Abstract: Systems and methods for reconstructing decomposed audio signals are presented. In exemplary embodiments, a decomposed audio signal is received. The decomposed audio signal may include a plurality of frequency sub-band signals having successively shifted group delays as a function of frequency from a filter bank. The plurality of frequency sub-band signals may then be grouped into two or more groups. A delay function may be applied to at least one of the two or more groups. Subsequently, the groups may be combined to reconstruct the audio signal, which may be outputted accordingly.

Abstract: Disclosed is the system and method to remove noises in voice signals in a voice communication. The at least one embodiment of the present disclosure performs a spectral subtraction (SS) for voice signals based on a gain function by a spectral subtraction apparatus, performs clustering of voice signals consecutive on a frequency axis of a spectrogram for the voice signals in which the spectral subtraction has been already performed to designate one or more clusters, and extracts musical noises by determining continuity of each of the designated clusters on the frequency axis and a time axis of the spectrogram to extract musical noises.

Abstract: A non-spatial speech detection system includes a plurality of microphones whose output is supplied to a fixed beamformer. An adaptive beamformer is used for receiving the output of the plurality of microphones and one or more processors are used for processing an output from the fixed beamformer and identifying speech from noise though the use of an algorithm utilizing a covariance matrix.

Abstract: A method comprising receiving at a user equipment encrypted content. The content is stored in said user equipment in an encrypted form. At least one key for decryption of said stored encrypted content is stored in the user equipment.

Abstract: A speech enhancement system enhances transitions between speech and non-speech segments. The system includes a background noise estimator that approximates the magnitude of a background noise of an input signal that includes a speech and a non-speech segment. A slave processor is programmed to perform the specialized task of modifying a spectral tilt of the input signal to match a plurality of expected spectral shapes selected by a Codec.

Abstract: A voice correction device includes a detector that detects a response from a user, a calculator that calculates an acoustic characteristic amount of an input voice signal, an analyzer that outputs an acoustic characteristic amount of a predetermined amount when having acquired a response signal due to the response from the detector, a storage unit that stores the acoustic characteristic amount output by the analyzer, a controller that calculates an correction amount of the voice signal on the basis of a result of a comparison between the acoustic characteristic amount calculated by the calculator and the acoustic characteristic amount stored in the storage unit, and a correction unit that corrects the voice signal on the basis of the correction amount calculated by the controller.

Abstract: An electrical apparatus a voice signal receiving method thereof are disclosed. The electrical apparatus includes a plurality of voice receivers, a voice activity detector, a voice channel switch and a noise eliminator. The voice receivers are used to receive the voice signals. The voice activity detector receives and detects the voice signals, and obtains a main voice signal from the voice signals. The voice channel switch transports the main voice signal to a voice transporting channel and transports a plurality of other voice signals of the voice signals other than the main voice signal to a noise transporting channel according to a detecting result of the voice activity detector. The noise eliminator reduces the noise in the main voice according to the voice signals from the noise transporting channel.

Abstract: A Voice Recognition Enhancement Method for wireless telephonic communication devices includes providing an input voice audio source, enhancing the voice audio input in one or more of harmonic and dynamic ranges and outputting the voice enhanced audio. The Voice Recognition Enhancement method is suitable for use of wireless telephony devices, such as cellular phones. The enhancement includes resynthesizing audio to an increased harmonic and dynamic range than original values.

Abstract: A Voice Call Enhancement Method for wireless telephonic communication devices includes providing an input voice audio source, enhancing the voice audio input in multiple harmonic and dynamic ranges and outputting the voice enhanced audio. The Voice Call Enhancement method is suitable for use of wireless telephony devices, such as cellular phones. The enhancement includes resynthesizing audio to an increased harmonic and dynamic range than original values.

Abstract: A method and an apparatus for processing an audio signal in a mobile terminal, in which an audio signal that is received from a counterpart mobile terminal is classified into a voice signal and a noise signal according to respective energy. A frequency of the classified voice signal and an energy of the classified noise signal is controlled according to a predetermined criteria, then the controlled voice signal and the controlled noise signal are coupled and output to a speaker.

Abstract: Methods and apparatus for signal processing are disclosed. Source separation can be performed to extract source signals from mixtures of source signals by way of independent component analysis. Source separation described herein involves mixed multivariate probability density functions that are mixtures of component density functions having different parameters corresponding to frequency components of different sources, different time segments, or some combination thereof.

