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).

Systems and techniques are described for audio coding. An audio system receives feature(s) corresponding an audio signal, for example from an encoder and/or a speech synthesis engine. The audio system generates an excitation signal, such as a harmonic signal and/or a noise signal, based on the feature(s). The audio system uses a filterbank to generate band-specific signals from the excitation signal. The band-specific signals correspond to frequency bands. The audio system inputs the feature(s) into a machine learning (ML) filter estimator to generate parameter(s) associated with linear filter(s). The audio system inputs the feature(s) into a voicing estimator to generate gain value(s). The audio system generates an output audio signal based on modification of the band-specific signals, application of the linear filter(s) according to the parameter(s), and amplification using the gain amplifier(s) according to the gain value(s).

Certain features relate to a telecommunications system with a modular frequency combiner combining multiple received signals at different frequency bands without using frequency-dependent multiplexers. The frequency combiner can include adjustable tuning elements for adjusting various signal-processing parameters of the frequency combiner while the frequency combiner is in the telecommunications system. For example, adjustable tuning elements can adjust the phases of phase shifters of each RF path so that the RF paths are matched for combining the received signals and outputting them through an output port. The adjustable tuning elements can also adjust the electrical length or physical length of the transmission lines that carry the received signals. The adjustable tuning elements can be adjusted manually or automatically while the frequency combiner is deployed in the field in the telecommunications system.

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).

The invention concerns an audio signal encoding method comprising the steps of: collecting the audio signal samples, determining sinusoidal components in subsequent frames, estimation of amplitudes and frequencies of the components for each frame, merging thus obtained pairs into sinusoidal trajectories, splitting particular trajectories into segments, transforming particular trajectories to the frequency domain by means of a digital transform performed on segments longer than the frame duration, quantization and selection of transform coefficients in the segments, entropy encoding, and outputting the quantized coefficients as output data. The method is characterized in that the length of the segments into which each trajectory is split is individually adjusted in time for each trajectory.

The invention concerns an audio signal encoding method comprising the steps of: collecting the audio signal samples, determining sinusoidal components in subsequent frames, estimation of amplitudes and frequencies of the components for each frame, merging thus obtained pairs into sinusoidal trajectories, splitting particular trajectories into segments, transforming particular trajectories to the frequency domain by means of a digital transform performed on segments longer than the frame duration, quantization and selection of transform coefficients in the segments, entropy encoding, and outputting the quantized coefficients as output data. The method is characterized in that the length of the segments into which each trajectory is split is individually adjusted in time for each trajectory.

A method includes determining, at an encoder, phase adjustment parameters based on a high-band residual signal. The method also includes inserting the phase adjustment parameters into an encoded version of the audio signal to enable phase adjustment during reconstruction of the audio signal from the encoded version of the audio signal.

A method includes determining, at an encoder, phase adjustment parameters based on a high-band residual signal. The method also includes inserting the phase adjustment parameters into an encoded version of the audio signal to enable phase adjustment during reconstruction of the audio signal from the encoded version of the audio signal.

A method performed in an audio decoder for reconstructing an original audio signal having a lowband portion and a highband portion is disclosed. The method includes receiving an encoded audio signal and extracting reconstruction parameters from the encoded audio signal. The method further includes decoding the encoded audio signal with a core audio decoder to obtain a decoded lowband portion and regenerating the highband portion based at least in part on a cross over frequency and the decoded lowband portion to obtain a regenerated highband portion. The method also includes creating a synthetic sinusoid with a level based at least in part on a spectral envelope value for the particular subband and a noise floor value for the particular subband and adding the synthetic sinusoid to the regenerated highband portion in the particular frequency band specified by the location information. Finally, the method includes combining the lowband portion and the regenerated highband portion to obtain a full bandwidth audio signal.

A method performed in an audio decoder for reconstructing an original audio signal having a lowband portion and a highband portion is disclosed. The method includes receiving an encoded audio signal and extracting reconstruction parameters from the encoded audio signal. The method further includes decoding the encoded audio signal with a core audio decoder to obtain a decoded lowband portion and regenerating the highband portion based at least in part on a cross over frequency and the decoded lowband portion to obtain a regenerated highband portion. The method also includes creating a synthetic sinusoid with a level based at least in part on a spectral envelope value for the particular subband and a noise floor value for the particular subband and adding the synthetic sinusoid to the regenerated highband portion in the particular frequency band specified by the location information. Finally, the method includes combining the lowband portion and the regenerated highband portion to obtain a full bandwidth audio signal.