Entropy coding of ITU G.728 speech parameters for low bit rateaudiovisual communications

Entropy coding principles are applied to the 16 kbit/s ITU G.728 speech codec. It is shown that the average bit rate can be reduced to 14.5 kbit/s without a significant increase in the codec complexity. In very low bit rate audiovisual communication applications such as the videophone, the saved bits can be used to improve the output video quality     

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.     

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。     

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     

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     

A Discretely Adaptive Delta Modulation Codec

A discretely variable slope delta modulation (DVSD) codec is described, which is suitable for integrated circuit realization. The step size is varied by a pulse number modulation method that does not require a precision digital-to-analog conversion circuit. An adaptation algorithm is discussed, taking into consideration the effect of transmission errors. The quantizer and integrator portion has been fabricated on a monolithic chip using MOS technology. Results obtained from an experimental 32 kbit/s codec demonstrate its excellent performance.     

A 64 kbit/s Videophone codec with forward analysis and control

A new combination of coding methods for a 64 kbit/s transmission system for typical videophone situations is investigated. The codec structure is based on a standard hybrid discrete cosine transform (DCT) codec with temporal prediction. The picture is divided blockwise into changed and unchanged areas. One motion vector with subpel accuracy is computed and transmitted for each block of the changed area. For the forward analysis, the prediction error is calculated in the whole picture. Only the blocks with the highest prediction errors are updated by a DCT with a perception adaptive quantization. The number of DCT update blocks depends on the remaining bits after the transmission of the overhead information. The codec is controlled by a forward analysis of the prediction error and is not based on a buffer control. The spatial resolution of the source signal is reduced in two steps to prevent a codec overload caused by too much activity between two frames.     

Subjective Evaluation of Several Efficient Speech Coders

The voice quality of several 9.6 - 32 kbit/s coders is determined with an extensive set of subjective listening tests. Single encodings of μ255 PCM, adaptive differential PCM (ADPCM), subband coding (SBC), vocoder-driven adaptive transform coding (ATC), adaptive predictive coding (APC), and time domain harmonic scaling combined with SBC are compared in an idealized situation, that is, no added impairments. It is shown that single encodings of modest complexity 32 kbit/s coders such as ADPCM and SBC and more complex 24 kbit/s coders such as vocoder-driven ATC and APC offer quality nearly equivalent to 64 kbit/s μ255 PCM. However, these conclusions are drawn in the absence of a realistic telephone network where tandem encodings, delay limitations, and nonvoice signals exist. Tandem encodings of 64 kbit/s μ255 PCM, 32 kbit/s ADPCM, 16 kbit/s SBC, and 16 kbit/s APC are also evaluated. These 32 kbit/s and 16 kbit/s coders offer degraded tandem performance as compared to 64 kbit/s PCM, with the exception of synchronous tandeming of 32 kbit/s ADPCM with 64 kbit/s PCM where several encodings are subjectively equivalent to a single encoding of 32 kbit/s ADPCM.     

Low bit-rate video coding with spatio-temporal geometric transforms

A low bit-rate video coding technique that uses spatio-temporal geometric transforms is presented. Motion compensation based on the bilinear transform is employed to reduce the temporal redundancy of the video. The spatial redundancy of the motion compensated error images is reduced by a combination of fractals and the DCT. It is shown that in the objects boundaries of the motion compensated error image fractals outperforms the DCT, while in the smooth areas the DCT is better than fractals. A hybrid combination of fractals and the DCT gives the best result. The performance of this hybrid codec with geometrically transformed motion compensation is compared against the H.261 standard video codec at 64 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.     

DRA音频解码算法及其DSP实现

在研究DRA多声道数字音频解码算法的基础上,设计了基于飞思卡尔公司DSPB56367的DRA多声道数字音频解码器,给出了软硬件设计方案并讨论了其关键技术。音频主观听音测试结果表明:所设计的解码器在128 kbit/s的立体声音质总体优于4.7分,在384 kbit/s的5.1环绕声音质总体优于4.5分,达到了实际应用的需求。     

一种谐波正弦语音模型的最佳相位估计算法

 基于谐波正弦语音模型(HSSM),利用最小二乘方法估计语音模型的最佳相位参数,给出了一种估计相位的批处理方法和迭代算法.把利用该算法得到的相位参数用于宽带语音编解码算法进行仿真,其结果与G.722.2标准宽带编码算法中的两种编码速率8.85kbit/s及6.60kbit/s的语音进行了比较,语音波形的比较和主客观测试结果表明该最佳相位估计算法相位参数估计准确有效,可由此建立的语音模型获得较高质量的合成语音.     

