Abstract: Techniques and tools related to delayed or lost coded audio information are described. For example, a concealment technique for one or more missing frames is selected based on one or more factors that include a classification of each of one or more available frames near the one or more missing frames. As another example, information from a concealment signal is used to produce substitute information that is relied on in decoding a subsequent frame. As yet another example, a data structure having nodes corresponding to received packet delays is used to determine a desired decoder packet delay value.

Abstract: An audio processing system includes a voice decoder and an audio processor. In one exemplary embodiment, the audio processing system is embedded in a headset unit that is wirelessly coupled to a game console. The voice decoder is used to decode a stream of incoming voice data packets carried over a wireless signal. The decoded voice data packets are used to drive an audio transducer of the headset unit. Upon detection of an error in the incoming stream, a decoded error-free voice data packet that has been stored in a replay buffer is used to generate an amplitude scaled audio signal. The voice decoder is disconnected from the audio transducer and the scaled audio signal is used to drive the audio transducer instead.

Abstract: A noise reducing apparatus includes: a voice signal inputting unit inputting an input voice signal; a noise occurrence period detecting unit detecting a noise occurrence period; a noise removing unit removing a noise for the noise occurrence period; a generation source signal acquiring unit acquiring a generation source signal with a time duration corresponding to a time duration corresponding to the noise occurrence period; a pitch calculating unit calculating a pitch of an input voice signal interval; an interval signal setting unit setting interval signals divided in each unit period interval; an interpolation signal generating unit generating an interpolation signal with the time duration corresponding to the noise occurrence period and alternately arranging the interval signal in a forward time direction and the interval signal in a backward time direction; and a combining unit combining the interpolation signal and the input voice signal, from which the noise is removed.

Abstract: There is provided a voice recognition device and a voice recognition method that enhance the function of noise adaptation processing in voice recognition processing and reduce the capacity of a memory being used. Acoustic models are subjected to clustering processing to calculate the centroid of each cluster and the differential vector between the centroid and each model, model composition between each kind of assumed noise model and the calculated centroid is carried out, and the centroid of each composition model and the differential vector are stored in a memory. In the actual recognition processing, the centroid optimal to the environment estimated by the utterance environmental estimation is extracted from the memory, model restoration is carried out on the extracted centroid by using the differential vector stored in the memory, and noise adaptation processing is executed on the basis of the restored model.

Abstract: A method for removing periodic noise pulses from a continuous audio signal generated in a pressurized air delivery system includes the steps of: detecting, in a time-windowed segment of the continuous audio signal generated in the pressurized air delivery system, a plurality of the periodic noise pulses having a pulse period and being representable in the form of a plurality of signal components combined by convolution; deconvolving the plurality of signal components to generate a plurality of deconvolved signal components; and removing at least a portion of the periodic noise pulses from the time-windowed segment of the continuous audio signal using the deconvolved signal components.

Abstract: The conventional error conceal processing generates a greatly fluctuating irregular sound which is unpleasant to ears and causes a remarkable echo effect and click noise. A notification signal detection unit (301) judges processing for an input frame. In case of an error frame, a sound detection unit (303) makes judgment whether a preceding non-error data frame is a sound signal. If it is a sound frame, a sound copying unit (304) generates a replacing frame. If it is a non-sound frame, a transient signal detection unit (305) judges whether it is an attack signal by the transient signal detection and selects an appropriate area from the preceding non-error frame.

Abstract: A system and method for performing frame loss concealment (FLC) when portions of a bit stream representing an audio signal are lost within the context of a digital communication system. The system and method utilizes a plurality of different FLC techniques, wherein each technique is tuned or designed for a different kind of audio signal. When a frame is lost, a previously-decoded audio signal corresponding to one or more previously-received good frames is analyzed. Based on the result of the analysis, the FLC technique that is most likely to perform well for the previously-decoded audio signal is chosen to perform the FLC operation for the current lost frame. In one implementation, the plurality of different FLC techniques include an FLC technique designed for music, such as a frame repeat FLC technique, and an FLC technique designed for speech, such as a periodic waveform extrapolation (PWE) technique.

