Class / Patent application number | Description | Number of patent applications / Date published |
704228000 | Post-transmission | 29 |
20090055171 | BUZZ REDUCTION FOR LOW-COMPLEXITY FRAME ERASURE CONCEALMENT - A system is described that performs periodic waveform extrapolation based frame erasure concealment (FEC) to generate frames of an output speech signal corresponding to erased frames of encoded bit-stream in a manner reduces buzzy and tonal artifacts in the output speech signal. An embodiment of the invention uses a multiple of a pitch period associated with previously-decoded speech to perform periodic waveform extrapolation for consecutively-erased frames in a frame erasure beyond the first erased frame. An embodiment of the invention also attenuates the extrapolated signal after a threshold number of erased frames so as to reduce the FEC output signal to zero, wherein the threshold number of erased frames is dependent at least in part on the pitch period associated with the previously-decoded speech. | 02-26-2009 |
20090144056 | Method and computer program product for generating recognition error correction information - A method for providing recognition error correction information, the method includes: obtaining metadata associated with a capture of a media item; and generating recognition error correction information in response to the metadata. The recognition error correction information is to be used in a recognition process selected out of a list consisting of an automatic speech recognition process and an optical characters recognition process. | 06-04-2009 |
20090216527 | POST FILTER, DECODER, AND POST FILTERING METHOD - A post filter and a decoder enabling improvement of the sound quality of a decoded signal even when the sound quality of the decoded signal is different from the bands are disclosed. A frequency converting section determines a decoded spectrum. A power spectrum computing section computes the power spectrum from the decoded spectrum. A correction band determining section determines the band in which the power spectrum is corrected according to layer information. A power spectrum correcting section corrects the power spectrum in the corrected band in such a way that the variation along the frequency axis is suppressed. An inverse converting section subjects the corrected power spectrum to inverse conversion to determine an autocorrelation function. An LPC analyzing section determines an LPC coefficient of the determined autocorrelation function. | 08-27-2009 |
20100063809 | DOUBLE TALK DETECTOR - 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. | 03-11-2010 |
20100088092 | Method and Arrangement for Controlling Smoothing of Stationary Background Noise - In a method of smoothing stationary background noise in a telecommunication speech session, initially receiving and decoding S | 04-08-2010 |
20100241427 | LIMITATION OF DISTORTION INTRODUCED BY A POST-PROCESSING STEP DURING DIGITAL SIGNAL DECODING - 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 (S | 09-23-2010 |
20110066429 | VOICE ACTIVITY DETECTOR AND A METHOD OF OPERATION - A voice activity detector ( | 03-17-2011 |
20110066430 | Robust Noise Estimation - An enhancement system improves the estimate of noise from a received signal. The system includes a spectrum monitor that divides a portion of the signal at more than one frequency resolution. Adaptation logic derives a noise adaptation factor of the received signal. A plurality of devices tracks the characteristics of an estimated noise in the received signal and modifies multiple noise adaptation rates. Weighting logic applies the modified noise adaptation rates derived from the signal divided at a first frequency resolution to the signal divided at a second frequency resolution. | 03-17-2011 |
20110246192 | Speech Quality Evaluation System and Storage Medium Readable by Computer Therefor - In prediction of a speech quality evaluation score such as a phone speech, even when a background noise exists, a subjective opinion score is predicted with high precision. A speech quality evaluation system that outputs a predicted value of the subjective opinion score for an evaluation speech such as a far-end speech of a phone, includes a speech distortion calculation unit conducts, after calculating frequency characteristics of the evaluation speech, a process of subtracting given frequency characteristics from frequency characteristics of the evaluation speech, and calculates the speech distortion on the basis of the frequency characteristics after the subtracting process has been conducted, and a subjective evaluation prediction unit that calculates the predicted value of the subjective opinion score on the basis of the speech distortion. | 10-06-2011 |
