Abstract: A method and an apparatus for processing a signal are provided. The method includes: obtaining an energy average value of each sub-band for a current frame frequency-domain signal; obtaining a current frame modification coefficient of each sub-band for the current frame frequency-domain signal according to a spectral envelope and the energy average value of each sub-band; obtaining a weighted modification coefficient of each sub-band for the current frame frequency-domain signal by using the current frame modification coefficient and a relevant frame modification coefficient; and modifying the spectral envelope of each sub-band for the current frame frequency-domain signal by using the weighted modification coefficient.

Abstract: The technology disclosed relates to audio signal processing. It includes a series of modules that individually are useful to solve audio signal processing problems. Among the problems addressed are buzz removal, selecting a pitch candidate among pitch candidates based on local continuity of pitch and regional octave consistency, making small adjustments in pitch, ensuring that a selected pitch is consistent with harmonic peaks, determining whether a given frame or region of frames includes harmonic, voiced signal, extracting harmonics from voice signals and detecting vibrato. One environment in which these modules are useful is transcribing singing or humming into a symbolic melody. Another environment that would usefully employ some of these modules is speech processing. Some of the modules, such as buzz removal, are useful in many other environments as well.

Abstract: A system and method for automatically adjusting floor controls based on conversational characteristics is provided. Audio streams are received, which each originate from an audio source. Floor controls for a current configuration including at least a portion of the audio streams are maintained. Conversational characteristics shared by two or more of the audio sources are determined. Possible configurations for the audio streams are identified based on the conversational characteristics. An analysis of the current configuration and the possible configurations is performed. A change threshold comprising a minimum number of timeslices for at least one of the current configuration and one of the possible configurations is applied to the analysis. When the analysis satisfies the change threshold, the floor controls are automatically adjusted. The audio streams are mixed into one or more outputs based on the adjusted floor controls.

Abstract: MDCT or FFT-based audio coding algorithms often have the problem named here spectral pre-echoes when coding an energy attack signal. This invention presents several possibilities to avoid the spectral pre-echoes existing in decoded signal segment before the energy attack point. The spectral envelope before the attack point can be improved by performing spectrum smoothing, replacing the segment of having spectral pre-echoes or filtering the segment with a combined filter obtained by doing LPC analysis.

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: It is possible to improve quality of a decoding signal in a band spread for estimating a high band from a low band of a decoding signal. A first layer encoder encodes a lower band portion below a predetermined frequency of an input signal so as to generate first layer encoded information. A first layer decoder decodes the first layer encoded information so as to generate a first layer demodulated signal. A second layer encoder divides a high band portion higher, than a predetermined frequency, of an input signal into a plurality of sub-bands and estimates each of the sub-bands from the input signal or the first layer decoded signal by using the estimation result of the sub-band adjacent to the lower band side so as to generate second encoded information including the estimation results of the sub-bands.

Abstract: Components of a method and system that allow identification of music from the song or sound using only the sound of the audio being played. A system built using the method and device components disclosed processes inputs sent from a mobile phone over a telephone or data connection, though inputs might be sent through any variety of computers, communications equipment, or consumer audio devices over any of their associated audio or data networks.

Abstract: An apparatus and method for adaptive sub-band allocation of spectral coefficients are disclosed. The sizes of sub-bands are determined according to the distribution of spectral coefficients transformed from an input speech/audio signal to perform more elaborate quantization in units of sub-bands. Thus, quantization noise of the spectral coefficients is reduced, and sound quality in a frequency region is enhanced, thereby improving the quality of the signal.

Abstract: A signal classifying method and apparatus are disclosed. The signal classifying method includes: obtaining a spectrum fluctuation parameter of a current signal frame determined as a foreground frame, and buffering the spectrum fluctuation parameter; obtaining a spectrum fluctuation variance of the current signal frame according to spectrum fluctuation parameters of all buffered signal frames, and buffering the spectrum fluctuation variance; and calculating a ratio of signal frames whose spectrum fluctuation variance is above or equal to a first threshold to all the buffered signal frames, and determining the current signal frame as a speech frame if the ratio is above or equal to a second threshold or determining the current signal frame as a music frame if the ratio is below the second threshold.