Abstract: In response to a first envelope within a kth frequency band of a first channel, a speech level within the kth frequency band of the first channel is estimated. In response to a second envelope within the kth frequency band of a second channel, a noise level within the kth frequency band of the second channel is estimated. A noise suppression gain for a time frame n is computed in response to the estimated speech level for a preceding time frame, the estimated noise level for the preceding time frame, the estimated speech level for the time frame n, and the estimated noise level for the time frame n. An output channel is generated in response to multiplying the noise suppression gain for the time frame n and the first channel.

Abstract: Methods and apparatus for signal processing are disclosed. Source separation can be performed to extract source signals from mixtures of source signals by way of independent component analysis. Source direction information is utilized in the separation process, and independent component analysis techniques described herein use multivariate probability density functions to preserve the alignment of frequency bins in the source separation process. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims.

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

Abstract: Disclosed in the present invention is a method for cancelling echo in joint time domain and frequency domain.

Abstract: A method and system is disclosed for control of discontinuous transmission based on vocoder and voice activity. An access terminal (AT) may engage in a communication session via an encoder-decoder in a network device in a wireless network. During silence intervals of the communication session, when the AT has no data to transmit, the AT may transmit periodic silence frames at a silence-frame rate to the encoder-decoder. The silence frames may contain parameters for generation of audio noise by the network device. Upon determining that the encoder-decoder has ceased transmitting data to the AT in response to a prolonged absence of transmissions from the AT, the AT may increase the silence-frame rate so as to reduce the duration of the absence of transmissions from the AT, and correspondingly cause the encoder-decoder to begin transmitting audio data to the AT.

Abstract: Systems and methods for adaptive power control are provided. In exemplary embodiments, a primary signal is received. A noise power level of the primary signal is then estimated. The noise power level may then be compared to at least one power threshold. Subsequently, a large power consuming system is controlled based on the comparison of the noise power level to the power threshold.

Abstract: In an apparatus and method for isolating a multi-channel sound source, the probability of speaker presence calculated when noise of a sound source signal separated by GSS is estimated is used to calculate a gain. Thus, it is not necessary to additionally calculate the probability of speaker presence when calculating the gain, the speaker's voice signal can be easily and quickly separated from peripheral noise and reverb and distortion are minimized. As such, if several interference sound sources, each of which has directivity, and speakers are simultaneously present in a room with high reverb, a plurality of sound sources generated from several microphones can be separated from one another with low sound quality distortion, and the reverb can also be removed.

Abstract: The embodiments described herein are directed to systems and methods for transmitting audio data and control segment in a single bitstream and reducing audio disturbance associated with the control segment when the bitstream is processed by an audio digital-to-analog converter. The system, according to one aspect, comprises a first audio unit, a transmitter coupled to the first audio unit, a receiver coupled to the transmitter, a second audio unit coupled to the receiver, a first processor coupled to at least one of the first audio unit and the transmitter, a second processor coupled to the second audio unit and the receiver, and an audio digital-to-analog converter connected to the second processor.

Abstract: A method of hierarchical coding of a digital audio frequency input signal into several frequency sub-bands, including a core coding of the input signal according to a first throughput and at least one enhancement coding of higher throughput, of a residual signal. The core coding uses a binary allocation according to an energy criterion. The method includes for the enhancement coding: calculating a frequency-based masking threshold for at least part of the frequency bands processed by the enhancement coding; determining a perceptual importance per frequency sub-band as a function of the masking threshold and as a function of the number of bits allocated for the core coding; binary allocation of bits in the frequency sub-bands processed by the enhancement coding, as a function of the perceptual importance determined; and coding the residual signal according to the bit allocation. Also provided are a decoding method, a coder and a decoder.

Abstract: A system for encoding an audio signal includes an audio console configured to receive a voice audio signal contained within a first audio spectrum, encode the voice audio signal with a background audio signal contained within a second audio spectrum wider than the first audio spectrum, encode the voice audio signal with a monitoring code and output a combined signal including the voice audio signal encoded with the background audio signal and the monitoring code. The combined signal is contained within an audio spectrum including the first audio spectrum and the second audio spectrum.

Abstract: The invention proposes extracting one or more speech signals (151-154) as well as one or more ambient signals (131) from sound signals captured by microphones, wherein each of the speech signals corresponds to a different speaker. The invention proposes to transmit both the one or more speech signals (151-154) and the one or more ambient signals (131) to a rendering side, as opposed to sending only speech signals. This enables to reproduce the speech and ambient signals in a spatially different way at the rendering side. By reproducing the ambient signals a feeling of “being together” is created. In an embodiment, the invention enables reproducing two or more speech signals spatially from each other and from the ambient signals so that speech intelligibility is increased despite the presence of the ambient signals.

