Abstract: An electronic device measures noise variability of background noise present in a sampled audio signal, and determines whether the measured noise variability is higher than a high threshold value or lower than a low threshold value. If the noise variability is determined to be higher than the high threshold value, the device categorizes the background noise as having a high degree of variability. If the noise variability is determined to be lower than the low threshold value, the device categorizes the background noise as having a low degree of variability. The high and low threshold values are between a high boundary point and a low boundary point. The high boundary point is based on an analysis of files including noises that exhibit a high degree of variability, and the low boundary point is based on an analysis of files including noises that exhibit a low degree of variability.

Abstract: A disclosed method includes monitoring an audio signal energy level while having a plurality of signal processing components deactivated and activating at least one signal processing component in response to a detected change in the audio signal energy level. The method may include activating and running a voice activity detector on the audio signal in response to the detected change where the voice activity detector is the at least one signal processing component. The method may further include activating and running the noise suppressor only if a noise estimator determines that noise suppression is required. The method may activate and runs a noise type classifier to determine the noise type based on information received from the noise estimator and may select a noise suppressor algorithm, from a group of available noise suppressor algorithms, where the selected noise suppressor algorithm is the most power consumption efficient.

Abstract: The disclosure is directed to pre-processing audio signals. In one implementation, an electronic device receives an audio signal that has audio information, obtains auxiliary information (such as location, velocity, direction, light, proximity of objects, and temperature), and determines, based on the audio information and the auxiliary information, a type of audio environment in which the electronic device is operating. The device selects an audio pre-processing procedure based on the determined audio environment type and pre-processes the audio signal according to the selected pre-processing procedure. The device may then perform speech recognition on the pre-processed audio signal.

Abstract: One provides (101) a digital audio signal having a corresponding signal bandwidth, and then provides (102) an energy value that corresponds to at least an estimate of out-of-signal bandwidth energy as corresponds to that digital audio signal. One then uses (103) the energy value to simultaneously determine both a spectral envelope shape and a corresponding suitable energy for the spectral envelope shape for out-of-signal bandwidth content as corresponds to the digital audio signal. By one approach, if desired, one then combines (104) (on, for example, a frame by frame basis) the digital audio signal with the out-of-signal bandwidth content to provide a bandwidth extended version of the digital audio signal to be audibly rendered to thereby improve corresponding audio quality of the digital audio signal as so rendered.

Abstract: A method and apparatus for prediction in a speech-coding system extends a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, to a multi-tap LTP filter. From another perspective, a conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. Such a multi-tap LTP filter offers a number of advantages over the prior-art. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients (?i's) of the multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component. Consequently their main function is to maximize the prediction gain of the LTP filter via modeling the degree of periodicity that is present and by imposing spectral shaping.

Abstract: A method (100) includes receiving (101) an input digital audio signal comprising a narrow-band signal. The input digital audio signal is processed (102) to generate a processed digital audio signal. An estimate of the high-band energy level corresponding to the input digital audio signal is determined (103). Modification of the estimated high-band energy level is done based on an estimation accuracy and/or narrow-band signal characteristics (104). A high-band digital audio signal is generated based on the modified estimate of the high-band energy level and an estimated high-band spectrum corresponding to the modified estimate of the high-band energy level (105).

Abstract: A method includes defining a transition band for a signal having a spectrum within a first frequency band, where the transition band is defined as a portion of the first frequency band, and is located near an adjacent frequency band that is adjacent to the first frequency band. The method analyzes the transition band to obtain a transition band spectral envelope and a transition band excitation spectrum; estimates an adjacent frequency band spectral envelope; generates an adjacent frequency band excitation spectrum by periodic repetition of at least a part of the transition band excitation spectrum with a repetition period determined by a pitch frequency of the signal; and combines the adjacent frequency band spectral envelope and the adjacent frequency band excitation spectrum to obtain an adjacent frequency band signal spectrum. A signal processing logic for performing the method is also disclosed.

Abstract: A signal processing platform (300) presents (101) a signal to be processed and identifies (102) signal portions with specific characteristics that are used (103) to automatically determine at least one bounding frequency that can be used to facilitate bandwidth extension for the signal. Identifying these signal portions can comprise identifying signal portions that exhibit at least a predetermined level of energy. The step of determining the bounding frequency can comprise computing a magnitude spectrum for each of the identified signal portions that can be used to determine a corresponding measure of flatness within a pass band as pertains to a corresponding normalized signal portion to thereby provide corresponding vetted signal portions. Determining the bounding frequency can then comprise accumulating the magnitude spectrum for these vetted signal portions and using the resultant accumulation to estimate a corresponding signal envelope.

Abstract: A method (100) includes receiving (101) an input digital audio signal comprising a narrow-band signal. The input digital audio signal is processed (102) to generate a processed digital audio signal. A high-band energy level corresponding to the input digital audio signal is estimated (103) based on a transition-band of the processed digital audio signal within a predetermined upper frequency range of a narrow-band bandwidth. A high-band digital audio signal is generated (104) based on the high-band energy level and an estimated high-band spectrum corresponding to the high-band energy level.

Abstract: A method (100) includes receiving (101) an input digital audio signal comprising a narrow-band signal. The input digital audio signal is processed (102) to generate a processed digital audio signal. An estimate of the high-band energy level corresponding to the input digital audio signal is determined (103). Modification of the estimated high-band energy level is done based on an estimation accuracy and/or narrow-band signal characteristics (104). A high-band digital audio signal is generated based on the modified estimate of the high-band energy level and an estimated high-band spectrum corresponding to the modified estimate of the high-band energy level (105).