Abstract: The invention relates to the processing of a digital signal originating from a decoder and a noise reduction post-processing step, including, in particular, limitation of distortion introduced by the post-processing step in order to deliver a corrected output signal (SOUT), assigning said corrected output signal (SOUT) with: a current amplitude having an intermediary value between a current amplitude value of the post-processed signal (SPOST) and a corresponding current amplitude value of the decoded signal (S?MIC), or the current amplitude of the post-processed signal (SPOST), according to the respective values of the current amplitude of the post-processed signal (SPOST) and by the corresponding current amplitude of the decoded signal (S?MIC).

Abstract: Presented herein are systems and methods for generating an adaptive noise codebook for use with electronic speech systems. The noise codebook includes a plurality of entries which may be updated based on environmental noise sounds. The speech system includes a speech codebook and the adaptive noise codebook. The system identifies speech sounds in an audio signal using the speech and noise codebooks.

Abstract: Channel normalization for automatic speech recognition is provided. Statistics are measured from an initial portion of a speech utterance. Feature normalization parameters are estimated based on the measured statistics and a statistically derived mapping relating measured statistics and feature normalization parameters. In some examples, the measured statistics comprise measures of an energy from the initial portion of the speech utterance. In some examples, measures of the energy comprise extreme values of the energy.

Abstract: A method and apparatus for enhancing voice intelligibility for network communications of speech such as, for example, VoIP (Voice-Over-Internet-Protocol), in the presence of packets which arrive too late for normal playout. When a late speech packet is received by a speech decoder, that packet and, if necessary, one or more additional packets subsequent thereto, are played out over a shorter than normal duration so that the decoder can “catch up” with the encoder. Since a voice frame is usually decoded in several sub-frames—typically two or three—this shortened playout may be achieved, for example, by skipping one sub-frame from each frame to be shortened.

Abstract: Various embodiments are disclosed relating to the real-time monitoring and control for audio devices. An apparatus may include a peripheral audio device configured to operate in an operational mode or a debug mode, the peripheral audio device including an audio enhancement logic configured to include at least one tunable parameter. The apparatus may also include the peripheral audio device being further configured to transmit and receive data via a data channel to allow a debug or test to be performed on the peripheral audio device, while operating in the debug mode, and the at least one tunable parameter to be adjusted.

Abstract: The present invention provides methods for universal lossy compression that provide performance at or near the rate-distortion limit and that are based on universal, implementable lossy source coding algorithms.

Abstract: The intelligibility of speech signals is improved in the many situations where a voice signal is communicated or stored. Means and methods are disclosed for developing a scheme with high voice signal intelligibility without sacrificing the voice quality. The disclosed method comprises certain steps, including, but not limited to: Learning the noise on near-end side and enhancing the far-end voice as a function of the noise type and noise level on the near-end side. The disclosed method and apparatus are especially useful to increase the intelligibility of the communication device's loudspeaker output. The invention includes processing of an input speech signal to generate an enhanced intelligent signal. The FFT spectrum of the speech received from the far-end is modified in accordance with the LPC spectrum of the local background noise to generate an enhanced intelligent signal.

Abstract: Provided is a code conversion device that is capable of converting codes even if an input code sequence is invalid, and is able to reduce the amount of processing. When a first code sequence is input, the code conversion device generates a decoded signal by decoding the codes of normal frames of the first code sequence at Step S1, stores and holds the decoded signal at Step S2, generates a signal corresponding to an invalid frame by interpolation with the decoded signal that is stored and held, at Step S3. Subsequently, the code conversion device generates codes corresponding to the invalid frame by encoding the generated signal at Step S4, and makes the normal frames of the first code sequence without conversion be the frames of the second code sequence while making the generated codes be the frame of the second code sequence, in place of the codes of the invalid frame, at Step S5.

Abstract: Preparing for the full-fledged aged society, measures to prevent senility are required. Senility is prevented by extracting signals of prescribed bands from a speech signal using a first bandpass filter section having a plurality of bandpass filters, extracting the envelopes of each frequency band signal using an envelope extraction section having envelope extractors, applying a noise source signal to a second bandpass filter section having a plurality of bandpass filters and extracting noise signals corresponding to the prescribed bands, multiplying the outputs from the first bandpass filter section and the second bandpass filter section in a multiplication section, summing up the outputs from the multiplication section in an addition section to produce a Noise-Vocoded Speech Sound signal, and presenting the Noise-Vocoded Speech Sound signal for listening.

