Abstract: A method includes detecting whether there is a very short pitch lag in a speech or audio signal that is shorter than a conventional minimum pitch limitation using a combination of time domain and frequency domain pitch detection techniques. The pitch detection techniques include using pitch correlations in a time domain and detecting a lack of low frequency energy in the speech or audio signal in a frequency domain. The detected very short pitch lag is coded using a pitch range from a predetermined minimum very short pitch limitation that is smaller than the conventional minimum pitch limitation.

Abstract: A system and method are presented for optimization of audio fingerprint search. In an embodiment, the audio fingerprints are organized into a recursive tree with different branches containing fingerprint sets that are dissimilar to each other. The tree is constructed using a clustering algorithm based on a similarity measure. The similarity measure may comprise a Hamming distance for a binary fingerprint or a Euclidean distance for continuous valued fingerprints. In another embodiment, each fingerprint is stored at a plurality of resolutions and clustering is performed hierarchically. The recognition of an incoming fingerprint begins from the root of the tree and proceeds down its branches until a match or mismatch is declared. In yet another embodiment, a fingerprint definition is generalized to include more detailed audio information than in the previous definition.

Abstract: Features are disclosed for injection services that allow a development team to quickly and easily include functionality developed by other teams. The main application server injects functionality into responses. The injected service content may include executable content (e.g., scripts) which may be retrieved from a content distribution network. This provides a framework for integrating various, decoupled features into a single main application.

Abstract: Various implementations described herein relate to systems and methods for encoding data having input bits to be stored in a non-volatile storage device, including mapping the input bits to a plurality of component codes of an error correction code (ECC) and encoding the input bits as the plurality of component codes, wherein first input bits of the input bits encoded by any of the plurality of component codes are encoded by every other component code of the plurality of component codes in a non-overlapping manner.

Abstract: A training apparatus includes an autoregressive model configured to estimate a current signal from a past signal sequence and a current context label, a vocal tract feature analyzer configured to analyze an input speech signal to determine a vocal tract filter coefficient representing a vocal tract feature, a residual signal generator configured to output a residual signal, a quantization unit configured to quantize the residual signal output from the residual signal generator to generate a quantized residual signal, and a training controller configured to provide as a condition, a context label of an already known input text for the input speech signal corresponding to the already known input text to the autoregressive model and to train the autoregressive model by bringing a past sequence of the quantized residual signals for the input speech signal and the current context label into correspondence with a current signal of the quantized residual signal.

Abstract: Embodiments include apparatuses, systems, and methods for a computer-aided or autonomous driving (CA/AD) system to identify and respond to an audio signal, e.g., an emergency alarm signal. In embodiments, the CA/AD driving system may include a plurality of microphones disposed to capture the audio signal included in surrounding sounds to a semi-autonomous or autonomous (SA/AD) vehicle. In embodiments, an audio analysis unit may receive the audio signal to extract audio features from the audio signal. In embodiments, a neural network such as a Deep Neural Network (DNN) may receive the extracted audio features from the audio analysis unit and to generate a probability score to allow identification of the audio signal. In embodiments, the CA/AD driving system may control driving elements of the SA/AD vehicle to autonomously or semi-autonomously drive the SA/AD vehicle in response to the identification. Other embodiments may also be described and claimed.

Abstract: It is shown an encoder for encoding an audio signal with reduced background noise using linear predictive coding. The encoder includes a background noise estimator configured to estimate background noise of the audio signal, a background noise reducer configured to generate background noise reduced audio signal by subtracting the estimated background noise of the audio signal from the audio signal, and a predictor configured to subject the audio signal to linear prediction analysis to obtain a first set of linear prediction filter (LPC) coefficients and to subject the background noise reduced audio signal to linear prediction analysis to obtain a second set of linear prediction filter (LPC) coefficients. Furthermore, the encoder includes an analysis filter composed of a cascade of time-domain filters controlled by the obtained first set of LPC coefficients and the obtained second set of LPC coefficients.