Abstract: In a sound effect applying apparatus, an input part frequency-analyzes an input signal of sound or voice for detecting a plurality of local peaks of harmonics contained in the input signal. A subharmonics provision part adds a spectrum component of subharmonics between the detected local peaks so as to provide the input signal with a sound effect. An output part converts the input signal of a frequency domain containing the added spectrum component into an output signal of a time domain for generating the sound or voice provided with the sound effect.

Abstract: An audio decoding device of the present invention includes: a decoding unit decoding a stream to a spectrum coefficient, and outputting stream information when a frame included in the stream cannot be decoded; an orthogonal transformation unit transforming the spectrum coefficient to a time signal; a correction unit generating a correction time signal based on an output waveform within a reference section that is in a section that overlaps between an error frame section to which the stream information is outputted and an adjacent frame section and that is a section in the middle of the adjacent frame section, when the decoding unit outputs the stream information: and an output unit generating the output waveform by synthesizing the correction time signal and the time signal.

Abstract: The invention describes a method of deriving a set of features (S) of an audio input signal (M), which method comprises identifying a number of first-order features (f1, f2, . . . , ff) of the audio input signal (M), generating a number of correlation values (?1, ?2, . . . , ?I) from at least part of the first-order features (f1, f2, . . . , ff), and compiling the set of features (S) for the audio input signal (M) using the correlation values (?1, ?2, . . . , ?I). The invention further describes a method of classifying an audio input signal (M) into a group, and a method of comparing audio input signals (M, M?) to determine a degree of similarity between the audio input signals (M, M?).

Abstract: There is disclosed an encoding device capable of appropriately adjusting the dynamic range of spectrum inserted according to the technique for replacing a spectrum of a certain band with a spectrum of another band. The device includes a spectrum modification unit (112) which modifies a first spectrum S1(k) of the band 0?k<FL in various ways to change the dynamic range so that a way of modification for obtaining an appropriate dynamic range is checked. The information concerning the modification is encoded and given to a multiplexing unit (115). By using a second spectrum S2(k) having a valid signal band 0?k.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: Provided are a method and apparatus for encoding the frequency of a continuation sinusoidal signal and a method and apparatus for decoding the same. In the encoding method, a continuation sinusoidal signal successive to a sinusoidal signal in a previous section is extracted from a current section; a frequency of the continuation sinusoidal signal at the boundary between the current and previous sections is changed to a first frequency, based on representative frequencies of the continuation sinusoidal signal and at least one sinusoidal signal that belongs to a section adjacent to the current section and is successive to the continuation sinusoidal signal; and the first frequency is encoded.

Abstract: In an apparatus and method, time-varying signals are processed and encoded via a frequency domain linear prediction (FDLP) scheme to arrive at an all-pole model. Residual signals resulted from the scheme are estimated and transformed into a time domain signal. Through the process of heterodyning, the time domain signal is frequency shifted toward the baseband level as a downshifted carrier signal. Quantized values of the all-pole model and the frequency transform of the downshifted carrier signal are packetized as encoded signals suitable for transmission or storage. To reconstruct the time-varying signals, the encoded signals are decoded. The decoding process is basically the reverse of the encoding process.

Abstract: Provided are a method, medium and apparatus for enhancing an acoustic signal using an auditory property. An acoustic signal is enhanced by generating a plurality of harmonic signals based on a predetermined acoustic signal frequency, selecting harmonic signals, which exist in an area masked by the predetermined harmonic signal, from among the generated plurality of harmonic signals, and outputting harmonic signals remaining after excluding the selected harmonic signals from the generated plurality of harmonic signals. The enhancement results in a bass signal of good sound quality and having a low distortion ratio, without changing the structure of a micro speaker.

Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.

Abstract: Likelihood calculation means extracts audio features expressing features of a voice signal and a non-voice signal from an acquired audio signal, and calculates likelihood expressing probability that the voice signal is included in the audio signal using the audio features. Spectral feature extraction means performs a frequency analysis to the audio signal to extract a spectral feature. Using the spectral feature, first basis matrix producing means produces a first basis matrix expressing the feature of the non-voice signal. Second basis matrix producing means specifies a component having a high association with the voice signal in the first basis matrix using the likelihood, and excludes the component to produce a second basis matrix. Spectral feature estimation means estimates a spectral feature of the voice signal or a spectral feature of the non-voice signal by performing nonnegative matrix factorization to the spectral feature using the second basis matrix.

