A New Speech Codec Based on ANN with Low Delay

The author designs a new speech codec in this paper, which is based on ANN to carry out nonlinear prediction. This new codec synthesizes speeches with better quality than the conventional waveform or hybrid codecs does at the same bit rate. Moreover, the most important characteristic of this codec is the low coding delay, which will benefit the enhancement of the speech communication QoS when we transmit speech signals in IP or ATM networks.     

MNB2算法在编解码器客观评估中的应用研究

本文讨论MNB2算法在编解码器质量的评估中的应用，探讨了编解码器进行客观评估的声学心理学基础，以及在此基础上建立的MNB2算法。深入剖析了MNB2算法的组成块TMNB，FMNB，并应用该算法对多种编解码器进行客观评估。实验证明该客观评估算法中主观与客观相关程度高，有较强的适应性，可靠性，实用性，完全可以应用于新型编解码器质量评估。     

DSP implementations of sophisticated speech codecs

Hardware implementation aspects of sophisticated speech codecs are addressed. The major points discussed are approaches to implementation of the sophisticated speech codecs, requirements for DSP (digital signal-processing) implementation of the codecs, such as the type of arithmetic processing and the necessity of bit-level specifications, and codec implementation and DSP programming techniques for three specific coding algorithms: 32-kb/s ADPCM (adaptive digital pulse code modulation), 64-kb/s (7 kHz) SB (subband)-ADPCM, and 16-kb/s APC-AB (adaptive predictive coding with adaptive but allocation) codecs     

G.723.1 Codec在OpenH323协议栈上的实现

OpenH323是一个开放源码的VoIP(Voice over IP)协议栈,支持H．323和SIP等多媒体通信协议,为多媒体应用提供了一个很好的开发平台。G．723．1是ITU-T建议在中低速率多媒体通信中使用的语音压缩算法,目前该算法已在IP电话系统中得到广泛应啊。基于OpenH323协议栈实现G．723．1Codec有着十分重要的应用价值。介绍在OpenH323的软件终端上实现G．723．1Codec的基本方法,并可推广到G．729等其它多种语音压缩算法。     

iLBC语音编解码器核心技术的研究

重点讨论了iLBC编解码器独立于帧的长期预测。独立于帧的长期预测是用来在编码语音没有遭受与传输丢失相关的多帧语音退化情况下,开发斜度标记相关的办法。然后介绍了iLBC,G.729A和G.723.1编解码器的平均主观得分MOS,并用信号为例说明基于独立于帧的长期预测编解码器和CELP编解码器之间的不同,最后用语音重构的例子说明二者语音质量间的差别。     

SA-AMR算法性能分析与研究

在AMR的基础上介绍了一种用来提高系统容量的源自适应(SA-AMR)算法.简要阐述了算法编解码原理,分析了在无线通信系统中引入SA-AMR技术的优点,并与AMR进行性能比较,测试结果表明:SA-AMR编码速率和模式可变,能够提高语音质量和发射稳健性,有效增大系统容量,性能优于AMR.     

一种基于SBC技术的16kbit/s高效语音编译码器

本文介绍一种基于SBC技术的16kbit／s高效语音编译码器．在该编码方案中，IIR型正义镜象滤波器（QMF）用于实现语音分带，而基于快速搜索技术的矢量量化（FVQA）用于对除基带（采用ADPCM技术）外的其余高频带语音编码。一片TMS320C25芯片实现了两路16kbit／s语音编译码器，并借此实现了两人双向实时通信。得到良好的语音可懂度、自然度及客观评测结果。     

A Low-rate Data Transfer Technique for Compressed Voice Channels

This study considers the problem of transmitting generic (non-speech) data through compressed voice channels such as those used in wireless communications networks. These networks employ voice codecs that are designed to efficiently encode and reproduce the relatively slow-changing signals of human speech, which leads to communication channels that are nonlinear and have long-term memory. A data modem is presented that utilizes finite alphabets of waveforms that are numerically optimized to be as separable as possible after passing through the voice codec. The optimization of the finite alphabet is performed using a pattern search algorithm. When used with the GSM Enhanced Full-Rate Voice Codec, this system demonstrated improved performance, in terms of error rates, compared to previously reported results. Simulation results for four other voice codecs are also presented.     

Source Models for Speech Traffic Revisited

In this paper, we analyze packet traces of widely used voice codecs and present analytical source models which describe their output by stochastic processes. Both the G.711 and the G.729.1 codec yield periodic packet streams with a fixed packet size, the G.723.1 as well as the iLBC codec use silence detection leading to an on/off process, and the GSM AMR and the iSAC codec produce periodic packet streams with variable packet sizes. We apply all codecs to a large set of typical speech samples and analyze the output of the codecs statistically. Based on these evaluations we provide quantitative models using standard and modified on/off processes as well as memory Markov chains. Our models are simple and easy to use. They are in good accordance with the original traces as they capture not only the complementary cumulative distribution function (CCDF) of the on/off phase durations and the packet sizes, but also the autocorrelation function (ACF) of consecutive packet sizes as well as the queueing properties of the original traces. In contrast, voice traffic models used in most of today's simulations or analytical studies fail to reproduce the ACF and the queueing properties of original traces. This possibly leads to underestimation of performance measures like the waiting time or loss probabilities. The models proposed in this paper do not suffer from this shortcoming and present an attractive alternative for use in future performance studies.      