A single-chip CMOS PCM codec with filters

A complete PCM codec using charge redistribution and switched-capacitor techniques will be described. The device is implemented in a two-level polysilicon CMOS technology using 23.4 mm/SUP 2/ of active area. It features all the required transmission filters needed for telephony, two on-chip voltage references, TTL compatible digital interfaces, and low-power dissipation. The architecture of the chip allows asynchronous operation, a variable PCM data rate from 100 kbit/s to 4.096 Mbit/s, /spl mu//A law operation via pin selection, and gain selection at either of two levels in each direction.     

`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 quality of north American cellular, personalcommunications, and public switched telephone networks

Future long distance, and especially international calls, will involve an increasing number of multilink circuits of cellular, personal communications, mobile satellite, and public switched telephone network (PSTN) type of connections incorporating a variety of speech coding devices. In particular, the rapid growth of cellular communications has highlighted the need to characterize the quality of switched networks when cellular terminals are attached at their termination nodes. At the same time, the nonlinear nature of low-rate parametric speech coding has rendered questionable analytical methods for estimating end-to-end voice quality of interconnected telecommunications networks. Instead, quantification of transmission performance appears to require direct subjective evaluation of the pertinent conditions of interest. In this paper the quality of interconnected North American digital cellular and future microcellular terminals with 16 kbit/s and 32 kbit/s DCME/PCME-based switched and private telephone networks is quantified. From these assessments it can be concluded that cellular networks employing the TIA IS-54 8 kbits/s VSELP algorithm may meet the end-to-end transmission planning criteria when interconnected with the switched network     

开放型量子保密通信系统架构及共纤传输技术研究与实验

量子保密通信因其具有理论上的无条件安全性,在国防、金融、政务、商业等领域的应用潜力巨大,同时也对通信网络提出了一定的要求。在研究量子保密通信的基础上,提出了开放型量子保密通信的系统架构;同时为了更好地与现有通信系统融合,提出了一种量子密钥分发（QKD）系统与大容量光通信系统共纤传输方案,并通过实验验证了QKD系统和80×100 Gbit/s DWDM系统共纤传输的可行性,在超过100 km共纤（G.654超低损光纤）传输条件下同时实现了QKD成码率超1 kbit/s和8 Tbit/s DWDM系统无误码传输。     

8Mbit／s数字电视系统的设计与实现

介绍了８Ｍｂｉｔ／ｓ数字电视系统的系统构成，给出了图象压缩编解码，声音压缩编解码和数据通信接口的实现方案，对系统中所采用的新缓存控制策略也作了分析。     

Highly Sensitive Speech Detector and High-Speed Voiceband Data Discriminator in DSI-ADPCM Systems

This paper describes a highly sensitive speech detector and a high-speed voiceband data classifier capable of discriminating between speech and voiceband data of a 4.8 kbit/s 8-phase PSK and 4.8 kbit/s 8-point QAM, and a 9.6 kbit/s 16-point QAM as described in a CCITT recommendation. The presence of a speech signal is detected by analyzing short-time energies, zero-crossing rates, and sign bit sequences of the input signal. The proposed speech detector, with a short hangover time of 32 ms, is able to reduce the average talk spurt activity in an international satellite link to 36 percent. This detector can also classify the detected speech into narrow-band or wide-band spectrum sounds or a low power sound for a variable rate ADPCM encoding. Discrimination between speech and high-speed voiceband data is based on short-time energies, a zeros-crossing rate and linear prediction coefficients of an adaptive predictor. Classification among a 4.8 kbit/s 8-phase PSK and 8-point QAM, and a 9.6 kbit/s 16-point QAM can also be performed by an average prediction gain and a coefficient of variation of the short-term amplitude distribution of the input signal. Discrimination of voiceband data was performed successfully, and erroneous discrimination of talk spurt of telephone speech as voiceband data were, respectively, four times for two two-party conversations lasting 5 minutes in an international satellite link. This is equivalent to less than 0.09 percent of the conversation time.     

H.261会议电视系统的硬件实现及其控制策略研究

介绍按国际建议设计并实现的会议电视系统，对其构成、设计特点及硬件实现，特别 对其核心技术－符合ＩＴＵ－ＴＨ．２６１建议的视频编解码硬件实现做了初步探讨，并提出了有效的传输缓存器控制和自适应量化策略。