Abstract: An electronic transmitter includes: a modulator to phase modulate a carrier signal with a baseband signal using a phase modulation sequence; and an emitter to emit the phase modulated signal. The phase modulated signal has poor autocorrelation, has a corresponding mismatched filter based on the phase modulation sequence, and is configured to demodulate into the baseband signal through poor cross-correlation with the mismatched filter. Sometimes, the transmitter is part of a sensing apparatus, where the emitter emits the phase modulated signal at a target and the emitted signal reflects off the target. The sensing apparatus includes a receiver that has a collector to collect the reflected signal, and a demodulator to demodulate the collected signal into the baseband signal through the poor cross-correlation with the mismatched filter. Sometimes, the transmitter is part of a communication system, where the emitter emits the phase modulated signal to an intended recipient.

Abstract: An encoder and a decoder are provided that are capable of reproducing a frequency-domain envelope sequence that provides high approximation accuracy around peaks caused by the pitch period of an audio signal by using a small amount of code. An encoder of the present invention comprises a periodic-combined-envelope generating part and a variable-length coding part. The periodic-combined-envelope generating part generates a periodic combined envelope sequence which is a frequency-domain sequence based on a spectral envelope sequence which is a frequency-domain sequence corresponding to a linear predictive coefficient code obtained from an input audio signal and on a frequency-domain period. The variable-length coding part encodes a frequency-domain sequence derived from the input audio signal. A decoder of the present invention comprises a periodic-combined-envelope generating part and a variable-length decoding part.

Abstract: An encoder and a decoder are provided that are capable of reproducing a frequency-domain envelope sequence that provides high approximation accuracy around peaks caused by the pitch period of an audio signal by using a small amount of code. An encoder of the present invention comprises a periodic-combined-envelope generating part and a variable-length coding part. The periodic-combined-envelope generating part generates a periodic combined envelope sequence which is a frequency-domain sequence based on a spectral envelope sequence which is a frequency-domain sequence corresponding to a linear predictive coefficient code obtained from an input audio signal and on a frequency-domain period. The variable-length coding part encodes a frequency-domain sequence derived from the input audio signal. A decoder of the present invention comprises a periodic-combined-envelope generating part and a variable-length decoding part.

Abstract: A method includes generating, by a processor, an audio fingerprint representative of an audio signal. The audio fingerprint is based on a plurality of first intensity values corresponding to one or more segments of the audio signal. The plurality of first intensity values are based on a Fast Fourier Transform (FFT) performed on at least one sampled segment of the audio signal. The method also includes comparing a plurality of second intensity values based on a recorded sound to determine whether the second intensity values match the first intensity values. The method additionally includes causing a message to be communicated to a device used to record the sound based on a determination that the plurality of second intensity values match the plurality of first intensity values.

Abstract: Provided are a signal processing method for determining an audience rating of media, and an additional information inserting apparatus, a media reproducing apparatus and an audience rating determining apparatus for performing the same method. In detail, the signal processing method for determining an audience rating of media is a method that may determine an audience rating of media with respect to a whole section of an audio signal by inserting additional information into a silence section through a noise signal.

Abstract: An apparatus for determining an estimated pitch lag is provided. The apparatus includes an input interface for receiving a plurality of original pitch lag values, and a pitch lag estimator for estimating the estimated pitch lag. The pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to the original pitch lag value.

Abstract: A method and apparatus to encoding or decoding an audio signal is provided. In the method and apparatus, a noise-floor level to use in encoding or decoding a high frequency signal is updated according to the degree of a voiced or unvoiced sound included in the signal.

Abstract: An encoding method, a decoding method, an encoding apparatus, a decoding apparatus, a transmitter, a receiver, and a communications system. The encoding method includes: dividing a to-be-encoded time-domain signal into a low band signal and a high band signal; performing encoding on the low band signal to obtain a low frequency encoding parameter; performing encoding on the high band signal to obtain a high frequency encoding parameter, and obtaining a synthesized high band signal; performing short-time post-filtering processing on the synthesized high band signal to obtain a short-time filtering signal; and calculating a high frequency gain based on the high band signal and the short-time filtering signal. A technical solution according to the embodiments of the present application can improve an encoding and/or decoding effect.