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.     

一种新的语音二项式正弦脉冲激励方案

本文提出一种新的用于LPC语音编码器的BSP激励信号,即根据语音产生的原理,以一个幅度受到二项式调制的正弦波BSP（Binomial Sine Pulse)作为LPC激励源,该二项式反映了激励信号在一个基音周期内的变化趋势。本文同时推导了BSP激励参数的求取和改进方法。实验结果表明,在此基础上构造的BSP语音编解码器具有低复杂度、低时延的优点,同时编码速率在低至2．65kb/s时,具有较高的合成语音质量。     

Speech codecs for personal communications

Some of the many factors that go into the design of a personal communications system (PCS) are considered. The ways in which these factors affect the speech codec are discussed. Cell size, multiple access methods, and the trade-offs in PCS design are examined. Possible future designs of PCSs and their codecs are described     

The adaptive multirate wideband speech codec: system characteristics, quality advances, and deployment strategies

Wideband speech is the major differentiation and attraction of third-generation network services in both the circuit and packet switched domain. Increased audio bandwidth introduces a significant leap in perceived quality of service compared to currently utilized narrowband telephony in second-generation mobile communications and the PSTN. The adaptive multirate wideband (AMR-WB) speech codec is the service enabler for improved user experience. It is an established 3GPP and ITU-T wideband speech codec standard and represents the state-of-the-art in speech quality as well as robustness in error prone radio channels. It is also the first codec algorithm standardized for wideband speech for mobile communications.     

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.     

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     

An adaptive noise canceller with low signal distortion for speechcodecs

This paper proposes an adaptive noise canceller (ANC) with low signal distortion for speech codecs. The proposed ANC has two adaptive filters: a main filter (MF) and a subfilter (SF). The signal-to-noise ratio (SNR) of input signals is estimated using the SF. To reduce signal distortion in the output signal of the ANC, a step size for coefficient update in the MF is controlled according to the estimated SNR. Computer simulation results using speech and diesel engine noise recorded in a special-purpose vehicle show that the proposed ANC reduces signal distortion in the output signal by up to 15 dB compared with a conventional ANC. Results of subjective listening tests show that the mean opinion scores (MOSs) for the proposed ANC with and without a speech codec are one point higher than the scores for the conventional ANC     

The design of a hybrid adaptive quantizer for speech codingapplications

A new adaptive quantizer which uses a combination of instantaneous and syllabic adaptation is presented for use in speech codecs. It can be designed to adapt to changes in the mean, variance, and pdf shape of its input signal, and to quantize the signal using one or more bits/sample. It is therefore called the generalized hybrid adaptive quantizer (GHAQ). An efficient procedure for optimizing the GHAQ using a training sequence of signal samples is described, and the effects on the performance of the GHAQ of varying the memory length and the syllabic compandor time constant are investigated. It is found that an optimized version of the two-bit GHAQ offers improved signal-to-noise ratio over Jayant's adaptive quantizer with a one-word memory when it is used in a predictive speech codec with a zero-, first-, or second-order fixed predictor     

MPEG-2 spatial scalable coding and transport stream errorconcealment for satellite TV broadcasting using Ka-band

An effective MPEG-2 spatial scalable video codec is designed, and error concealment technique of associated transport stream is proposed. The spatial scalability can provide robust error resilience for Ka-band rain attenuation, as well as the co-existence of HDTV and SDTV systems. In MPEG-2, the spatial scalable encoder combines both spatial and temporal predictions. This paper proposes a near optimal spatial-temporal weighting analyzer to properly assign weights. Simulation shows that the proposed spatial scalable codec structure outperforms the performance of the other MPEG-2 codecs for TV broadcasting in Ka-band using satellites. We also propose a suitable error protection and concealment method for MPEG-2 transport stream. With the combination of scalable coding and error protection systems, the proposed system will achieve high link availability     

英国电信开展视频监控与报警服务的研究

本文阐述了英国电信开展视频监控与报警服务的业务模式,分析了其业务特点以及其成功的运营经验,从而为我国运营商开展视频监控业务提供借鉴.     

Time-Shared Codecs for Multichannel Delta Modulation and Differential PCM

Current practice for multichannel delta modulation (DM) terminals uses one DM codec per channel-end. This paper describes methods by which one high-speed DM codec can be timeshared over a large number of channels. The methods are also applied to multichannel differential PCM (DPCM) terminals. In both cases the time-shared codecs include step-size adaptation determined by the recent past history of the coded digital signal. A method for digital conversions between multichannel linear PCM and adaptive DPCM (ADPCM) formats is also described.