Abstract: A method is disclosed for controlling a voice-activated device by interpreting a spoken command as a series of voiced and non-voiced intervals. A responsive action is then performed according to the number of voiced intervals in the command. The method is well-suited to applications having a small number of specific voice-activated response functions. Applications using the inventive method offer numerous advantages over traditional speech recognition systems including speaker universality, language independence, no training or calibration needed, implementation with simple microcontrollers, and extremely low cost. For time-critical applications such as pulsers and measurement devices, where fast reaction is crucial to catch a transient event, the method provides near-instantaneous command response, yet versatile voice control.

Abstract: An encoding method includes: extracting core layer characteristic parameters and enhancement layer characteristic parameters of a background noise signal, encoding the core layer characteristic parameters and enhancement layer characteristic parameters to obtain a core layer codestream and an enhancement layer codestream. The disclosure also provides an encoding device, a decoding device and method, an encapsulating method, a reconstructing method, an encoding-decoding system and an encoding-decoding method. By describing the background noise signal with the enhancement layer characteristic parameters, the background noise signal can be processed by using more accurate encoding and decoding method, so as to improve the quality of encoding and decoding the background noise signal.

Abstract: A Yule-Walker based, low-complexity voice activity detector (VAD) is disclosed. An input signal is typically noisy speech (i.e., corrupted with, for example, babble noise). In one embodiment, a first initialization stage of the VAD computes an occurrence of a silent period within the input signal and the AR parameters. The VAD could accordingly compute a tentative adaptive threshold and output hypothesis H1 (which means speech is present) during this stage. During the second initialization stage, the VAD generally builds a database of associated values and computes the adaptive threshold accordingly. The second initialization stage could also output tentative VAD decisions based on the tentative threshold computed in the first initialization stage. Finally, the VAD periodically retrains or updates AR parameters, threshold values and/or the database and outputs VAD decisions accordingly.

Abstract: A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.

Abstract: A device for improving the intelligibility of a signal arising from a source subjected to a noisy environment, said source marking the signal with a specific signature, the device comprising a processing circuit receiving the signal; and means for analyzing the signal and parameterizing the processing circuit according to characteristics of the signature present in the signal. A first channel with low distortion conveys the signal from the source to the means for analyzing, and a second channel, susceptible to introduce a distortion, conveys the signal from the source to the processing circuit.

Abstract: The present invention provides a speech enhancing method for communication earphone including two parts: sending end noise reduction processing and receiving end noise reduction processing, wherein the sending end noise reduction processing part includes: determining a wearing condition of the earphone by comparing energy difference of sound signals picked up by microphones of the communication earphone; if the earphone is normally worn, subjecting the sound signal first to multi-microphone noise reduction and then to single channel noise reduction to further suppress residuary stationary noise; otherwise suppressing stationary noise in the sound signal by single channel noise reduction directly.

Abstract: A noise estimation method for a noisy speech signal according to an embodiment of the present invention includes the steps of approximating a transformation spectrum by transforming an input noisy speech signal to a frequency domain, calculating a smoothed magnitude spectrum having a decreased difference in a magnitude of the transformation spectrum between neighboring frames, calculating a search spectrum to represent an estimated noise component of the smoothed magnitude spectrum, and estimating a noise spectrum by using a recursive average method using an adaptive forgetting factor defined by using the search spectrum. According to an embodiment of the present invention, the amount of calculation for noise estimation is small, and large-capacity memory is not required. Accordingly, the present invention can be easily implemented in hardware or software. Further, the accuracy of noise estimation can be increase because an adaptive procedure can be performed on each frequency sub-band.

Abstract: An encoder includes a precoder for encoding an input information object according to a preset encoding scheme and storing the encoded information object in a precoder buffer, a sample number/address generation unit for generating a sample number of each sample and an address, which corresponds to each bit of each sample and the address of the precoder buffer, a multiplexer for selecting a bit of the precoder buffer corresponding to the address generated by the sample number/address generation module, a sampling buffer for storing a bit of each sample output from the multiplexer, a control packet generation module for generating a control packet including information on the sample number generated by the sample number/address generation module, a packet assembling unit for assembling the sample stored in the sampling buffer with the control packet generated by the control data generation module, and a modulation module for modulating the packet output from the packet assembling unit into a sound signal accordi

Abstract: Contextual, focus-based language translation within a broadcast radio automation software application allows the user interface to enable interaction with users having various written or graphical language requirements. The software does not require special training or learning to enable the translation of text within the application. When the user enters a translation mode and selects a text string in the original language, the program displays a corresponding text string in a second language based on the specific display area, e.g., child window. The translated text string can be displayed proximate to original first text string within the specific display area, and can further be displayed as an entry in a table of a translation window which also displays additional text strings in the second language corresponding to additional text string fields for the specific display area.

