A Comparison of Digital Speech Coding Methods for Mobile Radio Systems

Subjective quality measurements on three digital speech coders, simulated with mobile radio channel transmission, were performed using the "mean opinion score (MOS)" method. The three speech coding methods tested were: continuously variable slope deltamodulation (CVSD) coding, adaptive predictive coding (APC), and residually excited linear predictive (RELP) coding. Several versions of each coder, with transmission rates in the range of 7.3 to 16.1 kbits/s, were simulated. Five different channel conditions, including three derived from land mobile radio field experiments, were applied to the speech coders' encoded output to study the effects. The results show that of the three coders, the CVSD coder is the most robust to channel errors, but produces reconstructed output speech of unacceptable quality. The 14.4 kbit/s RELP coder produces relatively good Output speech quality, exhibits a mild degree of robustness to mobile radio channel errors, and is slightly less complex than the APC coder. Of the three digital speech coders tested, the RELP coder appears the most suitable for use with land mobile radio. However none of the three coders was able to produce speech of telephone toll quality in a mobile radio environment.     

Design of a pitch synchronous innovation CELP coder for mobilecommunications

This paper describes the design of a speech coder called pitch synchronous innovation CELP (PSI-CELP) for low hit-rate mobile communications. PSI-CELP is based on CELP, but has more adaptive excitation structures. In voiced frames, instead of conventional random excitation vectors, PSI-CELP converts even the random excitation vectors to have pitch periodicity by repeating stored random vectors as well as by using an adaptive codebook, in silent, unvoiced, and transient frames, the coder stops using the adaptive codebook and switches to fixed random codebooks. The PSI-CELP coder also implements novel structures and techniques: an FIR-type perceptual weighting filter using unquantized LPC parameters, a random codebook with a conjugate structure trained to be robust against channel errors, codebook search with delayed decision, a gain quantization with sloped amplitude, and a moving average prediction coding of LSP parameters, Our speech coder is implemented by DSP chips. Its coded speech quality at 3.6 kb/s with 2.0 kb/s redundancy is comparable to that of the Japanese full-rate VSELP coder at 6.7 kb/s with 4.5 kb/s redundancy. The basic structure of this PSI-CELP coder has been chosen as the Japanese half-rate speech codec for digital cellular telecommunications     

A High-Performance Codec Suitable for the GSM Half-Rate Coding System

1IntroductionTheGSMpanEuropeandigitalradiosystemhas-beendesignedwithaparticularTDMAframestfllcturewhichenablestheusingofeitherfull-rateorhalf-ratechannels.Speechandchannelcodingalgorithmsforfull-ratechannelshavebeenindependentlystandardized,leadingrespectivelytotheRPE-LTPalgorithmandprotectionschemebasedonaconvolutionalcodewithaCRCforerrordetection.StandardiZationofacombinedspeechandchannelhalf-ratecodecataglobalrateofII.4kbpshasstartedunderthecontrolofETSI.Theobjectiveisverychalleng…     

New directions in subband coding

Two very different subband coders are described. The first is a modified dynamic bit-allocation-subband coder (D-SBC) designed for variable rate coding situations and easily adaptable to noisy channel environments. It can operate at rates as low as 12 kb/s and still give good quality speech. The second coder is a 16-kb/s waveform coder, based on a combination of subband coding and vector quantization (VQ-SBC). The key feature of this coder is its short coding delay, which makes it suitable for real-time communication networks. The speech quality of both coders has been enhanced by adaptive postfiltering. The coders have been implemented on a single AT&T DSP32 signal processor     

Multiband linear predictive speech coding at very low bit rates

The authors describe the multiband linear predictive (MB-LPC) vocoder and its operation at 2.4 kb/s and 1.2 kb/s. The MB-LPC vocoder uses mixed excitation and exploits the advantages of both time and frequency domain speech coding techniques to produce natural sounding, good quality speech. Subjective performance of speech at 2.4 kb/s produced by the MB-LPC is very close to that for the 4.15 kb/s INMARSAT-M IMBE speech coder. Informal listening tests have shown that in most cases people could not tell the difference between the new 2.4 kb/s MB-LPC coder and the 4.15 kb/s INMARSAT-M IMBE coder     

Wideband CDMA system for personal radio communications

This article describes a wideband spread code-division multiple access (W-CDMA) system for high-capacity and high-quality personal radio communication. This system has been authorized as an EIA/TIA Interim Standard IS-665, T1P1 Trial Use Standard J-STD-015, and ITU-R Recommendation M 1073. The system uses wideband spreading to accomplish good interference immunity, high-quality speech, and high-speed data transmission. The system uses coherent detection (CD) and an interference canceller system (ICS) to enhance the capacity. The CD and ICS use continuous pilot signals in the forward/reverse links to estimate the propagation path parameters. PN and Hadamard sequences are used as the spreading code for minimal mutual interference between traffic and pilot/sync/paging channels. A robust 32 kb/s waveform speech coder, ITU-T COM101+, has been developed to achieve toll-quality speech in the radio environment. This system provides up to 128 voice channels per cell and data transmission up to 64 kb/s by 5 MHz spreading. Unification of low- and high-mobility applications and reduced complexity of system and hardware configurations are accomplished     

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.     