Abstract: A method and apparatus identify a clean speech signal from a noisy speech signal. To do this, a clean speech value and a noise value are estimated from the noisy speech signal. The clean speech value and the noise value are then used to define a gain on a filter. The noisy speech signal is applied to the filter to produce the clean speech signal. Under some embodiments, the noise value and the clean speech value are used in both the numerator and the denominator of the filter gain, with the numerator being guaranteed to be positive.

Abstract: In a method of smoothing background noise in a telecommunication speech session; receiving and decoding S1O a signal representative of a speech session, the signal comprising both a speech component and a background noise component. Subsequently, determining LPC parameters S20 and an excitation signal S30 for the received signal. Thereafter, synthesizing and outputting (S40) an output signal based on the determined LPC parameters and excitation signal. In addition, modifying S35 the determined excitation signal by reducing power and spectral fluctuations of the excitation signal to provide a smoothed output signal.

Abstract: A method and system are provided for synthesizing a corrupted frame output from a decoder including one or more predictive filters. The corrupted frame is representative of one segment of a decoded signal output from the decoder. The method comprises extrapolating a replacement frame based upon another segment of the decoded signal and substituting the replacement frame for the corrupted frame. Finally, the internal states of the filters are updated based upon the substituting.

Abstract: In a noise suppression device, an audio detector detects presence or absence of audio in an input signal. A first noise spectrum estimator estimates a noise spectrum contained in the input signal based on the input signal and detection result of the audio detector. A second noise spectrum estimator estimates the noise spectrum based on the input signal regardless of the detection result of the audio detector. A noise spectrum calculator calculates a final noise spectrum estimation value according to a length of detecting time during which the audio detector continuously detects the audio and based on first and second noise spectrum estimation values that are obtained as estimation results by the first and second noise spectrum estimators. A gain calculator calculates a noise suppression gain based on the final noise spectrum estimation value. A noise suppressor suppresses noise contained in the input signal by applying the noise suppression gain to the input signal.

Abstract: A method and system for improving reception in wired and wireless systems through redundancy and iterative processing are provided. A multilayer decoding process may comprise a burst process and a frame process. Results from a first burst process may be utilized to generate a decoded bit sequence in the frame process. The frame process may utilize redundancy information and physical constraints to improve the performance of a decoding algorithm. Results from the frame process may be fed back for a second iteration of the burst process and of the frame process, to further improve the decoding operation. In some instances, the second iteration of the burst process may be based on a gradient search approach.

Abstract: Large scale subjective signal quality measurements for a mobile radio communications system are made using a large number of handheld subscriber radio communication units moving at various positions in the mobile radio communications system. Each handheld subscriber unit stores a copy of a test voice or video signal stream as does a quality management network node. An uplink subjective signal quality for each such handheld subscriber unit is determined based on a comparison of the stored test signal and the received test signal from the handheld subscriber unit. A downlink subjective signal quality to each handheld unit is based on the returned test signal stream received from the handheld subscriber unit and the stored test signal stream. Because the handheld units do not perform the subjective quality comparison calculations, ordinary subscriber units that do not require significant extra data processing resources associated with those calculations may be used.

Abstract: Techniques and tools are described for processing reconstructed audio signals. For example, a reconstructed audio signal is filtered in the time domain using filter coefficients that are calculated, at least in part, in the frequency domain. As another example, producing a set of filter coefficients for filtering a reconstructed audio signal includes clipping one or more peaks of a set of coefficient values. As yet another example, for a sub-band codec, in a frequency region near an intersection between two sub-bands, a reconstructed composite signal is enhanced.

Abstract: A sound processing system including a user headset for use in tactical military operations provides integrated sound and speech analysis including sound filtering and amplification, sound analysis and speech recognition for analyzing speech and non-speech sounds and taking programmed actions based on the analysis, recognizing language of speech for purposes of one-way and two-way voice translation, word spotting to detect and identify elements of conversation, and non-speech recognition and identification. The headset includes housings with form factors for insulating a user's ear from direct exposure to ambient sounds with at least one microphone for receiving sound around the user, and a microphone for receiving user speech. The user headset can further include interconnections for connecting the headset with out systems outside of the headset, including target designation systems, communication networks, and radio transmitters.