Abstract: An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm is provided. The apparatus includes a matrix determiner for determining an autocorrelation matrix R, and a codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R. The matrix determiner is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R includes a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein R(i, j)=r(|i?j|), wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R.

Abstract: The present invention relates to the continuous monitoring of an audio signal and identification of audio items within an audio signal. The technology disclosed utilizes predictive caching of fingerprints to improve efficiency. Fingerprints are cached for tracking an audio signal with known alignment and for watching an audio signal without known alignment, based on already identified fingerprints extracted from the audio signal. Software running on a smart phone or other battery-powered device cooperates with software running on an audio identification server.

Abstract: Provided is a method, non-transitory computer program product and system for an improved noise suppression technique for speech enhancement. It operates on speech signals from a single or multiple input sources. Background noise monitoring is performed with one or multiple input speech signals to determine if the input speech contains active voice. If the absence of active voice is detected, the spectrum of the input speech is used to update a long-term noise spectrum estimate. In addition, the input from one or more secondary microphones can be used to update a short-term noise spectrum estimate. The input speech spectrum is then compared to the long-term and/or short-term noise spectra, and a selective spectrum gain based shaping is applied to the input speech spectrum to reduce noise.

Abstract: Systems and methods for transmitting content to a client via a communication network are provided. An insertion server, running within a firewall device associated with a private IP network, detects establishment of a transport communication protocol connection between a client associated with the network and a destination located external to the network by examining packets as they pass through the network and pass by the insertion server. A content request of an application protocol initiated by the client and directed to the destination is observed by the insertion server. The content request is negated by the insertion server by causing a canceling message of the transport communication protocol to be sent to the destination. Unsolicited content is caused to be selected for delivery to the client by the insertion server. The unsolicited content is sent by the insertion server to the client via the application protocol.

Abstract: Vector Quantizer and method therein for efficient vector quantization, e.g. in a transform audio codec. The method comprises comparing an input target vector s with a plurality of centroids, each centroid representing a respective class of codevectors in a codebook. Further, a starting point for a search related to the input target vector in the codebook is determined, based on the result of the comparison. The codevectors in the codebook are sorted according to a distortion measure reflecting the distance between each codevector and the centroids of the classes. The Vector Quantizer and method enables that the class of codevectors comprising the most probable candidate codevectors in regard of the input vector s may be searched first.

Abstract: A method of detecting camera defect includes: obtaining an image by a processing unit, the processing unit having a surface fit module, a subtraction module, and a peak quantification module; determining a first autocorrelation map for a first sub-region in the image; determining, using the surface fit module, a first surface fit for first scene content in the first sub-region; subtracting, using the subtraction module, the first surface fit from the first autocorrelation map for the first sub-region in the image to obtain a first residual map; and quantifying, using the peak quantification module, a first noise in the first residual map.

Abstract: A speech signal processing apparatus and a speech signal processing method for enhancing speech intelligibility are provided. The speech signal processing apparatus includes an input signal gain determiner to determine a gain of an input signal based on a harmonic characteristic of a voiced speech, a voiced speech output unit to output a voiced speech in which a harmonic component is preserved by applying the gain to the input signal, a linear predictive coefficient determiner to determine a linear predictive coefficient based on the voiced speech, and an unvoiced speech preserver to preserve an unvoiced speech of the input signal based on the linear predictive coefficient.

Abstract: A device may receive data indicative of a plurality of speech sounds associated with first voice characteristics of a first voice. The device may receive an input indicative of speech associated with second voice characteristics of a second voice. The device may map at least one portion of the speech of the second voice to one or more speech sounds of the plurality of speech sounds of the first voice. The device may compare the first voice characteristics with the second voice characteristics based on the map. The comparison may include vocal tract characteristics, nasal cavity characteristics, and voicing characteristics. The device may determine a given representation configured to associate the first voice characteristics with the second voice characteristics. The device may provide an output indicative of pronunciations of the one or more speech sounds of the first voice according to the second voice characteristics based on the given representation.

