Embodiments of the present invention relate to audio object extraction. A method for audio object extraction from audio content of a format based on a plurality of channels is disclosed. The method comprises applying audio object extraction on individual frames of the audio content at least partially based on frequency spectral similarities among the plurality of channels. The method further comprises performing audio object composition across the frames of the audio content, based on the audio object extraction on the individual frames, to generate a track of at least one audio object. Corresponding system and computer program product are also disclosed.

The present invention reduces encoding distortion in frequency domain encoding compared to conventional techniques, and obtains LSP parameters that correspond to quantized LSP parameters for the preceding frame and are to be used in time domain encoding from coefficients equivalent to linear prediction coefficients resulting from frequency domain encoding. When p is an integer equal to or greater than 1, a linear prediction coefficient sequence which is obtained by linear prediction analysis of audio signals in a predetermined time segment is represented as a[1], a[2], . . . , a[p], and ω[1], ω[2], . . . , ω[p] are a frequency domain parameter sequence derived from the linear prediction coefficient sequence a[1], a[2], . . . , a[p], an LSP linear transformation unit (300) determines the value of each converted frequency domain parameter ˜ω[i] (i=1, 2, . . . , p) in a converted frequency domain parameter sequence ˜ω[1], ˜ω[2], . . . , ˜ω[p] using the frequency domain parameter sequence ω[1], ω[2], . . . , ω[p] as input, through linear transformation which is based on the relationship of values between ω[i] and one or more frequency domain parameters adjacent to ω[i].

The present invention reduces encoding distortion in frequency domain encoding compared to conventional techniques, and obtains LSP parameters that correspond to quantized LSP parameters for the preceding frame and are to be used in time domain encoding from coefficients equivalent to linear prediction coefficients resulting from frequency domain encoding. When p is an integer equal to or greater than 1, a linear prediction coefficient sequence which is obtained by linear prediction analysis of audio signals in a predetermined time segment is represented as a[1], a[2], . . . , a[p], and ω[1], ω[2], . . . , ω[p] are a frequency domain parameter sequence derived from the linear prediction coefficient sequence a[1], a[2], . . . , a[p], an LSP linear transformation unit (300) determines the value of each converted frequency domain parameter ˜ω[i] (i=1, 2, . . . , p) in a converted frequency domain parameter sequence ˜ω[1], ˜ω[2], . . . , ˜ω[p] using the frequency domain parameter sequence ω[1], ω[2], . . . , ω[p] as input, through linear transformation which is based on the relationship of values between ω[i] and one or more frequency domain parameters adjacent to ω[i].

In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis. The long-term prediction mode is tailored to where the generally periodic component of the speech is generally not stationary or less than completely periodic and requires greater frequency of updates from the adaptive codebook to achieve a desired perceptual quality of the reproduced speech under a long-term predictive procedure.

In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis. The long-term prediction mode is tailored to where the generally periodic component of the speech is generally not stationary or less than completely periodic and requires greater frequency of updates from the adaptive codebook to achieve a desired perceptual quality of the reproduced speech under a long-term predictive procedure.

In a CELP-type speech coding apparatus, switching between an orthogonal search of a fixed codebook and a non-orthogonal search is performed in a practical and effective manner. The CELP-type speech coding apparatus includes a parameter quantizer that selects an adaptive codebook vector and a fixed codebook vector so as to minimize an error between a synthesized speech signal and an input speech signal. The parameter quantizer includes a fixed codebook searcher that switches between the orthogonal fixed codebook search and the non-orthogonal fixed codebook search based on a correlation value between a target vector for the fixed codebook search and the adaptive codebook vector obtained as a result of a synthesis filtering process.

In a CELP-type speech coding apparatus, switching between an orthogonal search of a fixed codebook and a non-orthogonal search is performed in a practical and effective manner. The CELP-type speech coding apparatus includes a parameter quantizer that selects an adaptive codebook vector and a fixed codebook vector so as to minimize an error between a synthesized speech signal and an input speech signal. The parameter quantizer includes a fixed codebook searcher that switches between the orthogonal fixed codebook search and the non-orthogonal fixed codebook search based on a correlation value between a target vector for the fixed codebook search and the adaptive codebook vector obtained as a result of a synthesis filtering process.

An encoding method, a decoding method, an encoding apparatus, a decoding apparatus, a transmitter, a receiver, and a communications system, where the encoding method includes dividing a to-be-encoded time-domain signal into a low band signal and a high band signal, performing encoding on the low band signal to obtain a low frequency encoding parameter, performing encoding on the high band signal to obtain a high frequency encoding parameter, obtaining a synthesized high band signal; performing short-time post-filtering processing on the synthesized high band signal to obtain a short-time filtering signal, and calculating a high frequency gain based on the high band signal and the short-time filtering signal.

An encoding method, a decoding method, an encoding apparatus, a decoding apparatus, a transmitter, a receiver, and a communications system, where the encoding method includes dividing a to-be-encoded time-domain signal into a low band signal and a high band signal, performing encoding on the low band signal to obtain a low frequency encoding parameter, performing encoding on the high band signal to obtain a high frequency encoding parameter, obtaining a synthesized high band signal; performing short-time post-filtering processing on the synthesized high band signal to obtain a short-time filtering signal, and calculating a high frequency gain based on the high band signal and the short-time filtering signal.

A device includes an encoder and a transmitter. The encoder is configured to generate a first high-band portion of a first signal based on a left signal and a right signal. The encoder is also configured to generate a set of adjustment gain parameters based on a high-band non-reference signal. The high-band non-reference signal corresponds to one of a left high-band portion of the left signal or a right high-band portion of the right signal as a high-band non-reference signal. The transmitter is configured to transmit information corresponding to the first high-band portion of the first signal. The transmitter is also configured to transmit the set of adjustment gain parameters corresponding to the high-band non-reference signal.