The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Audio encoder and decoder

The present invention relates to a method and device for improving concealment of frame erasure caused by frames of an encoded sound signal erased during transmission from an encoder (106) to a decoder (110), and for accelerating recovery of the decoder after non erased frames of the encoded sound signal have been received. For that purpose, concealment/recovery parameters are determined in the encoder or decoder. When determined in the encoder (106), the concealment/recovery parameters are transmitted to the decoder (110). In the decoder, erasure frame concealment and decoder recovery is conducted in response to the concealment/recovery parameters. The concealment/recovery parameters may be selected from the group consisting of: a signal classification parameter, an energy information parameter and a phase information parameter. The determination of the concealment/recovery parameters comprises classifying the successive frames of the encoded sound signal as unvoiced, unvoiced transition, voiced transition, voiced, or onset, and this classification is determined on the basis of at least a part of the following parameters: a normalized correlation parameter, a spectral tilt parameter, a signal-to-noise ratio parameter, a pitch stability parameter, a relative frame energy parameter, and a zero crossing parameter.

Method and device for efficient frame erasure concealment in linear predictive based speech codecs

A method and device for concealing frame erasures caused by frames of an encoded sound signal erased during transmission from an encoder to a decoder and for recovery of the decoder after frame erasures comprise, in the encoder, determining concealment/recovery parameters including at least phase information related to frames of the encoded sound signal. The concealment/recovery parameters determined in the encoder are transmitted to the decoder and, in the decoder, frame erasure concealment is conducted in response to the received concealment/recovery parameters. The frame erasure concealment comprises resynchronizing, in response to the received phase information, the erasure-concealed frames with corresponding frames of the sound signal encoded at the encoder. When no concealment/recovery parameters are transmitted to the decoder, a phase information of each frame of the encoded sound signal that has been erased during transmission from the encoder to the decoder is estimated in the decoder. Also, frame erasure concealment is conducted in the decoder in response to the estimated phase information, wherein the frame erasure concealment comprises resynchronizing, in response to the estimated phase information, each erasure-concealed frame with a corresponding frame of the sound signal encoded at the encoder.

Method and device for efficient frame erasure concealment in speech codecs

Pitch determination using speech classification and prior pitch estimation

A random code vector reading section and a random codebook of a conventional CELP type speech coder/decoder are respectively replaced with an oscillator for outputting different vector streams in accordance with values of input seeds, and a seed storage section for storing a plurality of seeds. This makes it unnecessary to store fixed vectors as they are in a fixed codebook (ROM), thereby considerably reducing the memory capacity.

Excitation vector generator, speech coder and speech decoder

A codebook is searched in view of encoding a sound signal. This codebook consists of a set of codevectors each of 40 positions and comprising N non-zero-amplitude pulses assignable to predetermined valid positions. To reduce the search complexity, a depth-first search is used which involves a tree structure with levels ordered from 1 through M. A path-building operation takes place at each level whereby a candidate path from the previous level is extended by choosing a predetermined number of new pulses and selecting valid positions for said new pulses in accordance with a given pulse-order rule and a given selection criterion. A path originated at the first level and extended by the path-building operations of subsequent levels determines the respective positions of the N non-zero-amplitude pulse of a candidate codevector. Use of a signal-based pulse-position likelihood estimate during the first few levels enable initial pulse-screening to start the search on favorable conditions. A selection criterion based on maximizing a ratio is used to assess the progress and to choose the best one among competing candidate codevectors.

Depth-first algebraic-codebook search for fast coding of speech

Les connaissances sur les reseaux reguliers de points trouvent leurs applications aussi bien dans la conception d’un quantificateur de formes d’ ondes que dans la selection d’un jeu de signaux pour la modulation. Dans le domaine du codage de la parole, une nouvelle technique recemment proposee [1], la quantification vectorielle algebrique spherique, utilise des sous-ensembles spheriques extraits de reseaux reguliers de points. L’article concerne la quantification vectorielle algebrique spherique basee sur le reseau de Gosset de dimension huit. Un algorithme optimal rapide est decrit en detail qui permet de determiner le plus proche vecteur parmi une collection finie spherique du reseau de Gosset et de determiner le rang de ce plus proche voisin dans la collection. Les performances de ce quantificateur pour le bruit blanc gaussien sont comparees aux limites predites par la theorie de l’information. La comparaison des approches algebrique et statistique amene a envisager la conception de quantificateurs hybrides integrant ces deux techniques.

Algorithme de quantification vectorielle sphérique à partir du réseau de Gosset d’ordre 8

A method and device eradicate the occasional instability inherent in analysis-by-synthesis speech/audio codecs and caused in particular by channel errors during transmission of highly periodic signals such as high-frequency sine waves. Analysis-by-synthesis techniques involve production, in response to the speech/audio signal and at regular time intervals called frames, of (a) a set of spectral parameters for use in driving a synthesis filter in view of synthesizing the speech/audio signal, and (b) a pitch gain for constructing a past-excitation-signal component supplied to the synthesis filter. In accordance with the instability eradication method, the first step consists of detecting a set of conditions including (i) a resonance condition assessed from the spectral parameters, (ii) a duration condition detected when the resonance condition has prevailed for at least the M most recent frames, M being an integer greater than 1, and (iii) a gain condition which evidences consistently-high values of the pitch gain in the N most recent frames, N being an integer greater than 1. To eradicate the occasional instability, the pitch gain is reduced to a value lower than a given threshold whenever these three conditions are detected.

Method and device for eradicating instability due to periodic signals in analysis-by-synthesis speech codecs

In a method and device for encoding a speech and/or audio signal in view of enhancing perceptual quality, the time-domain speech and/or audio signal is broken down into a succession of overlapping finite-duration speech and/or audio signal segments, and a first linear transform is applied to each of these speech and/or audio signal segments to obtain short-term spectral components. Then, a non-linear transform is applied to the short-term spectral components in order to produce warped spectral components. A second linear transform is applied to the warped spectral components to obtain a time-domain signal interval, the time-domain signal interval is multiplied by a time window to produce a windowed-signal interval, and the successive overlapping windowed-signal intervals corresponding to the successive overlapping finite-duration speech and/or audio signal segments are added to obtain a pre-processed signal. This pre-processed signal is encoded to produce the encoded speech and/or audio signal. Corresponding speech/audio decoding method and device are also provided.

Speech/audio coding with non-linear spectral-amplitude transformation

The disclosure herein describes a method and a system for speech detection on PCM multiplexed voice channels; for each channel, a decision is reached every M samples regarding the channel activity; in addition, the nature of speech is detected as: voiced (compact or non-compact) or unvoiced (fricative or non-fricative) when the channel is active; pure silence, white noise or echo when the channel is inactive. The decision is based on the joint value of the amplitude, zero crossing of the signal and zero crossing of the signal derivative.

Jean-Pierre Adoul

Papers

Depth-first algebraic-codebook search for fast coding of speech

Algorithme de quantification vectorielle sphérique à partir du réseau de Gosset d’ordre 8

Method and device for eradicating instability due to periodic signals in analysis-by-synthesis speech codecs

Speech/audio coding with non-linear spectral-amplitude transformation

Method and apparatus for speech detection of PCM multiplexed voice channels