Abstract: Systems and methods for transmitting content to a client via a communication network are provided. According to one embodiment, an insertion server running within a firewall device of a network observes a content request of an application protocol by monitoring or proxying transport communication protocol connections established through the firewall device. The content request is (i) originated by a client device coupled to the network, (ii) directed to a destination device coupled to the network and (iii) associated with one of the multiple transport communication protocol connections. Responsive to observing the content request, the insertion server determines whether one or more conditions are satisfied. If so, the content request is negated by causing a canceling message of the transport communication protocol to be sent to the destination device and unsolicited content is selected and delivered to the client device via the application protocol.

Abstract: In one aspect, a device includes a processor, a display accessible to the processor, and memory accessible to the processor. The memory bears instructions executable by the processor to provide sound corresponding to a portion of text presented on the display with at least one portion of the sound being provided as if originating at least substantially from a location on the display at which the portion of text is presented on the display.

Abstract: A speech is recognized using ACF factors extracted from running autocorrelation functions calculated from the speech. The extracted ACF factors are a W?(0) (width of ACF amplitude around zero-delay origin), a W?(0)max (maximum value of the W?(0)), a ?1 (pitch period), a ?1 (pitch strength), and a ??1/?t (rate of the pitch strength change). Syllables in the speech are identified by comparing the ACF factors with templates stored in a database.

Abstract: A method includes receiving an audio fingerprint from a listening device. The method also includes, in response to determining that a portion of a stored audio fingerprint substantially matches a portion of the received audio fingerprint, identifying a longest unordered match between the received audio fingerprint and the stored audio fingerprint that satisfies a similarity threshold. The method further includes, in response to determining that the identified longest unordered match satisfies a length criterion, detecting a match between the received audio fingerprint and the stored audio fingerprint.

Abstract: A noise suppressing apparatus that calculates a suppression coefficient for suppressing noise of an input signal by using a frequency spectrum of the input signal includes a frequency converting section that converts the input signal into a frequency spectrum; a noise level estimating section that calculates an estimated noise level of the input signal; a weight coefficient calculating section that calculates N (N is 2 or more) weight coefficients at predetermined intervals; and a suppression coefficient calculating section that calculates a joint distribution model of sound by weighting N statistical distribution models with the N weight coefficients, derives an estimation expression for a sound spectrum of the input signal on the basis of posteriori probability using the calculated joint distribution model of sound as priori probability, and calculates the suppression coefficient on the basis of the derived estimation expression and level of the input signal.

Abstract: Vector Quantizer and method therein for efficient vector quantization, e.g. in a transform audio codec. The method comprises comparing an input target vector s with a plurality of centroids, each centroid representing a respective class of codevectors in a codebook. Further, a starting point for a search related to the input target vector in the codebook is determined, based on the result of the comparison. The codevectors in the codebook are sorted according to a distortion measure reflecting the distance between each codevector and the centroids of the classes. The Vector Quantizer and method enables that the class of codevectors comprising the most probable candidate codevectors in regard of the input vector. s may be searched first.

Abstract: A method and device for performing audio recognition, including: collecting a first audio document to be recognized; initiating calculation of first characteristic information of the first audio document, including: conducting time-frequency analysis for the first audio document to generate a first preset number of phase channels; and extracting at least one peak value characteristic point from each phase channel of the first preset number of phrase channels, where the at least one peak value characteristic point of each phase channel constitutes the peak value characteristic point sequence of said each phase channel; and obtaining a recognition result for the first audio document, wherein the recognition result is identified based on the first characteristic information, and wherein the first characteristic information is calculated based on the respective peak value characteristic point sequences of the preset number of phase channels.