20110264450 | SPEECH CAPTURING AND SPEECH RENDERING - The invention proposes extracting one or more speech signals ( | 10-27-2011 |
20120022860 | Speech and Noise Models for Speech Recognition - An audio signal generated by a device based on audio input from a user may be received. The audio signal may include at least a user audio portion that corresponds to one or more user utterances recorded by the device. A user speech model associated with the user may be accessed and a determination may be made background audio in the audio signal is below a defined threshold. In response to determining that the background audio in the audio signal is below the defined threshold, the accessed user speech model may be adapted based on the audio signal to generate an adapted user speech model that models speech characteristics of the user. Noise compensation may be performed on the received audio signal using the adapted user speech model to generate a filtered audio signal with reduced background audio compared to the received audio signal. | 01-26-2012 |
20120095760 | APPARATUS, A METHOD AND A COMPUTER PROGRAM FOR CODING - Various embodiments of the invention provide scalable and distributed input signal coding activity detection and coding thereof (e.g. VAD/DTX) processing framework. An apparatus comprising an encoder is shown. The apparatus can be a terminal, for example a mobile phone, computer or the like. The apparatus may act as transmitter etc. | 04-19-2012 |
20120123774 | APPARATUS, ELECTRONIC APPARATUS AND METHOD FOR ADJUSTING JITTER BUFFER - An apparatus, electronic apparatus and method for adjusting jitter buffer is provided. A previous jitter buffer size based on a jitter buffer size determined according to an adaptive jitter buffer size calculation algorithm is applied in predicting a jitter buffer size of future time such that the predicted jitter buffer size is applied to obtain a jitter buffer size of a valid time. The audio quality of the speech transmitted over a packet switched network is enhanced. | 05-17-2012 |
20120123775 | Post-noise suppression processing to improve voice quality - Provided are methods and systems for improving quality of speech communications. The method may be for improving quality of speech communications in a system having a speech encoder configured to encode a first audio signal using a first set of encoding parameters associated with a first noise suppressor. A method may involve receiving a second audio signal at a second noise suppressor which provides much higher quality noise suppression than the first noise suppressor. The second audio signal may be generated by a single microphone or a combination of multiple microphones. The second noise suppressor may suppress the noise in the second audio signal to generate a processed signal which may be sent to a speech encoder. A second set of encoding parameters may be provided by the second noise suppressor for use by the speech encoder when encoding the processed signal into corresponding data. | 05-17-2012 |
20130024194 | SPEECH ENHANCING METHOD AND DEVICE, AND NENOISING COMMUNICATION HEADPHONE ENHANCING METHOD AND DEVICE, AND DENOISING COMMUNICATION HEADPHONES - The present invention discloses a speech enhancing method, a speech enhancing device and a denoising communication headphone. In the solutions of the present invention, a first sound signal that comprises a user's speech signal transmitted through coupling vibration and an ambient noise signal transmitted through the air and a second sound signal that is mainly an ambient noise signal transmitted through the air are picked up by a primary vibration microphone and a secondary vibration microphone, respectively, that have a specific relative positional relationship therebetween, and the ambient noise signals picked up by the two vibration microphones are correlated with each other; a control parameter used to control an updating speed of an adaptive filter is determined according to the first sound signal and the second sound signal; the first sound signal is denoised and filtered according to the second sound signal and the control parameter; and the denoised and filtered speech signal is further denoised and speech high-frequency enhancement is performed thereon. The technical solutions of the present invention can effectively improve the signal to noise ratio (SNR) and the quality of speech in an environment of highly intense noises. | 01-24-2013 |