Abstract: An apparatus and method for processing an audio signal including extracting noise filling flag information indicating whether noise filling is used to a plurality of frames; extracting coding scheme information indicating whether a current frame included in the plurality of frames is coded in either a frequency domain or a time domain; when the noise filling flag information indicates that the noise filling is used for the plurality of frames and the coding scheme information indicates that the current frame is coded in the frequency domain, extracting noise level information for the current frame; when a noise level value corresponding to the noise level information meets a predetermined level, extracting noise offset information for the current frame; and, when the noise offset information is extracted, performs the noise-filling for the current frame based on the noise level value and the noise offset information.

Abstract: An audio decoder includes an arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically encoded representation of the spectral values, and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state described by a numeric current context value. The arithmetic decoder determines the numeric current context value in dependence on a plurality of previously decoded spectral values. The arithmetic decoder evaluates a hash table, entries of which define both significant state values and boundaries of intervals of numeric context values, in order to select the mapping rule. A mapping rule index value is individually associated to a numeric context value being a significant state value.

Abstract: An audio input signal is filtered using an adaptive filter to generate a prediction output signal with reduced noise, wherein the filter is implemented using a plurality of coefficients to generate a plurality of prediction errors and to generate an error from the plurality of prediction errors, wherein the absolute values of the coefficients are continuously reduced by a plurality of reduction parameters.

Abstract: Methods, systems, and computer readable media for fricatives and high frequencies detection are disclosed. According to one method, the method includes receiving a narrowband signal. The method also includes detecting, using one or more autocorrelation coefficients, a high frequency speech component associated with the narrowband signal.

Abstract: According to one embodiment, a speech processing apparatus includes a histogram calculation unit, a cumulative frequency calculation unit, and a filter production unit. The histogram calculation unit is configured to calculate a first histogram from a first speech feature extracted from speech data, and to calculate a second histogram from a second speech feature different from the first speech feature. The cumulative frequency calculation unit is configured to calculate a first cumulative frequency by accumulating a frequency of the first histogram, and to calculate a second cumulative frequency by accumulating a frequency of the second histogram. The filter production unit is configured to produce a filter having a characteristic to get the second cumulative frequency near to the first cumulative frequency.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed, by which the audio signal can be efficiently processed. The present invention includes obtaining start position information of a sub-frame from a header of the main frame and processing an audio signal based on the start position information of the sub-frame, wherein the main frame includes a plurality of sub-frames.

Abstract: For audio encoding and decoding, in order to enhance coded audio signals, the audio signal is divided into at least a low frequency band and a high frequency band, the high frequency band is divided into at least two high frequency sub-band signals, and parameters are generated that refer at least to the low frequency band signal sections which match best with high-frequency sub-band signals.

Abstract: This invention realizes a speech communication system and method, and a robot apparatus capable of significantly improving entertainment property. A speech communication system with a function to make conversation with a conversation partner is provided with a speech recognition means for recognizing speech of the conversation partner, a conversation control means for controlling conversation with the conversation partner based on the recognition result of the speech recognition means, an image recognition means for recognizing the face of the conversation partner, and a tracking control means for tracing the existence of the conversation partner based on one or both of the recognition result of the image recognition means and the recognition result of the speech recognition means. The conversation control means controls conversation so as to continue depending on tracking of the tracking control means.

Abstract: A method and system for multi-channel detection of pitch may comprise one or more of the following steps and/or means therefore: (a) sampling an audio input stream including at least a first channel and a second channel; (b) setting a search frequency for each of the first channel and the second channel; and (c) detecting a pitch of the first channel and a pitch of the second channel.

Abstract: Speech encoders and methods of speech encoding are disclosed that encode inactive frames at different rates. Apparatus and methods for processing an encoded speech signal are disclosed that calculate a decoded frame based on a description of a spectral envelope over a first frequency band and the description of a spectral envelope over a second frequency band, in which the description for the first frequency band is based on information from a corresponding encoded frame and the description for the second frequency band is based on information from at least one preceding encoded frame. Calculation of the decoded frame may also be based on a description of temporal information for the second frequency band that is based on information from at least one preceding encoded frame.

Abstract: A method and apparatus for selectively replacing damaged portions of a data stream. The method comprises analyzing the data stream to identify damaged portions therein; selecting a damaged portion for replacement; and replacing the selected damaged portion. The selected damaged portion is selected for replacement in dependence on a rate of replacement, the rate of replacement being that at which previous portions of the data stream have been replaced.

