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Post-transmission

Subclass of:

704 - Data processing: speech signal processing, linguistics, language translation, and audio compression/decompression

704200000 - SPEECH SIGNAL PROCESSING

704201000 - For storage or transmission

704226000 - Noise

Patent class list (only not empty are listed)

Deeper subclasses:

Class / Patent application numberDescriptionNumber of patent applications / Date published
704228000 Post-transmission 15
20100088092Method 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 S04-08-2010
20090144056Method 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
20090055171BUZZ 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
20100241427LIMITATION 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 (S09-23-2010
20110246192Speech 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
20090216527POST 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
20100063809DOUBLE 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
20110264450SPEECH CAPTURING AND SPEECH RENDERING - The invention proposes extracting one or more speech signals (10-27-2011
20120022860Speech 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
20110066430Robust 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
20110066429VOICE ACTIVITY DETECTOR AND A METHOD OF OPERATION - A voice activity detector (03-17-2011
20120123775Post-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
20120123774APPARATUS, 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
20120095760APPARATUS, 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
20130024194SPEECH 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

Patent applications in class Post-transmission