Abstract: An approach to performing inverse quantization on a quantized integral value is described. This approach involves determining whether a quantized integral value lies within a first range or a second range of possible values. An interpolated inverse quantization value is calculated from the quantized integral value, using a predetermined bit shifting operation, depending on whether the quantized integral value was in the first or the second range.

Abstract: A system is provided for transmitting information through a speech codec (in-band) such as found in a wireless communication network. A modulator transforms the data into a spectrally noise-like signal based on the mapping of a shaped pulse to predetermined positions within a modulation frame, and the signal is efficiently encoded by a speech codec. A synchronization sequence provides modulation frame timing at the receiver and is detected based on analysis of a correlation peak pattern. A request/response protocol provides reliable transfer of data using message redundancy, retransmission, and/or robust modulation modes dependent on the communication channel conditions.

Abstract: A correction spectrum calculation unit 6 obtains a correction spectrum by smoothing an estimated noise spectrum in accordance with the degree of its variations, and a suppression quantity limiting coefficient calculation unit 7 decides a suppression quantity limiting coefficient from the correction spectrum. A suppression quantity calculation unit 9 obtains a suppression coefficient based on the suppression quantity limiting coefficient, and the spectrum suppression unit 10 carries out amplitude suppression of spectral components of an input signal.

Abstract: A method of pre-processing an audio signal transmitted to a user terminal via a communication network and an apparatus using the method are provided. The method of pre-processing the audio signal may prevent deterioration of a sound quality of the audio signal transmitted to the user terminal by pre-processing the audio signal, and by enabling a codec module, encoding the audio signal, to determine the audio signal as a speech signal. The method of pre-processing may include separating the audio signal into channels, measuring the channel energy for each of the channels, selecting a specific channel energy, and amplifying the specific channel energy. The method may include encoding an audio signal using a speech codec and/or decoding an encoded audio signal using the speech codec.

Abstract: The invention concerns an encoder for an input audio signal (S(z)) comprising a combination module combining the input audio signal with an intermediate counter-reaction signal forming a modified input signal and a quantification module scalable for the rate (91) of said modified input signal, delivering a binary raster of quantification indexes of a predetermined rate.

Abstract: A system enhances the quality of a digital speech signal that may include noise. The system identifies vocal expressions that correspond to the digital speech signal. A signal-to-noise ratio of the digital speech signal is measured before a portion of the digital speech signal is synthesized. The selected portion of the digital speech signal may have a signal-to-noise ratio below a predetermined level and the synthesis of the digital speech signal may be based on speaker identification.

Abstract: A speech processing system includes a plurality of signal analyzers that extract salient signal attributes of an input voice signal. A difference module computes the differences in the salient signal attributes. One or more control modules control a plurality of speech generators using an output signal from the difference module in a speech-locked loop (SLL), the speech generators use the output signal to generate a voice signal.

Abstract: An audio signal processing system including a time-frequency conversion unit which converts an audio signal in time domain into frequency domain in frame units so as to calculate a frequency spectrum of the audio signal, a spectral change calculation unit which calculates an amount of change between a frequency spectrum of a first frame and a frequency spectrum of a second frame before the first frame based on the frequency spectrum of the first frame and the frequency spectrum of the second frame, and a judgment unit which judges the type of the noise which is included in the audio signal of the first frame in accordance with the amount of spectral change.

Abstract: A method, system and program for encoding and/or decoding a speech signal. The method comprises: generating a first signal representing a property of an input speech signal; transforming the first signal using a simulated random-noise signal, thus producing a second signal; quantizing the second signal based on a plurality of discrete representation levels, thus generating quantization values for transmission in an encoded speech signal, and also generating a third signal being a quantized version of the second signal; and performing an inverse of the transformation on the third signal, thus generating a quantized output signal, wherein the generation of the first signal is based on feedback of the quantized output signal. The method further comprises controlling the transformation in dependence on a property of the first signal so as to vary the magnitude of a noise effect created by the transformation relative to the representation levels.