Abstract: A selected channel of a multi-channel signal which is represented by frames composed from sampling values having a high time resolution can be encoded with higher quality when a wave form parameter representation representing a wave form of an intermediate resolution representation of the selected channel is derived, the wave form parameter representation including a sequence of intermediate wave form parameters having a time resolution lower than the high time resolution of the sampling values and higher than a time resolution defined by a frame repetition rate. The wave form parameter representation with the intermediate resolution can be used to shape a reconstructed channel to retrieve a channel having a signal envelope close to that one of the selected original channel. The time scale on which the shaping is performed is shorter than the time scale of a framewise processing, thus enhancing the quality of the reconstructed channel.

Abstract: Aspects of the present disclosure include systems and methods for injecting content into a webpage at or local to a network access gateway. For example, in an embodiment, a network access gateway is provided for accessing the internet. A user logs onto the internet through the network access gateway and request a webpage. The gateway requests the webpage from the webpage provider. Before the gateway delivers the webpage to the user, the gateway, or a content injection engine local to the gateway, injects content, such as, for example, advertisements or other useful information, into the webpage and then sends the altered webpage to the user.

Abstract: A speech analysis system uses one or more digital processors to reconstruct a speech signal by accurately extracting speech formants from a digitized version of the speech signal. The system extracts the formants by determining an estimated instantaneous frequency and an estimated instantaneous bandwidth of speech resonances of the digital version of the speech signal in real time. The system digitally filters the digital speech signal using a plurality of complex digital filters in parallel having overlapping bandwidths to ensure that substantially all of the bandwidth of the speech signal is covered. This virtual chain of overlapping complex digital filters produces a corresponding plurality of complex filtered signals. A first estimated frequency and a first estimated bandwidth is generated for each of the filtered signals, and speech resonances of the input speech signal are identified therefrom.

Abstract: A speech signal processing apparatus includes an amplitude and phase signal generation section that, based on an analyzing signal expressed by a complex signal generated from a speech signal applied with pitch marks every 1 pitch cycle, generates an amplitude signal and a phase signal on the time axis of the speech signal, a phase signal conversion section that converts the phase signal into a phase signal of a target pitch cycle width for each section of the 1 pitch cycle width based on the pitch marks, and a pitch conversion speech signal generation section that generates a speech signal in which pitch cycle is converted to the target pitch cycle based on an amplitude signal of the target pitch cycle width of a section corresponding to the section of the amplitude signal and based on a phase signal of the target pitch cycle width.

Abstract: An approach to forming output signals both permits flexible and temporally and/or frequency local processing of input signals while limiting or mitigating artifacts in such output signals. Generally, the approach involves first synthesizing prototype signals for the output signals, or equivalently characterizing such prototypes, for example, according to their statistical characteristics, and then forming the output signals as estimates of the prototype signals, for example, as weighted combinations of the input signals.

Abstract: The invention provides a method for estimating a fundamental frequency of a speech signal comprising the steps of receiving a signal spectrum of the speech signal, filtering the signal spectrum to obtain a refined signal spectrum, determining a cross-power spectral density using the refined signal spectrum and the signal spectrum, transforming the cross-power spectral density into the time domain to obtain a cross-correlation function, and estimating the fundamental frequency of the speech signal based on the cross-correlation function.

Abstract: An electronic device for coding a transient frame is described. The electronic device includes a processor and executable instructions stored in memory that is in electronic communication with the processor. The electronic device obtains a current transient frame. The electronic device also obtains a residual signal based on the current transient frame. Additionally, the electronic device determines a set of peak locations based on the residual signal. The electronic device further determines whether to use a first coding mode or a second coding mode for coding the current transient frame based on at least the set of peak locations. The electronic device also synthesizes an excitation based on the first coding mode if the first coding mode is determined. The electronic device also synthesizes an excitation based on the second coding mode if the second coding mode is determined.

Abstract: Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.

