The present disclosure provides methods, devices and computer program products for encoding and decoding a stereo audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and a discrete representation of the stereo audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

An encoder operable to filter audio signals into a plurality of frequency band components, generate quantized digital components for each band, identify a potential for pre-echo events within the generated quantized digital components, generate an approximate signal by decoding the quantized digital components using inverse pulse code modulation, generate an error signal by comparing the approximate signal with the sampled audio signal, and process the error signal and quantized digital components. The encoder operable to process the error signal by processing delayed audio signals and Q band values, determining the potential for pre-echo events from the Q band values, and determining scale factors and MDCT block sizes for the potential for pre-echo events. The encoder operable to transform the error signal into high resolution frequency components using the MDCT block sizes, quantize the scale factors and frequency components, and encode the quantized lines, block sizes, and quantized scale factors for inclusion in the bitstream.

An audio signal encoding method and apparatus, and an audio signal decoding method and apparatus, are described. The encoding method includes obtaining a target frequency-domain coefficient of a current frame and a reference target frequency-domain coefficient of the current frame. The encoding method further includes calculating a cost function based on the target frequency-domain coefficient and the reference target frequency-domain coefficient of the current frame, where the cost function is for determining whether to perform long-term prediction (LTP) processing on the current frame during encoding of the target frequency-domain coefficient of the current frame. Additionally, the method includes encoding the target frequency-domain coefficient of the current frame based on the cost function.

A method performed by an encoder. The method comprises determining envelope representation residual coefficients as first compressed envelope representation coefficients subtracted from the input envelope representation coefficients. The method comprises transforming the envelope representation residual coefficients into a warped domain so as to obtain transformed envelope representation residual coefficients. The method comprises applying, at least one of a plurality of gain-shape coding schemes on the transformed envelope representation residual coefficients in order to achieve gain-shape coded envelope representation residual coefficients, where the plurality of gain-shape coding schemes have mutually different trade-offs in one or more of gain resolution and shape resolution for one or more of the transformed envelope representation residual coefficients. The method comprises transmitting, over a communication channel to a decoder, a representation of the first compressed envelope representation coefficients, the gain-shape coded envelope representation residual coefficients, and information on the at least one applied gain-shape coding scheme.

A spectrum filler for filling non-coded residual sub-vectors of a transform coded audio signal includes a sub-vector compressor configured to compress actually coded residual sub-vectors. A sub-vector rejecter is configured to reject compressed residual sub-vectors that do not fulfill a predetermined sparseness criterion. A sub-vector collector is configured to concatenate the remaining compressed residual sub-vectors to form a first virtual codebook. A coefficient combiner is configured to combine pairs of coefficients of the first virtual codebook to form a second virtual codebook. A sub-vector filler is configured to fill non-coded residual sub-vectors below a predetermined frequency with coefficients from the first virtual codebook, and to fill non-coded residual sub-vectors above the predetermined frequency with coefficients from the second virtual codebook.

An apparatus for encoding an audio or image signal, includes: a controllable windower for windowing the audio or image signal to provide the sequence of blocks of windowed samples; a converter for converting the sequence of blocks of windowed samples into a spectral representation including a sequence of frames of spectral values; a transient location detector for identifying a location of a transient within a transient look-ahead region of a frame; and a controller for controlling the controllable windower to apply a specific window having a specified overlap length to the audio or image signal in response to an identified location of the transient, wherein the controller is configured to select the specific window from a group of at least three windows, wherein the specific window is selected based on the transient location.

A single-bit audio stream can be converted to a modified single-bit audio stream with a constant edge rate while maintaining a modulation index of the original audio stream using direct mapping. With direct mapping, a pre-filter bank may be combined with a multi-bit symbol mapper to select symbols for the modified audio stream with a constant edge rate per symbol and the same modulation index as the original audio stream. The output of the pre-filter bank may be an audio stream with no consecutive full-scale symbols. Using the output of the pre-filter bank, a multi-bit symbol mapper may use the symbol selector to output a symbol with a constant edge rate per symbol and the same modulation index as the original signal. The symbols may be converted to an analog signal for reproduction of audio content using a transducer.

Disclosed are methods of encoding and decoding an audio signal, and an encoder and a decoder for performing the methods. The method of encoding an audio signal includes identifying an input signal corresponding to a low frequency band of the audio signal, windowing the input signal, generating a first latent vector by inputting the windowed input signal to a first encoding model, transforming the windowed input signal into a frequency domain, generating a second latent vector by inputting the transformed input signal to a second encoding model, generating a final latent vector by combining the first latent vector and the second latent vector, and generating a bitstream corresponding to the final latent vector.

The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively. The method comprises determining a model parameter (613) of a signal model; determining a prediction coefficient to be applied to a previous sample (614) of a first decoded subband signals derived from the first subband signal, based on the signal model, based on the model parameter (613) and based on the analysis filterbank (612); wherein a time slot of the previous sample (614) is prior to a time slot of the first sample (615); and determining an estimate of the first sample (615) by applying the prediction coefficient to the previous sample (614).

An information encoder for encoding an information signal includes: a converter for converting the linear prediction coefficients of the predictive polynomial A(z) to frequency values f.sub.1 . . . f.sub.n of a spectral frequency representation of the predictive polynomial A(z), wherein the converter is configured to determine the frequency values f.sub.1 . . . f.sub.n by analyzing a pair of polynomials P(z) and Q(z) being defined as $P\left(z\right)=A\left(z\right)+z^{-m-l}A\left(z^{-1}\right)\mathrm{and}$$Q\left(z\right)=A\left(z\right)-z^{-m-l}A\left(z^{-1}\right),$
wherein m is