Since missing-packet recovery techniques for conventional PCM speech are not applicable to packetized speech communication systems with low-bit-rate coding schemes, quality degradation mechanisms are presented for missing-packet recovery techniques. These mechanisms are least significant bit (LSB) dropping, waveform substitution, and odd-even sample-interpolation schemes. A comparison of these techniques in terms of signal-to-noise ratio and perceptual distortion under packet loss conditions shows that the LSB-dropping scheme with embedded coding is the most promising technique for recovering missing packets.>

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.>

The International Telegraph and Telephone Consultative Committee (CCITT) has recommended an algorithm to code wideband speech and music (i.e. 7-kHz audio) with 64 kb/s as an international standard (recommendation G.722). A multimedia multipoint teleconference system using the 7-kHz audio coding standard is discussed. System requirements for teleconferencing are examined, and control procedures for the system are examined. A system using digital leased circuits is presented, and the audio-bridging technique used in this system is discussed.>

An adaptive predictive coding with dynamic quantization adjustment(APC-DQA) is proposed for speech coding at 16 kbits/s, which aims to reduce processing delay and hardware complexity, while attaining "toll quality". The proposed scheme utilizes time-domain processing to lessen the processing delay. Moreover, it employs backward processing in both prediction and quantization, which requires less side information than forward processing. It also incorporates adaptive bit allocation in sub-intervals of each frame so as to remove redundancies due to periodic concentration of the prediction residual energy. The performance evaluation results show that the processing delay of the APC-DQA is 2/3 that of an adaptive predictive coding with adaptive bit allocation scheme (APC-AB) [1]. Moreover, its hardware complexity is approximately 70% to 80% of the APC-AB. It was also shown that this scheme can provide speech quality subjectively equivalent to 6.6 bit Log-PCM.

CCITT (International Telegraph and Telephone Consultative Committee) Study Group XVIII is in charge of studies on speech processing, such as speech coding techniques. During its last study period (1981-84), a recommendation for 32 kbit/s speech coding was made. Afterwards, Study Group XVIII began standardization studies on wideband speech coding (i.e., 7 kHz audio coding) at 64 kbit/s or less. This paper reports on the progress of these standardization studies. The 32 kbit/s Adaptive Differential PCM (ADPCM) standard is briefly reviewed. Also, the 7 kHz audio coding at 64 kbit/s or less is mainly discussed focussing on its applications, design requirements and coding algorithm.

Recently much intensive research of 16kbps Speech coding algorithm has been conducted aiming to reduce the transmission bit rate and yet provides high speech quality. Adaptive predictive coding with adaptive bit allocation (APC-AB)[1] is considered to be one promising approach. However, the processing of this coding algorithm is so complicated that the implementation of the algorithm on a general-purpose signal processor, especially if fixed-point arithmetic DSPs are used, requires careful study of arithmetic operation precision and same way to reduce the number of processing cycles. Taking account of these points, real-time signal processing using a fixed-point signal processing chip (FDSP-3) has been studied, and a prototype codec has been realized. The prototype codec satisfied the CCITT mask of signal-to-total distortion ratio for PCM codecs and showed quality good enough for "toll" speech.

The synchronous tandem property of nonaccumulation of distortion in tandem-connected ADPCM coders with a 64 kbit/s PCM interface is discussed here. The synchronous tandem algorithm used to provide this property in the steady-state mode is described with a case-by-case analysis, so as to show how the synchronous tandem property is realized in an ADPCM coder. A 32 kbit/s ADPCM coder utilizing this algorithm has been standardized by the CCITT (International Telegraph and Telephone Consultative Committee). The synchronous tandem property of the 32 kbit/s ADPCM coder is of great interest in network applications, because the ADPCM coder appears likely to be introduced into digital networks built partially with existing 64 kbit/s PCM circuits.