction of speech from whispers

This paper investigates a method for the real-time reconstruction of normal speech from whispers. This system could be used by aphonic individuals as a voice prosthesis. It could also provide improved verbal communication when normal speech is not appropriate. The normal speech is synthesized using the mixed excitation linear prediction model. Differences between whispered and phonated speech are discussed and methods for estimating the parameters of this model from whispered speech for real-time synthesis are proposed. This includes smoothing the noisy linear prediction spectra, modifying the formants, and synthesizing of the excitation signal. Trade-offs between computational complexity, delay, and accuracy of different methods are discussed.

A flexible bit rate switching method for low bit rate vocoders

Robust low bit rate speech coders are essential in commercial and military communication systems. They operate at fixed bit rates and bit rates cannot be altered without major modifications in the vocoder design. In this paper we introduce a novel approach to vocoders by coding the time-scale modified input speech signal. The proposed method offers any bit rate from 2400 to 720 bits/ s without modifying the vocoder structure. Simulation results, which mainly concentrate on intelligibility, talker recognisability and voice quality versus codec complexity and delay, are also presented. The proposed scaled speech coder delivers communication quality speech at half the bit rate of the new US Federal Standard, mixed excitation linear prediction vocoder.

An 8.0-/8.4-kbps wideband speech coder based on mixed excitation linear prediction

A new method based on the Mixed Excitation Linear Prediction (MELP) model for coding wideband speech signals at medium bit rates is proposed. Modifications are made to the voiced/unvoiced strength analysis, pitch estimation and linear predictive (LP) analysis/synthesis of the Federal MELP Standard. An 8.0-/8.4-kbps wideband MELP coder is developed based on the modifications. Both the objective and subjective tests show that the subjective quality of the synthesized clean speech coded by the proposed 8.4-kbps wideband MELP coder is comparable to that of the ITU G.722 (P. Mermelstein, G. 722: a new CCITT coding standard for digital transmission of wideband audio signals, IEEE Comm. Mag. 26 (1) (1988) 8–15.) coder coding at 48 kbps. For the operating bit rate of 8 kbps, the performance of the coder is slightly better than that of MPEG4 CELP (Code-Excited Linear Prediction) (J.D. Johnson, S.R. Quackenbush, J. Herre, B. Grill, Review of MPEG-4 general audio coding, in: Multimedia Systems, Standards and Networks, A. Puri, T. Chen (Eds.), Marcel Dekker, New York, 2000 (Chapter 5)) coder operating at 14.4 kbps.

Joint MELP turbo code with unequal error protection

The mixed excitation linear prediction (MELP) algorithm has been recently selected as the new federal standard for 2.4 kbit/s coding of speech signals. The authors exploit the average residual inter-frame correlation and the error sensitivities of the bits in a MELP frame to enhance the robustness of the proposed joint MELP turbo coding schemes for operations over Rayleigh fading channels.