A remote auscultation support system was developed that compresses and records in real time the patient's breath sound and heart sound, obtained using a stethoscope, and sends this data to an attending doctor at a hospital via network. For real-time recording of the breath sound and heart sound, special-purpose, high-quality sound coding technology was developed and incorporated in the system. This sound coding technology enables the amount of data to be reduced to about 1/18 with virtually no deterioration of the properties of the auscultation sound, high-speed transmission of this data using network, and remote diagnosis of the auscultation sound by a medical specialist. The auscultation locations of each patient, together with the doctor, stethoscoper, and patient database are input into the system in advance at the hospital. At the patient's home or sanatorium, the auscultation sound is recorded according to a human body display that shows auscultation locations, and then sent to the hospital. To ensure patient confidentiality when the auscultation data is transmitted via network, the system scrambles the auscultation data and allows only the attending doctor to play and diagnose the auscultation sound. These features not only support an understanding of the condition of patients being treated at home, but they also enable the construction of an auscultation database for electronic charts that allows auscultation results to be shared within the hospital. When this remote auscultation support system was manufactured and its performance was assessed, virtually the same waveform was obtained for the recorded and played breath sound as for the original breath sound. Results showed that even at a sampling frequency of 11 kHz, remote diagnosis by a medical specialist was in fact possible. Furthermore, if auscultation data of 10 seconds per location for 10 locations is sent, the amount of data sent is only about 120 Kbytes. Since this amount of data converts to only about 25 pages of electronic mail text, even via the existing mobile network the auscultation sounds of many patients can be sent efficiently.

Low bit-rate speech and audio codings are key technologies for multimedia communications. A number of coding scheme have been developed for various applications. In Internet application, good speech and audio quality at very low bit-rate (8-16 kb/s) is valuable. Two recently proposed speech and audio-coding schemes, CS-ACELP (Conjugate Structure Algebraic Code Excited Linear Prediction, standardized by the ITU-T in Recommendation G. 729) and TwinVQ (Transform-domain Weighted INterleave Vector Quantization, one of the candidates for MPEG-4 audio) were compared from the viewpoints of coding schemes and quality. Although there are significant differences in their basic structures and frame lengths, this paper describes that both use the same compression techniques, such as LPC (Linear Predictive Coding)-analysis pitch-period estimation and vector quantization. While CS-ACELP provides toll quality for speech at 8 kb/s, the quality it provides for music signals is insufficient. The TwinVQ transform coder is based on LPC and vector quantization and is also capable of operating at 8 kb/s. Evaluation of these two schemes in terms of their fundamental technologies, quality, delay, and complexity showed that the quality of TwinVQ for music signals is better than that of CS-ACELP, and that the quality of CS-ACELP is better for speech signals. Therefore, TwinVQ may be better suited for one-directional Internet applications, and CS-ACELP may be better for two-directional communication.