Abstract: Character extraction section extracts character amounts, pertaining to a prosody of voice, from a voice signal sequentially in a time-serial manner. Difference value calculation calculates a difference value between each of the extracted character amounts and a reference value. Processing values, corresponding to the individual character amounts, are generated in accordance with the respective difference values, and a voice processing section controls the individual character amounts of the voice signal in accordance with the processing values corresponding to the character amounts and thereby generates an output signal having a prosody changed from the prosody of the voice signal.

Abstract: A method and an apparatus for detecting a pitch in input voice signals by using a subharmonic-to-harmonic ratio (SHR). The pitch detection method includes performing a Fourier transform on the input voice signals after performing a pre-processing on the input voice signals, performing an interpolation on the transformed voice signals, calculating a normalized local center of gravity (NLCG) on a spectrum of the interpolated voice signals, calculating a cumulated sum of the calculated NLCG, calculating an SHR from the spectrum based on the calculated cumulated sum, and extracting the pitch based on the calculated SHR.

Abstract: An audio playback speed control method and apparatus to control an audio playback speed using an optimal frame length with a small amount of calculation. The audio playback method includes extracting an audio sampling frequency and audio playback speed information from an audio signal which is reproduced, determining a length of an input frame, a length of an output frame, and a length of an overlapping region between frames, on a basis of the audio sampling frequency and the audio playback speed information and performing different overlapping and adding methods, according to the audio playback speeds, on a basis of the length of the input frame, the length of the output frame, and the length of the overlapping region between the frames.

Abstract: An encoding device reduces the encoding distortion as compared to the conventional technique and obtains a preferable sound quality for auditory sense. In the encoding device, a shape quantization unit quantizes the shape of an input spectrum with a small number of pulse positions and polarities. The shape quantization unit sets a pulse amplitude width to be searched later upon search of the pulse position to a value not greater than the pulse amplitude width which has been searched previously. A gain quantization unit calculates a gain of a pulse searched by the shape quantization unit for each of bands.

Abstract: A signal enhancement system reinforces signal content and improves the signal-to-noise ratio of a signal. The system detects, tracks, and reinforces non-stationary periodic signal components of a signal. The periodic signal components may represent vowel sounds or other voiced sounds. The system may detect, track, and attenuate quasi-stationary signal components in the signal.

Abstract: The disclosure provides a method for noise generation, including: determining an initial value of a reconstructed parameter; determining a random value range based on the initial value of the reconstructed parameter; taking a value in the random value range randomly as a reconstructed noise parameter; and generating noise by using the reconstructed noise parameter. The disclosure also provides an apparatus for noise generation.

Abstract: A method for processing a speech signal includes dividing the speech signal into a succession of frames, identifying one or more of the frames as click frames, and extracting phase information from the click frames. The speech signal is encoded using the phase information. Methods are also provided for modeling phase spectra of voiced frames and click frames.

Abstract: The invention comprises a lip reading device having a capacitive array for enhanced portable speech recognition. The capacitive array of the invention produces a sequence of signal frames or signal data sets (i.e., digitized output) representative of the proximity and motion of a user's lips at a predetermined sample rate and resolution. The sequence of signal data sets is stored in a first electronic memory and are compared against a reference data set representative of a predetermined acoustic signal stored in a second electronic memory. The attributes of signal data set are compared against the reference data set for likely data matches based on predetermined criteria.

Abstract: A method for modification of a cepstro-temporally smoothed gain function of a gain function resulting in a bias compensated spectral gain function is provided. The cepstro-temporal smoothing increases the quality of an enhanced output signal, as it affects only spectral outliers caused by estimation errors, while the speech characteristics are well preserved. However, due to the cepstral transform, the temporal smoothing is done in the logarithmic domain rather than the linear domain, and hence results in a certain bias. Thus, the method for a general bias compensation for a cepstro-temporal smoothing of spectral filter gain functions that is only dependent on the lower limit of the spectral filter-gain function.

Abstract: A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range. The speech enhancement system may be built into, may be a unitary part of, or may be configured to interface other systems that process audio or high frequency signals.

