A voice coder configured to resolve periodic and aperiodic components of spectra is disclosed. The method of voice coding includes parsing the speech signal into a plurality of speech frames; for each of the plurality of speech frames: (a) generating the spectra for the speech frame, (b) parsing the spectra of the speech frame into a plurality of sub-bands, (c) transforming each of the plurality of sub-bands into a time-domain envelope signal, and (d) generating a plurality of sub-band voicing factors, wherein each sub-band voicing factor indicates the harmonicity of one of the plurality of sub-bands, and each sub-band voicing factor is based on the periodicity of one of said time-domain envelope signals associated with one of the plurality of sub-bands. The voice coder may regenerate the speech signal by generating a plurality of recomposed frames, each recomposed frame being based on: (a) the spectra for one of said plurality of speech frames, and (b) the sub-band voicing factors associated with the plurality of sub-bands for one of said plurality of speech frames; and then generating a recomposed speech signal from the plurality of recomposed frames.

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).

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 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 which make use of linear prediction in combination with a filterbank. A method for estimating a sample of a subband signal from two or more previous samples of the subband signal is described. The subband signal corresponds to a plurality of subbands, having an equal subband spacing, of a subband-domain representation of an audio signal. The method comprises determining signal model data using a model parameter; determining a first prediction coefficient in response to the model parameter using a first lookup table and/or a first analytical function; determining a second prediction coefficient in response to the model parameter using a second lookup table and/or a second analytical function; and determining the estimate of the sample by applying the first prediction coefficient to the first previous sample and applying the second prediction coefficient to the second previous sample.

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 sample of a subband signal from two or more previous samples of the subband signal is described. The subband signal corresponds to a plurality of subbands, having an equal subband spacing, of a subband-domain representation of an audio signal. The method comprises determining signal model data using a model parameter; determining a first prediction coefficient in response to the model parameter using a first lookup table and/or a first analytical function; determining a second prediction coefficient in response to the model parameter using a second lookup table and/or a second analytical function; and determining the estimate of the sample by applying the first prediction coefficient to the first previous sample and applying the second prediction coefficient to the second previous sample.

A signal processor comprising: a modelling block, configured to receive a frequency-domain-input-signal, a fundamental-frequency-signal representative of a fundamental frequency of the frequency-domain-input-signal; and configured to provide a pitch-model-signal based on a periodic function, the pitch-model-signal spanning a plurality of discrete frequency bins, each discrete frequency bin having a respective discrete frequency bin index, wherein within each discrete frequency bin the pitch-model-signal is defined by: the periodic function; the fundamental frequency; the frequency-domain-input-signal; and the respective discrete frequency bin index. The signal processor further comprises a manipulation block, configured to provide an output-signal based on the frequency-domain-input-signal and the pitch-model-signal.

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).