20130151248 | Apparatus, System, and Method For Distinguishing Voice in a Communication Stream - An apparatus for distinguishing a voice is described. In one embodiment, the apparatus includes a server with a communication interface, a frame generator, and a sound analyzer. The communication interface processes an incoming communication stream with an echo canceller to cancel echo in the communication stream. The frame generator operates on a processor and generates a plurality of frames from the communication stream. Each of the plurality of frames contains data for a period of time from the communication stream. The frame generator also assigns a frame value to each of the plurality of frames. The sound analyzer determines a status of the communication stream by analyzing the frame values of the plurality of frames. | 06-13-2013 |
20130191120 | CONSTRAINED SOFT DECISION PACKET LOSS CONCEALMENT - Methods, systems, and apparatuses for performing packet loss concealment are disclosed. In response to determining that an encoded frame representing a segment of a signal is bad, an encoded parameter within the encoded frame is decoded based on bit information (such as soft bit information) associated with the encoded parameter to obtain a decoded parameter. Whether the decoded parameter violates a parameter constraint is determined. If a parameter constraint violation is detected, an estimate of the decoded parameter is generated. Either the decoded parameter or estimate of the decoded parameter is passed to a decoder for use in decoding the encoded frame. | 07-25-2013 |
20140214414 | DYNAMIC AUDIO PROCESSING PARAMETERS WITH AUTOMATIC SPEECH RECOGNITION - A communication system includes a front-end audio gateway or bridge and a hands-free device. An automatic speech recognition platform accessible to the hands-free device provides or makes available one or more preprocessing schemes and/or acoustic models to the front-end audio gateway or bridge. The preprocessing schemes or acoustic models can be identified by or provided before a connection is established between the front-end audio gateway and the automatic speech recognition platform, when a connection occurs between the front-end audio gateway and the automatic speech recognition platform, and/or during a speech recognition session. | 07-31-2014 |
20140236590 | COMMUNICATION APPARATUS AND VOICE PROCESSING METHOD THEREFOR - A voice processing method for use in a communication apparatus, in an embodiment, includes the following steps. A near-end audio signal is received by at least one microphone of the communication apparatus. Voice and noise energy data are generated by performing voice activity detection on the near-end audio signal. A noise amount is obtained by performing noise energy calculation with the noise energy data. Whether the noise amount exceeds a first noise amount threshold is determined. If the noise amount exceeds the first noise amount threshold, a sidetone mode of the communication apparatus is enabled to produce a sidetone signal according to the voice energy data and play the sidetone signal through a speaker thereof. A noise suppression mode is enabled to produce a far-end audio signal according to the voice energy data and transmitting the far-end audio signal by a communication module of the communication apparatus. | 08-21-2014 |
20140249810 | SIGNAL NOISE ATTENUATION - A noise attenuation apparatus receives a first signal comprising a desired and a noise signal component. Two codebooks ( | 09-04-2014 |
20140288927 | Procedure and Mechanism for Controlling and Using Voice Communication - In a method and system for controlling voice communication of a first person with at least a second person via a communication network a first microphone receives and converts vocal utterances from the first person to a voice signal. A first processor generates a transmission signal by processing the voice signal. A transmitter sends the transmission signal to a receiver. The receiver generates a listening signal by processing the received signal and transmits the listening signal to a speaker. The speaker converts the listening signal to an acoustic signal to be perceived by the first person. In this method a second processor generates the listening signal from the received signal by branching the voice signal and adding the branched voice signal to the received signal. The branched voice signal may be subjected to variable attenuation and/or amplification before being added to the branched voice signal to the received signal. | 09-25-2014 |
20140337021 | SYSTEMS AND METHODS FOR NOISE CHARACTERISTIC DEPENDENT SPEECH ENHANCEMENT - A method for noise characteristic dependent speech enhancement by an electronic device is described. The method includes determining a noise characteristic of input audio. Determining a noise characteristic of input audio includes determining whether noise is stationary noise and determining whether the noise is music noise. The method also includes determining a noise reference based on the noise characteristic. Determining the noise reference includes excluding a spatial noise reference from the noise reference when the noise is stationary noise and including the spatial noise reference in the noise reference when the noise is not music noise and is not stationary noise. The method further includes performing noise suppression based on the noise characteristic. | 11-13-2014 |