Evaluation of six medium bit-rate coders for the Pan-Europeandigital mobile radio system

A digital cellular mobile radio system has been under development in Europe since 1982 under the coordination of the working group CEPT GSM (groupe speciale mobile). In a recent coordinated experiment, listening opinion tests were performed on the speech output of six candidate 16 kb/s speech coding schemes for this system: one regular-pulse excited coder, one multiple-excited coder, and four subband coders. For comparison purposes, test conditions from a companded cellular FM system currently in operation were included in the experiment. The six codecs were companded in terms of subjective quality, transmission delay, and ease of implementation. In this overall comparison, no single codec was superior in all respects. However, the regular-phase-excited linear predictive coder, which provided the best speech quality, had acceptable complexity and delay and was singled out for further improvement. Ultimately, an improved version of this codec, a regular-pulse-excited/long-term-prediction LPC coder was selected     

1.8kb/s的多带线性预测语音编码算法

设计了一种数码率为１．８ｋｂ／ｓ的多带线性预测（ＭＢＬＰ）语音压缩编码算法。该算法采用基于谐振结构的线性预测分析和对激励信号采用多带处理的方法。试验结果表明，本算法提供了相当于码率为２．４ｋｂ／ｓ美国联邦声码器标准ＭＥＬＰ的重建语音质量，具有较高的清晰度和自然度。     

一种基于混沌系统的窄带超低速语音加密算法

为了在极低通信速率信道中实现保密语音通信,在北大西洋公约组织的0.6 Kb/s增强型混合激励线性预测声码器算法基础上,提出了基于混沌映射的保密通信算法。算法利用两个Logistic方程产生混沌序列,对声码器码流进行置换和移位。最后对该加密语音通信系统进行了仿真测试。测试结果证明,系统方案切实可行,产生的密钥空间大,能够实现加密强度较高的语音保密通信。     

基于GMM的甚低码率语音编码器

提出了一种新颖的基于高斯混合模型(GMM)的甚低码率语音编码系统.该编码器利用GMM对短时语音谱包络进行拟合的方法来对语音进行参数化表示.编码时,语音经预处理、分帧加窗后,再经FFT分析得到分帧语音的信号频谱,并获得平滑谱包络.然后采用GMM对谱包络进行拟合,用GMM参数(均值、方差、权重)对语音谱加以表示.由于GMM参数较少,从而可以使得码率甚低.解码时,根据编码逆运算生成谱包络,浊音信号利用正弦模型加以合成,清音信号经IFFT合成.实验仿真结果表明:该编码器在传输码率降低到2.35 kb/s时,仍可获得音质令人满意的解码语音.     

小波激励线性预测编码（WELP）：一种新的低速语音编码方法

在低速语音编译码系统中，常采用码本激励线性预测编码ＣＥＬＰ，其中随机码本的码本结构及应的索算法直接影响着语音编译码系统的语音质量和实时实现中的运算量。     

Characterizing the subjective performance of the ITU-T 8 kb/sspeech coding algorithm-ITU-T G.729

Three listening-only experiments were conducted to characterize the subjective performance (i.e., speech quality) of 8 kb/s G.729. These experiments evaluated the quality of coded speech under a variety of conditions: (i) interworking with other international and regional speech coding standards; (ii) input speech that had been corrupted by environmental noise; (iii) operation over degraded transmission channels (including random bit errors and a simulated radio channel). The results of these experiments indicate that 8 kb/s G.729 meets the performance requirements that were established at the beginning of the standardization process     

Subband speech coding and matched convolutional channel coding formobile radio channels

The effects of digital transmission errors on a family of variable-rate embedded subband speech coders (SBC) are analyzed in detail. It is shown that there is a difference in error sensitivity of four orders of magnitude between the most and the least sensitive bits of the speech coder. As a result, a family of rate-compatible punctured convolutional codes with flexible unequal error protection capabilities have been matched to the speech coder. These codes are optimally decoded with the Viterbi algorithm. Among the results, analysis and informal listening tests show that with a 4-level unequal error protection scheme transmission of 12 kb/s speech is possible with very little degradation in quality over a 16 kb/s channel with an average bit error rate (BER) of 2×10^-2 at a vehicle speed of 60 m.p.h. and with interleaving over two 16 ms speech frames     

