Algorithm of Adaptive Bit Allocation Wavelet Transform Audio Coding

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The big squeeze-a CELP speech codec

The coding algorithm widely recognised as offering the best prospects for delivering toll-quality speech at very low bit rates is called CELP (codebook-excited linear prediction) coding. The CELP codec by Delphi Systems operates in real time, uses a standard digital signal processing chip, and encodes speech at 4.8 and 6.5 kbit/s. The use of this speech compression codec (SCC) is also discussed     

Codage algébrique de la parole : prédiction linéaire à excitation par code ternaire

This article presents new speech coding methods for real time application (telephone, videophone) or offline applications (storage). Speech quality is in the classical telephone range, with a 4 kHz bandwidth and a sampling at 8 kHz. An elementary approach leads to a 16 kbit/s codec and a 24 kbit/s codec, using integer codebooks and fast computations. The speech quality of the two codecs has been measured in comparison with more complex ones and in realistic conditions, with noisy telecommunication channels. The elementary approach is completed by a synthetic model, with a systematic generalization of the algorithms (e.g. for a generalized vselp). Some methods for channel protection, which are already known by the speech coding researchers, are summed up in the Appendix. A change of representation for low density codes (less than 1 bit/sample) is proposed.     

基于小波变换和音质模型的音频编码算法研究

音频编码要解决的问题是以最小感知失真用低速率表达音频信号.本文设计了一种基于正交小波变换和音质模型的自适应比特分配音频编码算法,它可以将1411.2kbit/s的双声道立体声高保真音频信号压缩成低至32kbit/s的速率,并保持很好的音频质量.     

`Nearly instantaneous? digital compandor for transmitting six sound-programme signals in a 2.048 Mbit/s multiplex

A `near-instantaneous? digital compandor for the transmission of high-quality sound signals is described that reduces the bit rate from 416 kbit/s to about 322 kbit/s per channel without noticeable impairment of the sound quality. Hence six audio channels can be multiplexed to form a 2.048 Mbit/s stream including frame synchronisation and transmission error-protection facilities.     

TRANSMISSION PERFORMANCE EVALUATION OF VOICE ENCODING TECHNOLOGY FOR THE INMARSAT MINI-M SYSTEM

In this paper are presented the method and results of a subjective evaluation, which was conducted in order to select a new speech codec for the Inmarsat mini-M system. The mini-M system is designed to provide the next generation of global, notebook-sized satellite terminals for transportable, land-mobile and maritime voice, facsimile, and data communications. Overall, six different codecs operating at a combined source and channel rate of 4⋅8 kbit/s were evaluated in a series of six subjective tests. From this, it was concluded that one codec was able to deliver performance that is equivalent to, or better than, the IS-54 full-rate ditigal cellular 8 kbit/s VSELP codec, and was selected for use in the mini-M system.     

基于VC5409的双速率语音编译码器的实现

ITU-T.G.723.1为国际电信联盟(ITU)制定的5·3bit/s和6.3kbit/s双速率语音编码建议,分别采用代数码激励线性预测(ACELP)算法和多脉冲最大似然量化(MP-MLQ)算法。在阐述G.723.1建议编译码算法的原理和实现的基础上,重点介绍了在开发基于TMS320VC5409实时实现该建议的全双工编译器过程中所做的工作。该语音编译码器通过了G.723.1所有测试矢量的验证。     

Integration of harmonic and analysis by synthesis coders

Hybrid coding of speech has been proposed to overcome the limitations of a single model in representing the wide variety of characteristics of human speech. A new hybrid coding algorithm, which combines harmonic and analysis by synthesis coding techniques, is presented. To integrate the harmonic and analysis by synthesis coders, novel phase synchronisation and speech classification techniques are developed. The perceptual quality of the speech synthesised using the unquantised hybrid model is almost indistinguishable when compared with 128 kbit/s linear PCM. Two variable rate coders are developed based on the designed hybrid model, by quantising the parameters at different bit rates. Subjective listening tests show that the speech quality of the variable rate hybrid coders outperform the quality of 5.3 kbit/s and 6.3 kbit/s ITU G.723.1 coders, at maximum bit rates of 4 kbit/s and 6 kbit/s respectively.     

Conditional entropy encoding of log-PCM speech

The conditional dependence between the current and previously quantised samples in 64 kbit/s ?-law PCM speech is exploited to noiselessly reduce the average transmitted bit rate. From the conditional probabilities, the conditional entropy and average Huffman code word lengths were computed. The average data rate reduction for the first-order noiseless encoder was found to be 12 kbit/s.     

一种可变速率的LD—CELP编码的研究

设计了一种可变速率的低时延、码激励线性预测编码（ＬＤ－ＣＥＬＰ）的方案，它是通过修改码本来实现的。该方案工作在１１．２ｋｂｉｔ／ｓ。对其做了计算机仿真，并与１６ｋｂｉｔ／ｓ的ＬＤ－ＣＥＬＰ算法在信经（ＳＮＲ）、波形等方面进行了对比，仿真结果表明效果良好。     

Vector-quantised transform coder for speech coding at 9-6 kbit/s and below

The transform approach to speech coding has been established for some time, and has been shown to be very efficient in controlling the bit allocation and the shape of the noise spectrum. Various transform coders have been reported which produce high-quality digital speech at around 16 kbit/s. Although these coders can maintain good quality down to about 9.6 kbit/s, they perform poorly at lower bit rates. Here we discuss how vector quantisation (VQ) can be used to improve the quality of transform coders. We describe one specific design of vector-quantised transform coder (VQTC) which follows on from earlier work, and which is capable of producing good-quality speech at as low as 4.8 kbit/s.     

