University of California

Microsoft Technology Licensing, LLC

Bell Telephone Laboratories

Voicecraft, Inc.

Other

Nokia Mobile Phones Limited

Allen Gersho

Vladimir Cuperman

Kazuhito Koishida

Ajit V Rao

Eyal Shlomot

Sassan Ahmadi

Hosam Adel Khalil

Victor Bernard Lawrence

Tian Wang

Hong Wang

Fenghua Liu

Oded Gottesman

Chunyan Li

Kenneth Rose

Richard D Gitlin

David Joel Goodman

Edmond Y Ho

Amir H Majidimehr

Tong L Lim

Tung-Chiang Yang

Juin-Hwey Chen

Anil Wamanrao Ubale

Sangeeta Ramakrishnan

Amitava Das

Jan Linden

Grant Davidson

Ryan Heidari

An enhanced analysis-by-synthesis waveform interpolative speech coder able to operate at 2.8 kbps. Novel features include dual-predictive analysis-by-synthesis quantization of the slowly-evolving waveform, efficient parametrization of the rapidly-evolving waveform magnitude, and analysis-by-synthesis vector quantization of the rapidly evolving waveform parameter. Subjective quality tests indicate that it exceeds G.723.1 at 5.3 kbps, and of G.723.1 at 6.3 kbps.

REW parametric vector quantization and dual-predictive SEW vector quantization for waveform interpolative coding

An enhanced low-bit rate parametric voice coder that groups a number of frames from an underlying frame-based vocoder, such as MELP, into a superframe structure. Parameters are extracted from the group of underlying frames and quantized into the superframe which allows the bit rate of the underlying coding to be reduced without increasing the distortion. The speech data coded in the superframe structure can then be directly synthesized to speech or may be transcoded to a format so that an underlying frame-based vocoder performs the synthesis. The superframe structure includes additional error detection and correction data to reduce the distortion caused by the communication of bit errors.

LPC-harmonic vocoder with superframe structure

A method and system of performing postfiltering in the frequency domain to improve the quality of a speech signal, especially for synthesized speech resulting from codecs of low bit-rate, is provided. The method comprises LPC tilt computation and compensation methods and modules, a formant filter gain computation method and module, and an anti-aliasing method and module. The formant filter gain calculation employs an LPC representation, an all-pole modeling, a non-linear transformation and a phase computation. The LPC used for deriving the postfilter may be transmitted from an encoder or may be estimated from a synthesized or other speech signal in a decoder or receiver. The invention may be implemented in a linked decoder and encoder. A separate LPC evaluation unit that is responsible for processing and or deriving the LPC may be implemented within the invention.

Frequency domain postfiltering for quality enhancement of coded speech

The present invention provides a transform coding method efficient for music signals that is suitable for use in a hybrid codec, whereby a common Linear Predictive (LP) synthesis filter is employed for both speech and music signals. The LP synthesis filter switches between a speech excitation generator and a transform excitation generator, in accordance with the coding of a speech or music signal, respectively. For coding speech signals, the conventional CELP technique may be used, while a novel asymmetrical overlap-add transform technique is applied for coding music signals. In performing the common LP synthesis filtering, interpolation of the LP coefficients is conducted for signals in overlap-add operation regions. The invention enables smooth transitions when the decoder switches between speech and music decoding modes.

Method for coding speech and music signals

A variable bit rate coder, and an associated method, for encoding a frame of speech, such as frames of data generated during operation of a communication station operable in a cellular communication system. Selection of the coding rate is made responsive to indicia of actual coding performance of a coder at more than one coding rate.

Variable bit rate coder, and associated method, for a communication station operable in a communication system

A method and apparatus for encoding speech for communication to a decoder for reproduction of the speech where the speech signal is classified into steady state voiced (harmonic), stationary unvoiced, and &ldquo;transitory&rdquo; or &ldquo;transition&rdquo; speech, and a particular type of coding scheme is used for each class. Harmonic coding is used for steady state voiced speech, &ldquo;noise-like&rdquo; coding is used for stationary unvoiced speech, and a special coding mode is used for transition speech, designed to capture the location, the structure, and the strength of the local time events that characterize the transition portions of the speech. The compression schemes can be applied to the speech signal or to the LP residual signal.

Method and apparatus for hybrid coding of speech at 4KBPS having phase alignment between mode-switched frames

A speech coder and a method for speech coding wherein the speech signal is represented by an excitation signal applied to a synthesis filter. The speech is partitioned into frames and subframes. A classifier identifies which of several categories the speech frame belongs to, and a different coding method is applied to represent the excitation for each category. For some categories, one or more windows are identified for the frame where all or most of the excitation signal samples are assigned by a coding scheme. Performance is enhanced by coding the important segments of the excitation more accurately. The window locations are determined from a linear prediction residual by identifying peaks of the smoothed residual energy contour. The method adjusts the frame and subframe boundaries so that each window is located entirely within a modified subframe or frame. This eliminates the artificial restriction incurred when coding a frame or subframe in isolation, without regard for the local behavior of the speech signal across frame or subframe boundaries.

Adaptive windows for analysis-by-synthesis CELP-type speech coding

Method and apparatus for hybrid coding of speech at 4kbps

Vectors associated with multiple sources are quantized with a codebook. A mapping function associates each source with a subset of codevectors in the codebook, each subset being defined prior to quantization. Each vector from a source is matched to a codevector in the subset corresponding to the source. The universal codebook is designed by first providing a codebook and a mapping function. Using the codebook, the mapping function is updated by identifying subsets that reduce distortion when vectors from a source are quantized using a corresponding subset.

Universal codebook vector quantization with constrained storage

A variable dimension vector quantization method that uses a single 'universal' codebook. The method can be given the interpretation of sampling full-dimensioned codevectors in the universal codebook and generating subcodevectors of the same dimension as input data subvector, which dimension may vary in time. A subcodevector is selected from the codebook to have minimum distortion between it and the input data subvector. The subcodevector with minimum distortion corresponds to the representative, full-dimensioned codevector in the codebook. The codebook is designed by inverse sampling of training subvectors to obtain full-dimension vectors, then iteratively clustering the training set until a stable centroid vector is obtained.

Variable dimension vector quantization

A method of digitally compressing speech and music by use of multiple band ('multiband') fixed excitations stored in codebooks. The use of multiband fixed excitations, along with a coupling method for interconnecting the excitation codebooks and adaptive codebooks and for generating the composite excitation signal, improve the long-term and short-term prediction, and the use of voice-music classification allows the coding structure to be adapted to the statistical character of the audio signal.

Method and apparatus for efficient multiband celp wideband speech and

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.

Vector adaptive predictive coder for speech and audio

Growing community of inventors

Goleta, CA, United States of America

Allen Gersho