Abstract: Encoding audio signals with selecting an encoding mode for encoding the signal categorizing the signal into active segments having voice activity and non-active segments having substantially no voice activity by using categorization parameters depending on the selected encoding mode and encoding at least the active segments using the selected encoding mode.

Abstract: A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith.

Abstract: An information processing apparatus includes an acquisition unit configured to acquire a first sound recorded from a first recording apparatus and a second sound recorded from a second recording apparatus that is different from the first recording apparatus, a determination unit configured to determine a frequency band representing a voice by analyzing a frequency of the first sound, and a change unit configured to, from among frequency components representing the second sound, change a frequency component in the frequency band.

Abstract: A system and method may be configured to analyze audio information derived from an audio signal. The system and method may track sound pitch across the audio signal. The tracking of pitch across the audio signal may take into account change in pitch by determining at individual time sample windows in the signal duration an estimated pitch and a representation of harmonic envelope at the estimated pitch. The estimated pitch and the representation of harmonic envelope may then be implemented to determine an estimated pitch for another time sample window in the signal duration with an enhanced accuracy and/or precision.

Abstract: A noise suppressor for altering a speech signal is trained based on a speech recognition system. An objective function can be utilized to adjust parameters of the noise suppressor. The noise suppressor can be used to alter speech signals for the speech recognition system.

Abstract: Techniques pertaining to noise reduction are disclosed. According to one aspect of the present invention, noise in an audio signal is effectively reduced and a high quality of a target voice is recovered at the same time. In one embodiment, an array of microphones is used to sample the audio signal embedded with noise. The samples are processed according to a beamforming technique to get a signal with an enhanced target voice. A target voice is located in the audio signal sampled by the microphone array. A credibility of the target voice is determined when the target voice is located. The voice presence probability is weighted by the credibility. The signal with the enhanced target voice is enhanced according to the weighed voice presence probability.

Abstract: The invention relates to a technique of operating a call control node controlling at least one section of a call path. The call path includes between two opposite edge nodes a multi-section harmonization path along which codec selection is to be harmonized. A method embodiment of the technique, wherein the call control node is a transfer node in the harmonization path between the edge nodes, comprises the steps of determining if the call control node is a transfer node of the harmonization path; determining if a codec used for the at least one section controlled by the call control node fulfills a predefined harmonization criterion; and providing, in case the used codec does not fulfill the harmonization criterion, a harmonization trigger indication to at least one of the edge nodes of the harmonization path for initiating harmonization.

Abstract: The sound device includes an audio-information output unit, an analysis unit, an audio-division-spectrum output unit, a noise-division-spectrum output unit and a correction unit. The analysis unit receives audio information from the audio-information output unit, and then outputs sound spectrum information. The noise-division-spectrum output unit outputs sound-volume information for each critical band width of a noise, and the audio-division-spectrum output unit outputs the sound-volume information for each critical band width of the sound-spectrum information. The correction unit corrects the information from the audio-division-spectrum output unit based on the information from the noise-division-spectrum output unit. The audio-signal properties can be well corrected corresponding to the auditory-sense properties of the human, and thus the audio sound, in which an uncomfortable feeling to the auditory sense of the human has been adequately controlled, can be transmitted to a user.

Abstract: An apparatus of this invention is a speech processing apparatus that acquires pseudo speech from a mixture sound including desired speech and noise. The speech processing apparatus includes a first microphone that inputs a first mixture sound including desired speech and noise and outputs a first mixture signal, a second microphone that is opened to the same sound space as that of said first microphone and disposed at a focus position of an interface that is part of a boundary of the sound space and has one of a quadratic surface shape and a pseudo surface shape approximating a quadratic surface, inputs a second mixture sound including the desired speech reflected by the interface and the noise reflected by the interface at a ratio different from the first mixture sound, and outputs a second mixture signal, and a noise suppression circuit that suppresses an estimated noise signal based on the first mixture signal and the second mixture signal and outputs a pseudo speech signal.

Abstract: System and methods are provided for voice enhancement in audio conferencing among a plurality of participants. An example system includes a signal processor, a pre-processing component, and a voice-enhancement component. The signal processor is configured to generate a first mixed signal based at least in part on a first audio signal associated with a first remote participant and a local audio signal associated with a local participant. The pre-processing component is configured to generate a first input signal and a second input signal based at least in part on the first mixed signal and a second audio signal associated with a second remote participant. In addition, the voice-enhancement component is configured to generate a first output signal to be transmitted to the second remote participant based at least in part on the first input signal and the second input signal.