Class / Patent application number | Description | Number of patent applications / Date published |
704227000 | Pretransmission | 23 |
20080201138 | Headset for Separation of Speech Signals in a Noisy Environment - A headset is constructed to generate an acoustically distinct speech signal in a noisy acoustic environment. The headset positions a pair of spaced-apart microphones near a user's mouth. The microphones each receive the user s speech, and also receive acoustic environmental noise. The microphone signals, which have both a noise and information component, are received into a separation process. The separation process generates a speech signal that has a substantial reduced noise component. The speech signal is then processed for transmission. In one example, the transmission process includes sending the speech signal to a local control module using a Bluetooth radio. | 08-21-2008 |
20080243497 | STATIONARY-TONES INTERFERENCE CANCELLATION - An “Interference Canceller” provides a computationally efficient real-time technique for removing stationary-tone interference from signals. Typical sources of stationary tone contamination of signals include noise from power wiring (i.e., 50/60 Hz or 400 Hz and their harmonics), frame or line frequencies from electronic devices, and noise from computer fans, hard disk drives, etc. In general, the Interference Canceller adaptively builds and updates a model of stationary tone interference in consecutive frames of an input signal. This adaptively updated model is then used to extrapolate and subtract noise from subsequent frames of the input signal to generate a “clean” output signal. This output signal exhibits significant attenuation of stationary tone interference without eliminating important portions of the underlying signal or distorting the underlying signal with artifacts such as musical noise or nonlinear distortions. The Interference Canceller is applicable for use either alone, or as pre-processor to conventional noise suppression. | 10-02-2008 |
20080249769 | Method and Apparatus for Determining Audio Spatial Quality - Techniques for evaluating the audio quality of an audio test signal are disclosed. These techniques provide a quality analysis that takes into account spatial audio distortions between the audio test signal and a reference audio signal. These techniques involve, for example, determining a plurality of audio spatial cues for an audio test signal, determining a corresponding plurality of audio spatial cues for an audio reference signal, comparing the determined audio spatial cues of the audio test signal to the audio spatial cues of the audio reference signal, and determining the audio quality of the audio test signal. | 10-09-2008 |
20090306977 | SPEECH RECOGNITION DEVICE, SPEECH RECOGNITION METHOD, COMPUTER-EXECUTABLE PROGRAM FOR CAUSING COMPUTER TO EXECUTE RECOGNITION METHOD, AND STORAGE MEDIUM - A speech recognition device and method configured to include a computer, for recognizing speech, including: a storage location for storing a feature quantity acquired from a speech signal for each frame; storage portions for storing acoustic model data and language model data; a echo speech component for generating echo speech model data from a speech signal acquired prior to a speech signal to be processed at the current time point and using the echo speech model data to generate adapted acoustic model data; and a processing component for utilizing the feature quantity, the adapted acoustic model data, and the language model data to provide a speech recognition result of the speech signal. | 12-10-2009 |
20090313010 | AUTOMATIC PLAYBACK OF A SPEECH SEGMENT FOR MEDIA DEVICES CAPABLE OF PAUSING A MEDIA STREAM IN RESPONSE TO ENVIRONMENTAL CUES - 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. | 12-17-2009 |
20100153102 | SCALABLE CODING APPARATUS AND SCALABLE CODING METHOD - A scalable coding apparatus is provided to suppress deterioration of a quality of a coded signal in a normal frame next to a frame compensated for the occurrence of a data loss. The scalable coding apparatus is provided with a core-layer coding section ( | 06-17-2010 |
20100274562 | SYSTEM AND METHOD FOR TRANSMITTING VOICE INPUT FROM A REMOTE LOCATION OVER A WIRELESS DATA CHANNEL - A system and method for improving voice recognition processing at a server system that receives voice input from a remotely located user system. The user system includes a microphone, a processor that performs front-end voice recognition processing of the received user voice input, and a communication component configured to send the front-end processed user voice input to a destination wirelessly over a network. The server system includes a communication component configured to receive the sent front-end processed user voice input, and a processor configured to complete voice recognition processing of the sent front-end processed user voice input. | 10-28-2010 |
20110119056 | SUBWORDS CODING USING DIFFERENT INTERLEAVING SCHEMES - In a communications system that demultiplexes user data words into multiple sub-words for encoding and decoding within different subword-processing paths, the minimum distance between bit errors in an extrinsic codeword can be increased by having corresponding interleavers/deinterleavers in the different subword-processing paths use different interleaving/deinterleaving algorithms. | 05-19-2011 |
20120116760 | DEVICE FOR IMPROVING THE INTELLIGIBILITY OF SPEECH IN A MULTI-USER COMMUNICATION SYSTEM - 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. | 05-10-2012 |
20120136656 | Communication System - A method for reducing ringing in a signal output from a filter comprising inputting a signal into a filter; filtering a first portion of the input signal to generate a filtered portion of the output signal; analyzing the filtered portion of the output signal; detecting if ringing is present in the filtered portion of the output signal based on said analysis; and adjusting the filter characteristics to reduce ringing in a subsequent filtered portion of the output signal if it is determined that ringing is present. | 05-31-2012 |
