A new concept for encoding speech amplitude time quantization

For digital modulations applied to the coding of speech signals, a fixed sampling and transmission rate is always chosen. For commercial telephone these rates are respectively, 8 KHz and 64 Kbits/sec, corresponding to a filtered signal bandwidth of 300 to 3300 Hz. A new processing concept (variable sampling and digital quantization) is proposed where a sample is coded with a single binary word. The code word corresponds to an information pair: a variable and adaptive sampling time and a coding angle associated to the signal. The basic principle is conceived around a distribution of the coded samples in amplitude and time (amplitude-time coding) along an adaptive coding curve associated to each coded/decoded sample of the original signal. A variable sampling rate requires a buffer, thus a delay, for transmission at a fixed rate. The transmitted signal so obtained is a transposition of the original speech signal and consequently its characteristics (bandwith, amplitude dynamic range) are modified. Some of the characteristics of the transmitted signal are ultimately used for the digital or even the analog transmission of the signal.