Abstract: A decoding apparatus that decodes a first encoded data that is encoded into a first time range from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of the audio signal from the low-frequency component and encoded into a second time range, into the audio signal. In the decoding apparatus, a high-frequency component compensating unit that compensates the high-frequency component created from the second encoded data based on the first time range. A decoding unit that decodes into the audio signal by synthesizing the high-frequency component compensated by the high-frequency component compensating unit, and the low-frequency component decoded from the first encoded data.

Abstract: An apparatus and method for estimating audio signal spectrum information. The method including the steps of performing a morphological operation on a received audio signal, extracting peaks by using various peak extraction methods and extracting a remainder signal region from the extracted peaks, selecting a high-order peaks spectrum from the extracted remainder signal region. In addition, spectral envelopes are detected by performing an interpolation operation on the high-order peaks spectrum.

Abstract: A state detecting device includes an input unit that receives an input voice sound; an analyzer that calculates a feature parameter of each of plurality of frames extracted from the voice sound; a calculator that calculates the average of the feature parameters of the frames, determines a threshold on the basis of the average and statistical data representing relationships between other averages of other feature parameters obtained from a plurality of speakers and cumulative frequencies of the other feature parameters, and calculates an appearance frequency of a frame that is among the plurality of frames and whose feature parameter is larger than the threshold; a determining unit that determines, on the basis of the appearance frequency, a strained state of a vocal cord that has made the voice sound; and an output unit that outputs a result of the determination.

Abstract: A signature is extracted from the audio of a program received by a tunable receiver such that the signature characterizes the program. In order to extract the signature, blocks of the audio are converted to corresponding spectral moments. At least one of the spectral moments is then converted to the signature. Also, a test audio signal from a receiver is correlated to a reference audio signal by converting the test audio signal and the reference audio signal to corresponding test and reference spectra, determining test slopes corresponding to coefficients of the test spectrum and reference slopes corresponding to coefficients of the reference spectrum, and comparing the test slopes to the reference slopes in order to determine a match between the test audio signal and the reference audio signal.

Abstract: The present invention relates to a method of automatically controlling the volume level of communication speech for Mean Opinion Score (MOS) measurement, which, before evaluating the quality of communication speech using a MOS measurement method, automatically controls the volume level of actual communication speech to a predetermined optimal level, thus improving the reliability of MOS values.

Abstract: An encoder/decoder for multi-channel audio data, and in particular for audio reproduction through wave field synthesis. The encoder comprises a two-dimensional filter-bank to the multi-channel signal, in which the channel index is treated as an independent variable as well as time, and and the resulting spectral coefficient are quantized according to a two-dimensional psychoacoustic model, including masking effect in the spatial frequency as well as in the temporal frequency. The coded spectral data are organized in a bitstream together with side information containing scale factors and Huffman codebook identifiers.

Abstract: The invention relates to the field of voice over Internet protocol (VoIP) and more specifically to a system and method of terminating a VoIP call.

Abstract: A down-sampler 101 down-samples the sampling rate of an input signal from sampling rate FH to sampling rate FL. A base layer coder 102 encodes the sampling rate FL acoustic signal. A local decoder 103 decodes coding information output from base layer coder 102. An up-sampler 104 raises the sampling rate of the decoded signal to FH. A subtracter 106 subtracts the decoded signal from the sampling rate FH acoustic signal. An enhancement layer coder 107 encodes the signal output from subtracter 106 using a decoding result parameter output from local decoder 103.

Abstract: This invention realizes a speech communication system and method, and a robot apparatus capable of significantly improving entertainment property. A speech communication system with a function to make conversation with a conversation partner is provided with a speech recognition means for recognizing speech of the conversation partner, a conversation control means for controlling conversation with the conversation partner based on the recognition result of the speech recognition means, an image recognition means for recognizing the face of the conversation partner, and a tracking control means for tracing the existence of the conversation partner based on one or both of the recognition result of the image recognition means and the recognition result of the speech recognition means. The conversation control means controls conversation so as to continue depending on tracking of the tracking control means.

Abstract: During operation an input signal to be coded is received and coded to produce a coded audio signal. The coded audio signal is then scaled with a plurality of gain values to produce a plurality of scaled coded audio signals, each having an associated gain value and a plurality of error values are determined existing between the input signal and each of the plurality of scaled coded audio signals. A gain value is then chosen that is associated with a scaled coded audio signal resulting in a low error value existing between the input signal and the scaled coded audio signal. Finally, the low error value is transmitted along with the gain value as part of an enhancement layer to the coded audio signal.