20130246061 | AUTOMATIC REALTIME SPEECH IMPAIRMENT CORRECTION - Automatic correcting of user's speech impairment in speech may include obtaining the audio signal of a given user's speech, and analyzing the obtained audio signal to identify artifacts caused by the user's impairment. The obtained audio signal may be modified by eliminating the identified artifacts from it. The modified audio signal may be provided, e.g., to be played or broadcast or transmitted. | 09-19-2013 |
20130297302 | Systems And Methods For Voice Enhancement In Audio Conference - 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. | 11-07-2013 |
20130297303 | SPEECH PROCESSING APPARATUS, CONTROL METHOD THEREOF, STORAGE MEDIUM STORING CONTROL PROGRAM THEREOF, AND VEHICLE, INFORMATION PROCESSING APPARATUS, AND INFORMATION PROCESSING SYSTEM INCLUDING THE SPEECH PROCESSING APPARATUS - 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. | 11-07-2013 |
20140172421 | SPEECH ENHANCING METHOD, DEVICE FOR COMMUNICATION EARPHONE AND NOISE REDUCING COMMUNICATION EARPHONE - 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. | 06-19-2014 |
20140372110 | VOIC CALL ENHANCEMENT - 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. | 12-18-2014 |
20140372111 | VOICE RECOGNITION ENHANCEMENT - 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. | 12-18-2014 |
20150332704 | Method for Controlling Acoustic Echo Cancellation and Audio Processing Apparatus - A method for controlling acoustic echo cancellation and an audio processing apparatus are described. In one embodiment, the audio processing apparatus includes an acoustic echo canceller for suppressing acoustic echo in a microphone signal, a jitter buffer for reducing delay jitter of a received signal, and a joint controller for controlling the acoustic echo canceller by referring to at least one future frame in the jitter buffer. | 11-19-2015 |
20150340049 | ACOUSTIC ECHO CANCELLATION (AEC) FOR A CLOSE-COUPLED SPEAKER AND MICROPHONE SYSTEM - Embodiments are directed towards providing acoustic echo cancellation in a closely-coupled microphone/speaker system. A speaker may produce an audible signal from a reference signal, which may be captured with a microphone. Full band cancellation (FBC) may modify the captured signal to suppress an echo of the reference signal caused by a direct acoustic path between the microphone and speaker. FBC may include a fixed filter and an adaptive filter. The fixed filter may modify the captured signal based on the reference signal. The adaptive filter may automatically adapt based on the captured signal and the reference signal. If a comparison of a performance of the adaptive filter and the fixed filter is above a threshold, then the fixed filter may be updated based on the adaptive filter. Subband acoustic echo cancellation may generate an output signal that suppresses residual echoes of the reference signal based on the modified signal. | 11-26-2015 |
20150380010 | METHOD AND APPARATUS FOR GENERATING A SPEECH SIGNAL - An apparatus comprises microphone receivers ( | 12-31-2015 |
20160019904 | Adaptive Vehicle State-Based Hands-Free Phone Noise Reduction With Learning Capability - This disclosure generally relates to a system, apparatus, and method for achieving an adaptive vehicle state-based hands free noise reduction feature. A noise reduction tool is provided for adaptively applying a noise reduction strategy on a sound input that uses feedback speech quality measures and machine learning to develop future noise reduction strategies, where the noise reduction strategies include analyzing vehicle operational state information and external information that are predicted to contribute to cabin noise and selecting noise reducing pre-filter options based on the analysis. | 01-21-2016 |
20160064008 | SYSTEMS AND METHODS FOR NOISE REDUCTION USING SPEECH RECOGNITION AND SPEECH SYNTHESIS - The present disclosure describes a system ( | 03-03-2016 |
20160064010 | METHOD AND APPARATUS FOR ELIMINATING MUSIC NOISE VIA A NONLINEAR ATTENUATION/GAIN FUNCTION - A system including first and second gain modules, an operator module, and a priori and posteriori modules. The first gain module applies a non-linear function to generate a gain signal based on an amplitude of a first speech signal and an estimated a priori variance of noise included in the first speech signal. The operator module generates an operator based on the gain signal and the estimated a priori variance of noise. The a priori module determines an a priori signal-to-noise ratio based on the operator. The posteriori module determines a posteriori signal-to-noise ratio based on the amplitude of the first speech signal and (ii) the estimated a priori variance of noise. The second gain module: determines a gain value based on the a priori signal-to-noise ratio and the a posteriori signal-to-noise ratio; and generates, based on the amplitude of the first speech signal and the gain value, a second speech signal that corresponds to an estimate of an amplitude of the first speech signal, where the second speech signal is substantially void of music noise. | 03-03-2016 |
20160189726 | MECHANISM FOR FACILITATING DYNAMIC ADJUSTMENT OF AUDIO INPUT/OUTPUT (I/O) SETTING DEVICES AT CONFERENCING COMPUTING DEVICES - A mechanism is described for facilitating dynamic adjustment of audio input/output setting devices at conferencing computing devices according to one embodiment. A method of embodiments, as described herein, includes maintaining awareness of proximity between a plurality of computing devices participating in a conference, detecting audio disturbance relating to the plurality of computing devices, and calculating adjustments to settings of one or more audio input/output (I/O) devices coupled to one or more of the plurality of computing devices to eliminate the audio disturbance. The adjustments may be dynamically applied to the settings of the one or more audio I/O devices. | 06-30-2016 |