Abstract: Provided is an audio decoding device which can adjust the high-range emphasis degree in accordance with a background noise level.

Abstract: Disclosed are an image processing apparatus and an image processing method capable of changing a quality of audio data according to a degree of significance of an image. The image processing apparatus includes an image encoding unit for encoding image data inputted, an audio data encoding unit for encoding audio data inputted together with the image data, a significant scene setting unit for setting to encode a part of region of the image with a high image quality, and an audio data encoding setting unit for setting to process the audio data with a high quality in accordance with the setting by the significant scene setting unit.

Abstract: In a method of smoothing stationary background noise in a telecommunication speech session, initially receiving and decoding S10 a signal representative of a speech session, where the signal comprises both a speech component and a background noise component. Subsequently, providing S20 a noisiness measure for the signal, and adaptively S30 smoothing the background noise component based on the provided noisiness measure.

Abstract: Embodiments herein may receive a ranging request message with a sleep-mode indication from a mobile node at a destination base station in a wireless packet-switched network. System paging information may be accessed to determine a base station identifier associated with an originating base station that last served the mobile node. The originating base station may be contacted to retrieve a service context associated with the mobile node and any downlink packets buffered for the mobile node by the originating base station. Other embodiments may be described and claimed.

Abstract: The system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, a transmitter analysis filter for receiving the digital speech signal and dividing it into a baseband signal and an enhancement residual band signal, a standard baseband encoder for accepting the baseband signal and coding it using an ITU-T encoder, an additional baseband encoder for reducing standard coding distortion in the baseband signal, an enhancement residual band encoder for coding a signal obtained by removing the coded baseband signal from the original digital speech signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

Abstract: Method and system for measuring transmission quality of an audio transmission system under test. Specifically, an input signal (X), such as an original input speech signal, is applied to the audio transmission system which results in an output signal (Y) produced by the transmission system. Both signals X and Y are mutually processed to yield a perceived quality signal. In accordance with the invention, output signal Y and/or input signal X are scaled such that, depending on a ratio of power of these two signals, relatively small deviations of power between these signals are compensated, while relatively larger deviations are only partially compensated. Further, an artificial reference speech signal may be created for which noise levels present in the input speech signal are reduced by a scale factor which reflects a local level of the noise in that input signal.

Abstract: A double talk detector for controlling the echo path estimation in a telecommunication system by indicating when a received coded speech signal is dominated by a non-echo signal; i.e., that so-called double talk exists. This is determined by extracting LSPs from a coded speech frame of the received coded speech signal when the signal power exceeds a first threshold value, converting each of said extracted LSPs into LSFs, and calculating the distance between each two adjacent LSFs. For each distance that is smaller than a second threshold, a spectral peak is located between the two LSFs, and it is determined whether said spectral peak is an echo or not. When a predetermined number of non-echo spectral peaks are located in the received speech signal, double talk will be indicated, and the echo path estimation may be disabled.

Abstract: A method, device and system to implement hiding the loss packet are provided. The provided method, device and system recover the lost frame according to the data before and after the lost frame and enhances the correlation of the recovered lost frame data and the data after the lost frame. A method and device for estimating pitch period are also provided which select a pitch period from the initial pitch period and the pitch periods corresponding to the frequencies which are one or more times higher than the frequencies corresponding to the initial pitch period as the final estimated pitch period, may improve frequency multiplication when estimating the pitch period; in addition, by tuning of the pitch period by matching the waves, the error of estimating pitch period may be reduced and the quality of the audio data may be improved.

Abstract: A method and apparatus for dynamically enabling the activation and deactivation of comfort noise over a VoIP media path or channel are disclosed. The present method detects all sound levels in the media path and only activates the comfort noise in the absence of sound and when the background noise level or the telephone line noise level is low rather than only in the absence of speech.

