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: A disclosed speech sound enhancement device includes an SNR calculation unit for calculating an SNR which is a ratio of received speech sound to ambient noise; a first-frequency-range enhancement magnitude calculation unit for calculating, based on the SNR and frequency-range division information indicating a first and a second frequency range, enhancement magnitude of the first frequency range, the first frequency range contributing to an improvement of subjective intelligibility of the received speech sound, the second frequency range contributing to an improvement of subjective articulation of the received speech sound; a second-frequency-range enhancement magnitude calculation unit for calculating enhancement magnitude of the second frequency range based on the enhancement magnitude of the first frequency range; and a spectrum processing unit for processing spectra of the received speech sound using the enhancement magnitude of the first frequency range, the enhancement magnitude of the second frequency r

Abstract: A multimedia device can be used to play audio. Speech in an environment proximate to a multimedia device can be detected. The detected speech can be recorded. The playing of the audio can be paused. The recorded speech can be audibly presented. A condition to resume the paused audio can be detected. The paused audio can be resumed from the previously paused position.

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 speech enhancement system improves speech conversion within an encoder and decoder. The system includes a first device that converts sound waves into operational signals. A second device selects a template that represents an expected signal model. The selected template models speech characteristics of the operational signals through a speech codebook that is further accessed in a communication channel.

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

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: A technique is provided for limiting distortion of an audio signal being processed for playback by an audio device. In accordance with the technique, the audio signal is compressed to generate a compressed audio signal having a level that does not exceed a compression limit. The compressed audio signal is then soft clipped signal to generate a soft-clipped audio signal having a level that does not exceed a soft clipping limit, wherein the compression limit exceeds the soft clipping limit. The technique may also include passing the audio signal through a shaping filter prior to compressing the audio signal, wherein passing the audio signal through a shaping filter comprises modifying the level of selected frequency components of the audio signal.

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: Disclosed is a gain control system in which speech model constituted from a sound pressure and a feature is stored in a speech model storage unit for each of a plurality of phonemes or for each of clusters into which a speech is divided. When an input signal is given, a feature conversion unit calculates a feature and a sound pressure of the input signal. A sound pressure comparison unit determines a sound pressure ratio between the input signal and each of speech models. A distance calculation unit calculates a distance between the feature of the input signal and the feature of each of the speech models. A gain calculation unit calculates a gain value from the sound pressure ratio and information on the distance. A sound pressure compensation unit thereby compensates for the sound pressure of the input signal.

Abstract: A communication apparatus for adjusting a received voice signal in accordance with an ambient noise, the communication apparatus includes: a microphone for receiving an ambient noise and input voice and outputting a voice input signal corresponding to a level of the input voice and the ambient noise; a receiver for receiving the voice signal; a processer for extracting a voice component originated by a sender and an ambient noise component originated by the ambient noise, determining the ratio between the voice component and the ambient noise component, and adjusting the amplitude of the received voice signal in accordance with the ratio; and a speaker for outputting a reception voice corresponding to the adjusted reception voice signal.

Abstract: A sound encoder enabling prevention of deterioration of the sound quality of a reproduced signal even if the harmonic structure is broken in a part of the sound signal. The filter state position determining section (111) of the sound encoder judges the noise characteristic of the first-layer decoding spectrum and thereby determines the band of the first-layer decoding spectrum to be used to set the filter state. A filter state setting section (112) sets the first-layer decoding spectrum contained in the determined band out of the first-layer decoding spectrum as the filter state. A filtering section (113) performs filtering of the first-layer decoding spectrum according to the set filter state and the pitch coefficient and computes an estimate spectrum of the input spectrum. An optimal pitch coefficient is determined by a closed loop processing from the filtering section (113) through a search section (114) to a filter information setting section (115).

Abstract: A method and apparatus for concealing frame loss and an apparatus for transmitting and receiving a speech signal that are capable of reducing speech quality degradation caused by packet loss are provided. In the method, when loss of a current received frame occurs, a random excitation signal having the highest correlation with a periodic excitation signal (i.e., a pitch excitation signal) decoded from a previous frame received without loss is used as a noise excitation signal to recover an excitation signal of a current lost frame. Furthermore, a third, new attenuation constant (AS) is obtained by summing a first attenuation constant (NS) obtained based on the number of continuously lost frames and a second attenuation constant (PS) predicted in consideration of change in amplitude of previously received frames to adjust the amplitude of the recovered excitation signal for the current lost frame.

Abstract: An audio/speech sender and an audio/speech receiver and methods thereof. The audio/speech sender comprising a core encoder adapted to encode a core frequency band of an input audio/speech signal having a first sampling frequency, wherein the core frequency band comprises frequencies up to a cut-off frequency. The audio/speech sender further comprises a segmentation device adapted to perform a segmentation of the input audio/speech signal into a plurality of segments, a cut-off frequency estimator adapted to estimate a cut-off frequency for each segment and adapted to transmit information about the estimated cut-off frequency to a decoder, a low-pass filter adapted to filter each segment at said estimated cut-off frequency, and a re-sampler adapted to resample the filtered segments with a second sampling frequency that is related to said cut-off frequency in order to generate an audio/speech frame to be encoded by said core encoder.

Abstract: A system enhances speech by detecting a speaker's utterance through a first microphone positioned a first distance from a source of interference. A second microphone may detect the speaker's utterance at a different position. A monitoring device may estimate the power level of a first microphone signal. A synthesizer may synthesize part of the first microphone signal by processing the second microphone signal. The synthesis may occur when power level is below a predetermined level.

Abstract: A method of and apparatus for detecting noise are provided. The method of detecting noise includes: receiving an input of a voice frame and converting the voice frame into a filter bank vector; converting the converted filter bank vector into band data; calculating a weight Gaussian mixture model (GMM) for each band by using the converted band data; and detecting noise in the voice frame based on the calculation result.