20150100310 | APPARATUS AND METHOD OF REDUCING NOISE AND AUDIO PLAYING APPARATUS WITH NON-MAGNET SPEAKER - An audio apparatus is provided. The audio apparatus includes an input configured to receive an audio signal containing noise; a period estimation unit configured to estimate a period of a noise pattern in the audio signal; a noise reducer configured to subtract and remove the noise pattern from the audio signal in a frequency domain by using the estimated period of the noise pattern; a noise updater configured to update the noise pattern according to a change in amplitude of the noise; and an output configured to output the audio signal obtained by removing the noise pattern. | 04-09-2015 |
20160019906 | SIGNAL PROCESSOR AND METHOD THEREFOR - A signal processor is configured to suppress a noise component contained in an input voice signal by means of coherence filtering. The processor includes an iterative coherence filtering function for repeatedly conducting the coherence filtering on a signal, which has been subjected to the coherence filtering and then input to the processor again as input signal, and performs the iteration processing on the signal obtained by the coherence filtering until a condition for terminating the iteration is satisfied, thereby preventing musical noise from generating while a noise component is suppressed. | 01-21-2016 |
20160027447 | SPATIAL COMFORT NOISE - A method, an apparatus, logic (e.g., executable instructions encoded in a non-transitory computer-readable medium to carry out a method), and a non-transitory computer-readable medium configured with such instructions. The method is to generate and spatially render spatial comfort noise at a receiving endpoint of a conference system, such that the comfort noise has target spectral characteristics typical of comfort noise, and at least one spatial property that at least substantially matches at least one target spatial property. On version includes receiving one or more or more audio signals from other endpoints, combining the received audio signals with the spatial comfort noise signals, and rendering the combination of the received audio signals and the spatial comfort noise signals to a set of output signals for loudspeakers, such that the spatial comfort noise signals are continually in the output signal sin addition to output from the received audio signals. | 01-28-2016 |
20160035367 | SPEECH DEREVERBERATION METHODS, DEVICES AND SYSTEMS - Improved audio data processing method and systems are provided. Some implementations involve dividing frequency domain audio data into a plurality of subbands and determining amplitude modulation signal values for each of the plurality of subbands. A band-pass filter may be applied to the amplitude modulation signal values in each subband, to produce band-pass filtered amplitude modulation signal values for each subband. The band-pass filter may have a central frequency that exceeds an average cadence of human speech. A gain may be determined for each subband based, at least in part, on a function of the amplitude modulation signal values and the band-pass filtered amplitude modulation signal values. The determined gain may be applied to each subband. | 02-04-2016 |
20160086617 | SYSTEM AND METHOD FOR ADDRESSING DISCONTINUOUS TRANSMISSION IN A NETWORK DEVICE - Embodiments included herein are directed towards a system and method for addressing discontinuous transmission (DTX) in a network device. Embodiments may include receiving, at a computing device, an audio signal and generating at least one silence descriptor (SID) frame associated with the audio signal. Embodiments may also include generating at least one no data frame associated with the audio signal. Embodiments may also include initiating a speech decoder, voice enhancement, and speech encoder operation for the at least one SID frame during a DTX operation and bypassing the speech decoder, voice enhancement, and speech encoder functions for the at least one no data frame. | 03-24-2016 |
20160118052 | VOCODER PROCESSING METHOD, SEMICONDUCTOR DEVICE, AND ELECTRONIC DEVICE - In a semiconductor device, a vocoder processing unit requests, after executing a first vocoder process being one of an encoding process and a decoding process and before executing a following second vocoder process being other one of the encoding process and the decoding process, a cache memory to prefetch first program data to be used for the second vocoder process from an external memory. | 04-28-2016 |
20160155446 | METHOD AND APPARATUS FOR CONTROLLING AUDIO FRAME LOSS CONCEALMENT | 06-02-2016 |