Abstract: The present invention is a system and method that improves upon voice activity detection by packetizing actual noise signals, typically background noise. In accordance with the present invention an access network receives an input voice signal (including noise) and converts the input voice signal into a packetized voice signal. The packetized voice signal is transmitted via a network to an egress network. The egress network receives the packetized voice signal, converts the packetized voice signal into an output voice signal, and outputs the output voice signal. The egress network also extracts and stores noise packets from the received packetized voice signal and converts the packetized noise signal into an output noise signal. When the access network ceases to receive the input voice signal while the call is still ongoing, the access network instructs the egress network to continually output the output noise signal.

Abstract: A noise suppression device comprises subband SN ratio calculation means which receives a noise likeness signal, an input signal spectrum and a subband-based estimated noise spectrum, calculates the subband-based input signal average spectrum, calculates a subband-based mixture ratio of the subband-based estimated noise spectrum to the subband-based input signal average spectrum on the basis of the noise likeness signal, and calculates the subband-based SN ratio on the basis of the subband-based estimated noise spectrum, the subband-based input signal average spectrum and the mixture ratio.

Abstract: A system and method is provided for determining whether a data frame of a coded speech signal corresponds to voice or to noise. In one embodiment, a voice activity detector determines a cross-correlation of data. If the cross-correlation is lower than a predetermined cross-correlation value, then the data frame corresponds to noise. If not, then the voice activity detector determines a periodicity of the cross-correlation and a variance of the periodicity. If the variance is less than a predetermined variance value, then the data frame corresponds to voice. In another embodiment, a method determines energy of the data frame and an average energy of the coded speech signal. If the data frame is one of a predetermined number of initial data frames, then a comparison between the average energy to the energy of the data frame is used to determine whether the data frame is noise or voice.

Abstract: Techniques described herein include the use of equalization techniques to improve intelligibility of a reproduced audio signal (e.g., a far-end speech signal).

Abstract: The present invention relates to a voice region detection apparatus and method capable of accurately detecting a voice region even in a voice signal with color noise. The voice region detection method comprises the steps of, if a voice signal is input, dividing the input voice signal into frames; performing whitening of surrounding noise by combining white noise with the frames; extracting random parameters indicating randomness of frames from the frames subjected to the whitening; classifying the frames into voice frames and noise frames based on the extracted random parameters; and detecting a voice region by calculating start and end positions of a voice based on the voice and noise frames. According to the present invention, the voice region can be accurately detected even in a voice signal with a large amount of color noise mixed therewith.

Abstract: A code conversion method for converting first code string data conforming to a first speech coding scheme into second code string data conforming to a second speech coding scheme has the steps of decoding the first code string data to generate a first decoded speech, correcting the signal characteristics of the first decoded speech to generate a second decoded speech, and encoding the second decoded speech in accordance with the second speech coding scheme to generate the second code string data.

Abstract: Various embodiments of the present invention provide methods and systems for determining, representing, and using variable-length contexts in a variety of different computational applications. In one embodiment of the present invention, a balanced tree is used to represent all possible contexts of a fixed length, where the depth of the balanced tree is equal to the fixed length of the considered contexts. Then, in the embodiment, a pruning technique is used to sequentially coalesce the children of particular nodes in the tree in order to produce an unbalanced tree representing a set of variable-length contexts. The pruning method is selected, in one embodiment, to coalesce nodes, and, by doing so, to truncate the tree according to statistical considerations in order to produce a representation of a variably sized context model suitable for a particular application.

Abstract: A method and apparatus for detecting speech segments of a speech signal processing device is provided. A critical band is divided into a certain number of regions according to noise frequency characteristics, a signal threshold and a noise threshold are set for each of the regions, and it is determined whether each frame is a speech segment or noise segment by comparing the log energy calculated for each region to the corresponding signal threshold and noise threshold. Therefore, a speech segment can be detected rapidly and accurately by using a small number of operations even in a noise environment.

Abstract: A bit error concealment (BEC) system and method is described herein that detects and conceals the presence of click-like artifacts in an audio signal caused by bit errors introduced during transmission of the audio signal within an audio communications system. A particular embodiment of the present invention utilizes a low-complexity design that introduces no added delay and that is particularly well-suited for applications such as Bluetooth® wireless audio devices which have low cost and low power dissipation requirements.