Abstract: A non-spatial speech detection system includes a plurality of microphones whose output is supplied to a fixed beamformer. An adaptive beamformer is used for receiving the output of the plurality of microphones and one or more processors are used for processing an output from the fixed beamformer and identifying speech from noise though the use of an algorithm utilizing a covariance matrix.

Abstract: A vector joint encoding/decoding method and a vector joint encoder/decoder are provided, more than two vectors are jointly encoded, and an encoding index of at least one vector is split and then combined between different vectors, so that encoding idle spaces of different vectors can be recombined, thereby facilitating saving of encoding bits, and because an encoding index of a vector is split and then shorter split indexes are recombined, thereby facilitating reduction of requirements for the bit width of operating parts in encoding/decoding calculation.

Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

Abstract: A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.

Abstract: A speech analyzer includes a speech acquiring section, a frequency converting section, an autocorrelation section, and a pitch detection section. The frequency converting section converts the speech signal acquired by the speech acquiring section into a frequency spectrum. The autocorrelation section determines an autocorrelation waveform by shifting the frequency spectrum along the frequency axis. The pitch detection section determines the pitch frequency from the distance between two local crests or troughs of the autocorrelation waveform.

Abstract: A method for overlap-adding signals useful for performing frame loss concealment (FLC) in an audio decoder as well as in other applications. The method uses a dynamic mix of windows to overlap two signals whose normalized cross-correlation may vary from zero to one. If the overlapping signals are decomposed into a correlated component and an uncorrelated component, they are overlap-added separately using the appropriate window, and then added together. If the overlapping signals are not decomposed, a weighted mix of windows is used. The mix is determined by a measure estimating the amount of cross-correlation between overlapping signals, or the relative amount of correlated to uncorrelated signals.

Abstract: The present invention discloses a method and a device for pulse encoding, and a method and a device for pulse decoding. The method for pulse encoding includes: calculating an index value of an input pulse; selecting an adjustment threshold value according to the number of pulses, and comparing the index value of the pulse with the adjustment threshold value; if the index value is smaller than the adjustment threshold value, adopting the first number of encoding bits to encode the index value, if the index value is not smaller than the adjustment threshold value, adopting the second number of encoding bits to encode the index value plus an offset value, where the first number is smaller than the second number, the first number and the second number are both positive integers, and the offset value is greater than or equal to the adjustment threshold value.

Abstract: A method for training a ranking application. The method includes ranking the help postings to create an initial ranking using initial parameter values, and storing user interactions with the help postings to obtain stored interactions. Simulations are performed using the stored interactions to generate revised parameter values for the ranking application. Performing the simulations includes calculating relevance values from the stored interactions, creating a test posting, assigning, to the test posting, an initial score and a relevance value randomly selected from the relevance values to generate a test ranking, and simulating user interactions with the test ranking to generate simulated rankings. The simulated rankings are analyzed to obtain revised parameter values. The method further includes ranking, using the revised parameter values, the help postings to generate a revised ranking, and displaying the help postings in the forum according to the revised ranking.

Abstract: Provided are an encoding apparatus and a decoding apparatus of a multi-channel signal. The encoding apparatus of the multi-channel signal may process a phase parameter associated with phase information between a plurality of channels constituting the multi-channel signal, based on a characteristic of the multi-channel signal. The encoding apparatus may generate an encoded bitstream with respect to the multi-channel signal using the processed phase parameter and a mono signal extracted from the multi-channel signal.

Abstract: Speech recognition device uses visual information to narrow down the range of likely adaptation parameters even before a speaker makes an utterance. Images of the speaker and/or the environment are collected using an image capturing device, and then processed to extract biometric features and environmental features. The extracted features and environmental features are then used to estimate adaptation parameters. A voice sample may also be collected to refine the adaptation parameters for more accurate speech recognition.

Abstract: Methods, systems, and computer readable media for fricatives and high frequencies detection are disclosed. According to one method, the method includes receiving a narrowband signal. The method also includes detecting, using one or more autocorrelation coefficients, a high frequency speech component associated with the narrowband signal.