Satoshi Miki

Acoustic signal transform coding method and decoding method having a high efficiency envelope flattening method therein

An input acoustic signal is subjected to modified discrete cosine transform processing to obtain its spectrum characteristics. Linear prediction coefficients are derived from the input acoustic signal in a linear prediction coding analysis part, and the prediction coefficients are subjected to Fourier transform in a spectrum envelope calculation part to obtain the envelope of the spectrum characteristics of the input acoustic signal. In a normalization part the spectrum characteristics are normalized by the envelope thereof to obtain residual coefficients. Another normalization part normalizes the residual coefficients by a residual-coefficients envelope predicted in a residual-coefficients envelope calculation part, thereby obtaining fine structure coefficients, which are vector-quantized in a quantization part. A de-normalization part de-normalizes the quantized fine structure coefficients. The residual-coefficients envelope calculation part uses the reproduced residual coefficients to predict the envelope of residual coefficients of the subsequent frame.

High-quality audio-coding at less than 64 kbit/s by using transform-domain weighted interleave vector quantization (TwinVQ)

A new audio-coding method is proposed. This method is called transform-domain weighted interleave vector quantization (TwinVQ) and achieves high-quality reproduction at less than 64 kbit/s. The method is a transform coding using modified discrete cosine transform (MDCT). There are three novel techniques in this method: flattening of the MDCT coefficients by the spectrum of linear predictive coding (LPC) coefficients; interframe backward prediction for flattening the MDCT coefficients; and weighted interleave vector quantization. Subjective evaluation tests showed that the quality of the reproduction of TwinVQ exceeded that of an MPEG Layer II coder at the same bitrate.

Speech coding and decoding methods using adaptive and random code books

An excitation vector of the previous frame stored in an adaptive codebook is cut out with a selected pitch period. The excitation vector thus cut out is repeated until one frame is formed, by which a periodic component codevector is generated. An optimum pitch period is searched for so that distortion of a reconstructed speech obtained by exciting a linear predictive synthesis filter with the periodic component codevector is minimized. Thereafter, a random codevector selected from a random codebook is cut out with the optimum pitch period and is repeated until one frame is formed, by which a repetitious random codevector is generated. The random codebook is searched for a random codevector which minimizes the distortion of the reconstructed speech which is provided by exciting the synthesis filter with the repetitious random codevector.

Extension and complexity reduction of TwinVQ audio coder

This paper proposes two novel techniques for twinVQ (transform domain weighted interleave VQ) high-quality audio coding scheme for rates lower than 64 kbit/s. One is an extension of the weighted interleave technique to the time and input channel domains as well as the frequency domain. The other is an efficient representation scheme of the spectral envelope by means of a interpolated square root LPC (linear predictive coding) spectrum.

Coding of LSP Parameters Using Interframe Moving Average Prediction and Multi-Stage Vector Quantization

This paper proposes a new efficient line spectrum pair (LSP) vector quantization method which uses moving average (MA) interframe prediction of the parameters. This method has two advantages over auto-regressive prediction: the degradation of decoded parameters caused by bit errors affects only a few of the following frames, and codeword decoding can start at any frame. Using MA prediction, the final reconstructed LSP vector (quantized vector) of the current frame is represented as a linear combination of current and previous frame code vectors. In this paper we describe ways to achieve a more efficient LSP coder when each frame has four 10-ms subfraines. The spectral distance obtained by this method at 30 bits per 40 ms is better than that obtained using conventional multi-stage VQ without interframe correlation at 40 bits per 40 ms.

A design of transform coder for both speech and audio signals at 1 bit/sample

This paper proposes a speech and audio coder which operates at 1 bit/sample, namely an 8 kbit/s coder for 8 kHz sampling or a 16 kbit/s coder for 16 kHz sampling. The basic structure is inherited from a Twin VQ (transform domain weighted interleave vector quantization) high-quality audio coding scheme. A periodical component extraction scheme is newly added to the quantization of the MDCT coefficients. This scheme is found to be effective for reducing distortion and improving the robustness against channel errors. The qualities for music signals at 8 kbit/s are better than those of G.729 at the same bit rates, while they are worse for clean speech. The qualities at 16 kbit/s are comparable to or better than those of G.722 at 48 kbit/s.

Design of a pitch synchronous innovation CELP coder for mobile communications

This paper describes the design of a speech coder called pitch synchronous innovation CELP (PSI-CELP) for low hit-rate mobile communications. PSI-CELP is based on CELP, but has more adaptive excitation structures. In voiced frames, instead of conventional random excitation vectors, PSI-CELP converts even the random excitation vectors to have pitch periodicity by repeating stored random vectors as well as by using an adaptive codebook, in silent, unvoiced, and transient frames, the coder stops using the adaptive codebook and switches to fixed random codebooks. The PSI-CELP coder also implements novel structures and techniques: an FIR-type perceptual weighting filter using unquantized LPC parameters, a random codebook with a conjugate structure trained to be robust against channel errors, codebook search with delayed decision, a gain quantization with sloped amplitude, and a moving average prediction coding of LSP parameters, Our speech coder is implemented by DSP chips. Its coded speech quality at 3.6 kb/s with 2.0 kb/s redundancy is comparable to that of the Japanese full-rate VSELP coder at 6.7 kb/s with 4.5 kb/s redundancy. The basic structure of this PSI-CELP coder has been chosen as the Japanese half-rate speech codec for digital cellular telecommunications. >

Error-Protected TwinVQ Audio Coding at Less Than 64 kbit/s/ch

An error-protection method for transform-domain weighted interleave vector quantization (Twin VQ) audiocoding has been developed. Twin VQ achieves high-quality at less than 64 kbit/s/ch. About 90% of the Twin VQ encoded bits were found to be insensitive to channel errors: by using the bit-selectiveforward m o r con-ection (BS-FEC) method, morprotection k weighed to the sensitive bib. Computationul simulation showed that Twin VQ with BS-FEC maintains good p l i t y over high bit-error-rate channels. 13ris method is thus applicable to wireless systems, which have inferior transfer conditions.

A pitch synchronous innovation CELP (PSI-CELP) coder for 2-4 kbit/s

This paper proposes high-quality and low bit-rate (3.6 and 2.4 kbit/s) coders using a pitch synchronous innovation CELP (PSI-CELP) method or a phase adaptive PSI-CELP. PSI-CELP, which is used as the excitation structure of the half-rate codec for the standard of Japanese digital mobile telephony, is based on CELP but adds pitch synchronous innovation, which means that even random codevectors are adaptively converted to have pitch periodicity for voiced frames. Phase adaptive PSI-CELP makes not only the periodicity, like in PSI-CELP, but also the phase of random codevectors equal to those of an adaptive codevector. The subjective qualities of the 3.6- and 2.4-kbit/s coders exceed those of the 6.7-kbit/S VSELP coder, which is the full-rate codec for the standard of Japanese digital mobile telephony, and-the 4.8-kbit/s U.S. Federal Standard 1016 CELP coder, respectively, in the error-free condition.

Audio coding using transform-domain weighted interleave vector quantization (twin VQ)

Twin VQ (transform-domain weighted interleave vector quantization) is a method that encodes the wideband acoustic signal with a low bit rate. It is transform coding with a basic structure that transforms the input signal to the frequency domain by MDCT; vector quantization is applied after flattening. This encoding method has characteristic features such as weighted interleave vector quantization, normalization of the frequency characteristics by the linearly predicted spectrum, and interframe prediction in the frequency domain. Especially, high performance is realized for lower bit rates. Another feature is robustness against the error, since adaptive bit assignment is not applied.