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 system and a method for automobile noise suppression in an automobile are provided. The system comprises a processor and a noise suppression device. The noise suppression device is configured for receiving a voice signal, which includes a speech signal and a noise signal. The processor is configured for determining an adjusting parameter set according to an automobile speed signal corresponding to a speed of the automobile. The noise suppression device can suppress the noise signal according to the adjusting parameter set, whereby enhancing the voice quality.

Abstract: An apparatus for adjusting a prompt voice depending on an environment comprises a receiver module used for receiving a background sound, an analyzer module generating a control signal according to the background sound and an output module adjusting an output frequency of a prompt voice through the control signal and outputting the adjusted prompt voice.

Abstract: According to the invention, an excitation signal is generated as a result of sampled excitation values in order to excite an audio synthesis filter, the generated sampled excitation values being continuously stored in an adaptive codebook. A noise generator is provided which continuously generates random sampled values. A sequence of the stored sampled excitation values is selected from the adaptive codebook based on a fed audio fundamental frequency parameter by means of which a time gap between the sequence that is to be selected and the actual time reference is predefined. The excitation signal is generated by mixing the selected sequence with a random sequence encompassing actual random sampled valued of the noise generator.

Abstract: A method of estimating the reverberations in an acoustic signal (y) comprises the steps of determining the frequency spectrum (Y) of the signal (y), providing a first parameter (?) indicative of the decay of the reverberations part (r) of the signal over time, and providing a second parameter (?) indicative of the amplitude of the direct part (d) of the signal relative to the reverberations part (r). An estimated frequency spectrum ({hacek over (R)}) of the reverberations signal (r) is produced using the frequency spectrum (Y) of a previous frame, the first parameter (?), and the second parameter (?). The second parameter (?). The second parameter (?) is preferably inversely proportional to the early-to-late ratio of the signal (y).

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: Methods and apparatus for post-processing of speech signals are disclosed herein.

Abstract: When there are missing voice-transmission-signals, a repetition-section calculating unit sets a plurality of repetition sections of different lengths that are determined to be similar to the voice-transmission-signals preceding the missing voice-transmission-signal, the repetition sections being determined with respect to stationary voice-transmission-signals stored in a normal signal storage unit, the stationary voice-transmission-signals being selected from the previously input voice-transmission-signals. A controller generates a concealment signal using the repetition sections.

Abstract: An apparatus for adaptively suppressing noise in an input signal frequency spectrum derived from overlapping input frames is provided. The system includes a psychoacoustic power computation module configured to compute a noisy signal power in psychoacoustic bands, a voice activity scoring module configured to compute a probabilistic score for a presence of a speech, and a noise estimation module configured to estimate a noise power in the psychoacoustic bands based on information of past frames, the probabilistic score, and the computed noisy signal power. The system also includes a gain computation module configured to compute a gain for each frequency, based on a probabilistic heuristic, the probabilistic score and the information on the past frames, and a gain post-processing module configured to perform a gain time smoothing, a gain frequency smoothing, and a gain regulation for the computed gain.

Abstract: Provided is a signal distortion elimination apparatus comprising: an inverse filter application means that outputs the signal obtained by applying an inverse filter to an observed signal as a restored signal when a predetermined iteration termination condition is met and outputs the signal obtained by applying the inverse filter to the observed signal as an ad-hoc signal when the predetermined iteration termination condition is not met; a prediction error filter calculation means that segments the ad-hoc signal into frames and outputs a prediction error filter of each frame obtained by performing linear prediction analysis of the ad-hoc signal of each frame; an inverse filter calculation means that calculates an inverse filter such that a concatenation of innovation estimates of the respective frames becomes mutually independent among their samples, where the innovation estimate of a single frame (an innovation estimate) is the signal obtained by applying the prediction error filter of the corresponding frame

Abstract: A method of extracting a voice command produced in an enclosed or partially enclosed environment, includes providing an impulse response signal of the enclosed or partially enclosed environment; recording the voice command and ambient sounds; and using the impulse response signal to extract the recorded voice command.

Abstract: In a pitch conversion method and device which can reduce data throughput while suppressing a degradation of sound quality due to a pitch conversion as much as possible, an input signal pitch pattern per predetermined processing unit and a target pitch pattern are inputted, and a degradation degree indicating how a waveform of the input signal degrades upon pitch conversion from the input signal pitch pattern to the target pitch pattern is calculated. Alternatively, a degradation degree corresponding to a voice state and a phonemic type of the input signal is extracted from a database in which all of combinations of voice states and phonemic types estimated are associated with the degradation degrees to be recorded. Then, a pitch converter which performs a pitch conversion with small data throughput and a pitch converter which performs a pitch conversion with large data throughput are switched over depending on the degradation degree.

Abstract: A duplexer for a communication device having a transmission unit, a reception unit and a shared antenna is disclosed. The duplexer comprises a first signal path between the transmission unit and the reception unit, the first signal path comprises a filter unit filtering the reception signal, the filtering unit provides a filtered signal to the reception unit. The duplexer further comprises a second signal path between the transmission unit and the reception unit. The second signal path comprises a cancellation unit which receives a sample of the transmission signal and produces a compensation signal. The injection of the compensation signal to the first signal path reduces the leakage signal, thereby producing a substantially leakage-free reception signal.

Abstract: There is disclosed a method and apparatus for generating a control signal for processing a speech signal comprising the steps of: adjusting the signal relative to a threshold level; and responsive to detection of a falling edge of the signal, holding the signal level for a holding period. The technique further comprises ‘slowing’ each rising edge of the signal. The technique further comprises attenuating each falling edge of the signal. The steps are carried out on a signal representing the envelope of the speech signal.