Abstract: A method (100) includes receiving (101) an input digital audio signal comprising a narrow-band signal. The input digital audio signal is processed (102) to generate a processed digital audio signal. An estimate of the high-band energy level corresponding to the input digital audio signal is determined (103). Modification of the estimated high-band energy level is done based on an estimation accuracy and/or narrow-band signal characteristics (104). A high-band digital audio signal is generated based on the modified estimate of the high-band energy level and an estimated high-band spectrum corresponding to the modified estimate of the high-band energy level (105).

Abstract: A method and apparatus for prediction in a speech-coding system extends a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, to a multi-tap LTP filter. From another perspective, a conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. Such a multi-tap LTP filter offers a number of advantages over the prior-art. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients (?i's) of the multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component. Consequently their main function is to maximize the prediction gain of the LTP filter via modeling the degree of periodicity that is present and by imposing spectral shaping.

Abstract: A method and apparatus for prediction in a speech-coding system is provided herein. The method of a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, is extended to a multi-tap LTP filter, or, viewed from another vantage point, the conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. This novel formulation of a multi-tap LTP filter offers a number of advantages over the prior-art LTP filter configurations. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients of such a multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component.

Abstract: A method includes defining a transition band for a signal having a spectrum within a first frequency band, where the transition band is defined as a portion of the first frequency band, and is located near an adjacent frequency band that is adjacent to the first frequency band. The method analyzes the transition band to obtain a transition band spectral envelope and a transition band excitation spectrum; estimates an adjacent frequency band spectral envelope; generates an adjacent frequency band excitation spectrum by periodic repetition of at least a part of the transition band excitation spectrum with a repetition period determined by a pitch frequency of the signal; and combines the adjacent frequency band spectral envelope and the adjacent frequency band excitation spectrum to obtain an adjacent frequency band signal spectrum. A signal processing logic for performing the method is also disclosed.

Abstract: A signal processing platform (300) presents (101) a signal to be processed and identifies (102) signal portions with specific characteristics that are used (103) to automatically determine at least one bounding frequency that can be used to facilitate bandwidth extension for the signal. Identifying these signal portions can comprise identifying signal portions that exhibit at least a predetermined level of energy. The step of determining the bounding frequency can comprise computing a magnitude spectrum for each of the identified signal portions that can be used to determine a corresponding measure of flatness within a pass band as pertains to a corresponding normalized signal portion to thereby provide corresponding vetted signal portions. Determining the bounding frequency can then comprise accumulating the magnitude spectrum for these vetted signal portions and using the resultant accumulation to estimate a corresponding signal envelope.

Abstract: A method (100) includes receiving (101) an input digital audio signal comprising a narrow-band signal. The input digital audio signal is processed (102) to generate a processed digital audio signal. An estimate of the high-band energy level corresponding to the input digital audio signal is determined (103). Modification of the estimated high-band energy level is done based on an estimation accuracy and/or narrow-band signal characteristics (104). A high-band digital audio signal is generated based on the modified estimate of the high-band energy level and an estimated high-band spectrum corresponding to the modified estimate of the high-band energy level (105).

Abstract: A method (100) includes receiving (101) an input digital audio signal comprising a narrow-band signal. The input digital audio signal is processed (102) to generate a processed digital audio signal. A high-band energy level corresponding to the input digital audio signal is estimated (103) based on a transition-band of the processed digital audio signal within a predetermined upper frequency range of a narrow-band bandwidth. A high-band digital audio signal is generated (104) based on the high-band energy level and an estimated high-band spectrum corresponding to the high-band energy level.

Abstract: One provides (101) a digital audio signal having a corresponding signal bandwidth, and then provides (102) an energy value that corresponds to at least an estimate of out-of-signal bandwidth energy as corresponds to that digital audio signal. One then uses (103) the energy value to simultaneously determine both a spectral envelope shape and a corresponding suitable energy for the spectral envelope shape for out-of-signal bandwidth content as corresponds to the digital audio signal. By one approach, if desired, one then combines (104) (on, for example, a frame by frame basis) the digital audio signal with the out-of-signal bandwidth content to provide a bandwidth extended version of the digital audio signal to be audibly rendered to thereby improve corresponding audio quality of the digital audio signal as so rendered.

Abstract: A speech communication system provides a speech encoder that generates a set of coded parameters representative of the desired speech signal characteristics. The speech communication system also provides a speech decoder that receives the set of coded parameters to generate reconstructed speech. The speech decoder includes an equalizer that computes a matching set of parameters from the reconstructed speech generated by the speech decoder, undoes the set of characteristics corresponding to the computed set of parameters, and imposes the set of characteristics corresponding to the coded set of parameters, thereby producing equalized reconstructed speech.

Abstract: A codebook excited linear prediction coding system providing improved digital speech coding for high quality speech at low bit rates with side-by-side codebooks for segments of the modeled input signal to reduce the complexity of the codebook search. A linear predictive filter responsive to an input signal desired to be modeled is used for identifying a basis vector from a first codebook over predetermined intervals as a subset of the input signal. A long term predictor and a vector quantizer provide synthetic excitation of modeled waveform signal components corresponding to the input signal desired to be modeled from side-by-side codebooks by providing codevectors with concatenated signals identified from the basis vector over the predetermined intervals with respect to the side-by-side codebooks. Once a codevector is identified, the codebook at the next segment is searched and a concatenation of codevectors is provided by selecting up to but not including the current segment.