A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

The present disclosure relates to processing a plurality of audio signals. A device receives the plurality of audio signals in the frequency domain and determining an overall attenuation multiplier based on the plurality of audio signals and an overall lookup table that relates decibel values to different overall attenuation multipliers. The device determines an attenuation vector comprising a plurality of bin-specific attenuation multipliers, each bin-specific attenuation multiplier respectively corresponding to a different frequency bin of the plurality of frequency bins. The device scales each bin-specific attenuation value in the attenuation vector with the overall attenuation multiplier, and edits each of the audio signals based on the scaled bin-specific attenuation values in the attenuation vector.

The present disclosure relates to processing a plurality of audio signals. A device receives the plurality of audio signals in the frequency domain and determining an overall attenuation multiplier based on the plurality of audio signals and an overall lookup table that relates decibel values to different overall attenuation multipliers. The device determines an attenuation vector comprising a plurality of bin-specific attenuation multipliers, each bin-specific attenuation multiplier respectively corresponding to a different frequency bin of the plurality of frequency bins. The device scales each bin-specific attenuation value in the attenuation vector with the overall attenuation multiplier, and edits each of the audio signals based on the scaled bin-specific attenuation values in the attenuation vector.

Audio encoder for encoding a multichannel signal is shown. The audio encoder includes a downmixer for downmixing the multichannel signal to obtain a downmix signal, a linear prediction domain core encoder for encoding the downmix signal, wherein the downmix signal has a low band and a high band, wherein the linear prediction domain core encoder is configured to apply a bandwidth extension processing for parametrically encoding the high band, a filterbank for generating a spectral representation of the multichannel signal, and a joint multichannel encoder configured to process the spectral representation including the low band and the high band of the multichannel signal to generate multichannel information.

A schematic block diagram of a decoder for decoding an encoded audio signal is shown. The decoder includes an adaptive spectrum-time converter and an overlap-add-processor. The adaptive spectrum-time converter converts successive blocks of spectral values into successive blocks of time values, e.g. via a frequency-to-time transform. Furthermore, the adaptive spectrum-time converter receives a control information and switches, in response to the control information, between transform kernels of a first group of transform kernels including one or more transform kernels having different symmetries at sides of a kernel, and a second group of transform kernels including one or more transform kernels having the same symmetries at sides of a transform kernel. Moreover, the overlap-add-processor overlaps and adds the successive blocks of time values to obtain decoded audio values, which may be a decoded audio signal.

An audio quantizer for quantizing a plurality of audio information items has: a first stage vector quantizer for quantizing the plurality of audio information items to determine a first stage vector quantization result and a plurality of intermediate quantized items corresponding to the first stage vector quantization result; a residual item determiner for calculating a plurality of residual items from the plurality of intermediate quantized items and the plurality of audio information items; and a second stage vector quantizer for quantizing the plurality of residual items to obtain a second stage vector quantization result, wherein the first stage vector quantization result and the second stage vector quantization result are a quantized representation of the plurality of audio information items.

Vector Quantizer and method therein for efficient vector quantization, e.g. in a transform audio codec. The method comprises comparing an input target vector s with a plurality of centroids, each centroid representing a respective class of codevectors in a codebook. Further, a starting point for a search related to the input target vector in the codebook is determined, based on the result of the comparison. The codevectors in the codebook are sorted according to a distortion measure reflecting the distance between each codevector and the centroids of the classes. The Vector Quantizer and method enables that the class of codevectors comprising the most probable candidate codevectors in regard of the input vector s may be searched first.

Vector Quantizer and method therein for efficient vector quantization, e.g. in a transform audio codec. The method comprises comparing an input target vector s with a plurality of centroids, each centroid representing a respective class of codevectors in a codebook. Further, a starting point for a search related to the input target vector in the codebook is determined, based on the result of the comparison. The codevectors in the codebook are sorted according to a distortion measure reflecting the distance between each codevector and the centroids of the classes. The Vector Quantizer and method enables that the class of codevectors comprising the most probable candidate codevectors in regard of the input vector s may be searched first.

Described herein are methods, apparatus and computer products for decoding an encoded MPEG-D USAC bitstream. Described herein are such methods, apparatus and computer products that reduce a computational complexity.