Two approaches are presented for accommodating 9.6 kb/s modem signals (e.g. V.29) through 32 kb/s ADPCM (adaptive digital pulse-code modulation) links. These are small changes in the existing algorithm and coding with ADPCM incorporating a 5-bit, rather than 4-bit quantizer. For each approach, tradeoffs between performance and implementation complexity are described.>

Ratings obtained from subjective tests indicate that for single encodings of speech, near equivalence to 64 kb/s PCM is obtained at 32 kb/s with ADPCM/AP and SBC, and at 24 kb/s with APC and ATC/VD. Tandem encodings of 32 kb/s ADPCM/AP are significantly better than 16 kb/s APC and SEC. Eight synchronous encodings of 32 kb/s ADPCM/AP are rated about the same as eight asynchronous encodings of 64 kb/s PCM. The subjective test results are used to develop preliminary coder performance functions in terms of Q (subjective ratio of speech power and speech-dependent noise power) in terms consistent with a previously reported quantization distortion opinion model. These relations, in conjunction with a proposed CCITT planning rule, indicate that a 32 kb/s ADPCM/AP coder can be allowed in each of the national extensions of an international connection. Reported results apply only to coder quantization effects on speech quality; other signals (e.g., voiceband data, signaling tones) and impact of coder delay on echo performance are not covered.

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.

This paper describes a minicomputer facility developed to aid in the evaluation of the grade-of-service impact of digital codecs. Two minicomputers are interfaced with 15 bit A/D and D/A converters so that an analog signal such as speech can be quantized and stored in real time at sampling rates up to 72 kHz. The acquired digital samples can then be processed by various PCM, ADPCM, and ADM algorithms programmed on the system. After the processing is completed, the facility can automatically generate high quality audio tapes which are suitable for subjective evaluation in an absolute judgment or balance test format. The two minicomputers operate on a stand-alone basis so that the capability exists for simulating either similar or dissimilar tandem encodings in a synchronous or asynchronous mode. A few sample outputs of the facility are also given in the form of subjective test results for single and tandem encodings of μ255 PCM and Nearly Instantaneous Companded (NIC) PCM.