Fractional rate multitree speech coding

The authors present both forward and backward adaptive speech coders that operate at 9.6, 12, and 16 kb/s using integer and fractional rate trees, weighted squared error distortion measures, the (M, L) tree search algorithm, and incremental path map symbol release. They introduce the concept of multitree source codes and illustrate how the multitree structure allows scalar quantizer-based codes and scalar adaptation rules to be used for fractional rate tree coding. With a frequency weighted distortion measure, the forward and backward adaptive multitree coders produce near toll quality speech at 16 kb/s, while the backward adaptive 9.6 kb/s multitree coder substantially outperforms adaptive predictive coding and has an encoding delay of less than 2 ms. Performance results are present in terms of unweighted and weighted signal-to-noise ratio and segmental signal-to-noise ratio, sound spectrograms, and subjective listening tests     

A speech coder for PC multimedia net‐to‐net communication

The paper presents a speech coding algorithm for operation at 11025 samples/s. The coder provides improved speech quality and compatibility with the MS‐Windows multimedia environment. The coding algorithm has been developed by adapting the ITU G729 and enhancing it with some recent developments in the medium band coding. The coder operates over a band of frequencies ranging from 20 to 5400 Hz at a bit rate of 8.9 kbit/s. Application of this coder includes intranet VoIP, voice chatting, multimedia communications, and voice archiving. Copyright © 2001 John Wiley & Sons, Ltd.     

Design and simulation of an efficient adaptive delta modulationembedded coder

An embedded coding version of hybrid companding delta modulation (HCDM) is described that operates from 16 to 48 kb/s in 8 kb/s steps. The embedded HCDM coder employs the explicit noise coding technique to transmit an adaptive PCM (APCM) coded version of the HCDM reconstruction error signal as a supplementary bit stream that may be partly or wholly deleted in transmission. SNR performance with speech input depends critically on the design of the supplemental APCM code and two new coding algorithms are investigated. In algorithm 1, the basic cue for step size adaptation is obtained from the RMS slope energy of the HCDM output whereas in algorithm 2, the HCDM reconstruction error is logarithmically compressed before quantisation and the basic step size is derived from peak input magnitudes. Instantaneous adaptation for both algorithms is achieved by using step size multipliers which are optimised for operation at single fixed bit rates and also for decoding with an unknown number of input bit deletions. Simulation results show that SNR performance is significantly enhanced using either algorithm and a graceful reduction of reconstructed speech quality with progressive bit deletion is achieved over the range from 48 kb/s to 16 kb/s. On the whole, the SNR performance of the embedded HCDM system is superior in comparison with conventional HCDM     

Adaptive prediction in speech differential encoding systems

The design of speech coders that produce high-quality highly intelligible speech at 6 to 16 kb/s while retaining robustness to background and transmission impairments is an area of current research interest. Differential encoding structures employing adaptive quantization and adaptive prediction constitute one of the most promising approaches to achieving these design objectives. This paper focuses on the design and analysis of adaptive predictors for differential encoders. Several differential encoding systems, including adaptive predictive coding, differential pulse-code modulation, noise feedback coding, direct feedback coding, and prediction error coding, are described and related. Adaptive quantizers are briefly discussed and quantitative and qualitative indicators of speech coder performance are defined. The channel model, the speech model, and the research problem statements used in the design of differential encoders and adaptive predictors are presented. The nomenclature and theory of forward and backward adaptive prediction are developed, and several new backward adaptive algorithms based on various assumptions are presented. A detailed survey of theoretical and simulation results on adaptive prediction for speech differential encoders is given, and the effects of background and transmission impairments on these systems are discussed, Finally, the impact of adaptive predictors on rate distortion theory motivated coders is indicated. Numerous areas for future research are highlighted.     

A high-quality multirate real-time CELP coder

The design and implementation of a real-time CELP coder for mobile communication applications are discussed. To realize a single-chip implementation, several tradeoffs were made without compromising speech quality. In addition, techniques that make the coder more robust under a variety of channel conditions are discussed. The real-time coder can be operated at different bit rates (8, 6.8, 4.6 kb/s) by simply changing the frame update rates. The speech quality was evaluated through a formal listening test, and it was found that this coder compares favorably with other (standardized) coders operating at similar or higher rates