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 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 comprising of their amplitude and frequency variations 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 comprising of their amplitude and frequency variations 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 signal processing device, method, and program that may obtain audio at a higher audio quality when decoding an audio signal. An envelope information generating unit generates envelope information representing an envelope form of high frequency components of an audio signal to be encoded. A sine wave information generating unit extracts a sine wave signal from the high frequency components of the audio signal, and generates a sine wave information representing an emergence start position of the sine wave signal. An encoding stream generating unit multiplexes the envelope information, the sine wave information, and low frequency components of the audio signal that have been encoded, and outputs an encoding stream obtained as the result. The high frequency components included in the sine wave signal may be predicted at a higher accuracy from the envelope information and the sine wave information at the receiving side of the encoding stream.

A signal processing device, method, and program that may obtain audio at a higher audio quality when decoding an audio signal. An envelope information generating unit generates envelope information representing an envelope form of high frequency components of an audio signal to be encoded. A sine wave information generating unit extracts a sine wave signal from the high frequency components of the audio signal, and generates a sine wave information representing an emergence start position of the sine wave signal. An encoding stream generating unit multiplexes the envelope information, the sine wave information, and low frequency components of the audio signal that have been encoded, and outputs an encoding stream obtained as the result. The high frequency components included in the sine wave signal may be predicted at a higher accuracy from the envelope information and the sine wave information at the receiving side of the encoding stream.

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 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 present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilising high frequency reconstruction (HFR). It utilises a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR input.

The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilising high frequency reconstruction (HFR). It utilises a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR input.