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Chen, Juin-Hwey

Publications and source records attributed to Chen, Juin-Hwey.

Vector adaptive predictive coder for speech and audio

A real-time vector adaptive predictive coder which approximates each vector of K speech samples by using each of M fixed vectors in a first codebook to excite a time-varying synthesis filter and picking the vector that minimizes distortion. Predictive analysis for each frame determines parameters used for computing from vectors in the first codebook zero-state response vectors that are stored at the same address (index) in a second codebook. Encoding of input speech vectors s.sub.n is then carried out using the second codebook. When the vector that minimizes distortion is found, its index is transmitted to a decoder which has a codebook identical to the first codebook of the decoder. There the index is used to read out a vector that is used to synthesize an output speech vector s.sub.n. The parameters used in the encoder are quantized, for example by using a table, and the indices are transmitted to the decoder where they are decoded to specify transfer characteristics of filters used in producing the vector s.sub.n from the receiver codebook vector selected by the vector index transmitted.

Chen, Juin-Hwey

Vector Adaptive/Predictive Encoding Of Speech

Vector adaptive/predictive technique for digital encoding of speech signals yields decoded speech of very good quality after transmission at coding rate of 9.6 kb/s and of reasonably good quality at 4.8 kb/s. Requires 3 to 4 million multiplications and additions per second. Combines advantages of adaptive/predictive coding, and code-excited linear prediction, yielding speech of high quality but requires 600 million multiplications and additions per second at encoding rate of 4.8 kb/s. Vector adaptive/predictive coding technique bridges gaps in performance and complexity between adaptive/predictive coding and code-excited linear prediction.

Chen, Juin-Hwey

Speech coding at 4800 bps for mobile satellite communications

A speech compression project has recently been completed to develop a speech coding algorithm suitable for operation in a mobile satellite environment aimed at providing telephone quality natural speech at 4.8 kbps. The work has resulted in two alternative techniques which achieve reasonably good communications quality at 4.8 kbps while tolerating vehicle noise and rather severe channel impairments. The algorithms are embodied in a compact self-contained prototype consisting of two AT and T 32-bit floating-point DSP32 digital signal processors (DSP). A Motorola 68HC11 microcomputer chip serves as the board controller and interface handler. On a wirewrapped card, the prototype's circuit footprint amounts to only 200 sq cm, and consumes about 9 watts of power.

Gersho, Allen

A prototype 4800 bps voice terminal for the mobile satellite experiment

The authors report the final stage of a research project for the NASA mobile satellite experiment (MSAT-X). The work has resulted in two alternative coding techniques which have achieved good communications quality at 4.8 kb/s while tolerating vehicle noise, fading, and a fairly high-error-rate, bursty channel. The authors have constructed a circuit board based on three AT&T 32-bit floating-point DSP32 digital signal processors (DSP) and providing both excess processing power and considerable flexibility for handling different variations of the algorithms. A Motorola 68HC11 microcomputer chip serves as the board controller and interface handler. Mounted on a wirewrapped card, the prototype's circuit footprint amounts to only 225 cm2 and consumes about 13 W of power. A description is given of the architecture, signal processing and communication functions, circuitry, and firmware of the prototype. Issues such as synchronization, framing, error control, and modem interface are also addressed.

Chan, Wai-Yip