多媒体终端中声音和数据的集成传输

本文描述了采用包复用方式在固定带宽内集成传输声音和多媒体数据的多媒体终端通信系统，系统中的声音编码采用了静默检测技术，声音编码的速率可以根据信道的拥挤程度在３２ｋｂｉｔ／ｓ和１６ｋｂｉｔ／ｓ之间动态地变化。本文提出了一种利用增减静默抽样来同步声音编解码时钟的方法，本文还提出了利用数据队列的短时平均长度来判断信道繁忙程度的算法，在多媒体数据突发性强、数据量大时，该算法比利用声音或数据队列的瞬时长度判断更为准确。     

Natural quality two-band LPC coding of speech at 880 bit/s withframe interpolation

A novel frame interpolation technique for two-band linear predictive coding (LPC) vocoders is proposed for maintaining natural speech quality at bit rates below 1 kbit/s. Experimental results show that the speech quality of the proposed vocoder is quite natural at bit rates 880 bit/s and comparable to that of 4.8 kbit/s CELP     

G.718: A New Embedded Speech and Audio Coding Standard with High Resilience to Error-Prone Transmission Channels

This article is an overview of the standardization, architecture, and performance of the new ITU-T Recommendation G.718. G.718 is an embedded variable bit rate codec providing a scalable solution for compression of 8 and 16 kHz sampled speech and audio signals at rates between 8 kb/s and 32 kb/s. It comprises five layers where higher-layer bitstreams can be discarded without affecting the lower layersiquest decoding. The codec also has an optional core layer interoperable with ITU-T G.722.2 (3GPP AMR-WB) at 12.65 kb/s. G.718 was designed to provide high speech quality at low bit rates and to be robust to significant rates of frame erasures or packet losses. It is also targeting good quality for generic audio at higher rates.     

Design and performance of robust embedded ADPCM coder

An embedded adaptive DPCM (EADPCM) speech coder is described which allows bit rate reductions to be achieved by progressive deletion of bits from output codewords. Optimised step size multipliers are given for a robust implementation using an improved algorithm for adaptive quantisation. Simulation shows that a graceful reduction in speech quality with bit rate is achieved in the range 16-48 kbit/s.<>     

Implementation of a 32 Kbit/s ADPCM Codec Using a General-Purpose Digital Signal Processor

This paper describes an implementation of a CCITT G.721 compatible 32kbit/s ADPCM codec, using a general-purpose digital signal processor FDSP-3 (MB8764). A single-channel ADPCM codec is realized by two FDSP-3 chips-one for the encoder and the other for the decoder. Meticulous programming techniques are employed to achieve exact computation of the CCITT algorithm exploiting all the available resources of the 16-bit fixed-point DSP. It is shown that the whole codec computation can be accomplished in about 2350 machine cycles. Thus, two FDSP-3 chips operating at 10 MHz machine cycle can handle the whole computation. The paper also covers the comparison of straight fixed-point format and the G.721 realization, and briefly examines the compatibility issue between these two methods.     

Investigation into adaptive-rate modems for meteor burstcommunication

Current meteor burst communication systems operate at constant bit rates of 4 kbit/s and are limited to average throughputs of 50 bit/s. The paper documents the design and implementation of an adaptive-bit-rate modem for optimising the throughput. A novel extension of differentially encoded binary phase shift keying is proposed so that two frequency division multiplexed baseband channels can be transmitted: a main channel for the data and an auxiliary channel for conveying bit rate change information. In this way, two modems, one at each end of the link, can automatically adapt to a new operating speed without the intervention of a host computer. The results prove that the two channels can operate independently with minimal mutual interference. A theoretical improvement in bit rate of 18 times over the fixed-bit-rate modem is possible     

深空通信Ka频段数传发射机基带电路单元实现

基于Xilinx FPGA电路的全数字化设计方案,研制完成适用于深空通信下行链路Ka频 段发射机中基带数据编码调制一体化电路单元。参照CCSDS（Consultative Committee for Space Data Systems）相关深空通信建议标准,电路单元实现了按码速率的变化灵活选择调 制方式的工作模式,利用外部控制指令,完成码速率16 bit/s～20 kbit/s、20～ 200 kbit/s、200 kbit/s～2 Mbit/s分段分别选择PCM/BPSK/PM、N RZ/BPSK和SRRC-QPSK数据调制方式 。在X频段的测试结果表明,BPSK和SRRC-QPSK幅度误差和相位不平衡分别小于3.1%和1.7° ,符合CCSDS关于深空通信的建议标准。电路单元满足深空通信工程应用需求。