Abstract: Techniques for suppressing noise from a signal comprised of speech plus noise. A first signal detector (e.g., a microphone) provides a first signal comprised of a desired component plus an undesired component. A second signal detector (e.g., a sensor) provides a second signal comprised mostly of an undesired component. The adaptive canceller removes a portion of the undesired component in the first signal that is correlated with the undesired component in the second signal and provides an intermediate signal. The voice activity detector provides a control signal indicative of non-active time periods whereby the desired component is detected to be absent from the intermediate signal. The noise suppression unit suppresses the undesired component in the intermediate signal based on a spectrum modification technique and provides an output signal having a substantial portion of the desired component and with a large portion of the undesired component removed.

Abstract: A system and method are provided that reduce noise in pattern recognition signals. To do this, embodiments of the present invention utilize a prior model of dynamic aspects of clean speech together with one or both of a prior model of static aspects of clean speech, and an acoustic model that indicates the relationship between clean speech, noisy speech and noise. In one embodiment, components of a noise-reduced feature vector are produced by forming a weighted sum of predicted values from the prior model of dynamic aspects of clean speech, the prior model of static aspects of clean speech and the acoustic-environmental model.

Abstract: Provided is a method and an apparatus for vocal-cord signal recognition. A signal processing unit receives and digitalizes a vocal cord signal, and a noise removing unit which channel noise included in the vocal cord signal. A feature extracting unit extracts a feature vector from the vocal cord signal, which has the channel noise removed therefrom, and a recognizing unit calculates a similarity between the vocal cord signal and the learned model parameter. Consequently, the apparatus is robust in a noisy environment.

Abstract: A non-intrusive speech quality assessment system. A method and apparatus for training a quality assessment tool in which a database comprising a plurality of samples, each with an associated mean opinion score, is divided into a plurality of distortion sets of samples according to a distortion criterion; and a distortion specific assessment handler for each distortion set is trained, such that a fit between a distortion specific quality measure generated from a distortion specific plurality of parameters for a sample and the mean opinion score associated with said sample is optimised. A method and apparatus for assessing speech quality in a telecommunications network in which a dominant distortion type is determined for a sample; a distortion specific plurality of parameters are combined to provide a distortion specific quality measure for each sample; and a quality measure is generated in dependence upon the distortion specific quality measure.

Abstract: Disclosed is a system and method for channel decoding speech frames in a receiver capable of multiple (M) codec modes, wherein channel encoded speech frames include an inband bit portion and a speech portion. An inband bit decoder decodes the inband bit portion (700) of a received frame to obtain confidence levels associated with each of the M codec modes. Using these confidence levels, the codec modes are ordered from most to least likely. The speech frame is then decoded by a channel decoder using the most likely codec mode (704). A frame determination check (720) is performed to determine the quality of the decoded speech frame. If the decoded speech frame is determined to be of poor quality, then the channel decoding process is repeated using the next most likely codec mode (736) corresponding to the next highest inband bit decoding confidence level. This process is repeated until a good speech frame is decoded or some exit criteria is reached.

Abstract: A method and apparatus identify a clean speech signal from a noisy speech signal. The noisy speech signal is converted into frequency values in the frequency domain. The parameters of at least one posterior probability of at least one component of a clean signal value are then determined based on the frequency values. This determination is made without applying a frequency-based filter to the frequency values. The parameters of the posterior probability distribution are then used to estimate a set of frequency values for the clean speech signal. A clean speech signal is then constructed from the estimated set of frequency values.

Abstract: Precision piecewise polynomial approximation for Ephraim-Malah filter is described herein. In one embodiment, an exemplary process includes computing a first parameter based on Wiener filter weights and posterior signal-to-noise (SNR) via a polynomial approximation mechanism without using a mathematical division operation, and generating Ephrain-Malah filter coefficients based on the first parameter. Other methods and apparatuses are also described.

Abstract: A method to achieve signals which are essentially devoid of unwanted signal components uses a search and comparison process. Media signals are received through receiving means, the media signals containing unwanted signal components, a representation for the media signals is chosen, and the media signals are processed in such a way that the unwanted signal components are essentially removed and the remaining signal components are saved.