Abstract: Systems and methods of synchronizing media are provided. A client device may be used to capture a sample of a media stream being rendered by a media rendering source. The client device sends the sample to a position identification module to determine a time offset indicating a position in the media stream corresponding to the sampling time of the sample, and optionally a timescale ratio indicating a speed at which the media stream is being rendered by the media rendering source based on a reference speed of the media stream. The client device calculates a real-time offset using a present time, a timestamp of the media sample, the time offset, and optionally the timescale ratio. The client device then renders a second media stream at a position corresponding to the real-time offset to be in synchrony to the media stream being rendered by the media rendering source.

Abstract: An audio encoder has a window function controller, a windower, a time warper with a final quality check functionality, a time/frequency converter, a TNS stage or a quantizer encoder, the window function controller, the time warper, the TNS stage or an additional noise filling analyzer are controlled by signal analysis results obtained by a time warp analyzer or a signal classifier. Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.

Abstract: A method and system for teaching non-lexical speech effects includes delexicalizing a first speech segment to provide a first prosodic speech signal and data indicative of the first prosodic speech signal is stored in a computer memory. The first speech segment is audibly played to a language student and the student is prompted to recite the speech segment. The speech uttered by the student in response to the prompt, is recorded.

Abstract: Techniques disclosed herein include using a Maximum A Posteriori (MAP) adaptation process that imposes sparseness constraints to generate acoustic parameter adaptation data for specific users based on a relatively small set of training data. The resulting acoustic parameter adaptation data identifies changes for a relatively small fraction of acoustic parameters from a baseline acoustic speech model instead of changes to all acoustic parameters. This results in user-specific acoustic parameter adaptation data that is several orders of magnitude smaller than storage amounts otherwise required for a complete acoustic model. This provides customized acoustic speech models that increase recognition accuracy at a fraction of expected data storage requirements.

Abstract: A band broadening apparatus includes a processor configured to analyze a fundamental frequency based on an input signal bandlimited to a first band, generate a signal that includes a second band different from the first band based on the input signal, control a frequency response of the second band based on the fundamental frequency, reflect the frequency response of the second band on the signal that includes the second band and generate a frequency-response-adjusted signal that includes the second band, and synthesize the input signal and the frequency-response-adjusted signal.

Abstract: An apparatus for encoding includes a first domain converter, a switchable bypass, a second domain converter, a first processor and a second processor to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain converter allow the generation of a decoded audio signal with high quality and low bit rate.

Abstract: In an audio output terminal device, a buffer control unit adjusts the buffer size of a jitter buffer in accordance with the setting of a sound output mode instructed in an instruction receiving unit. If the instruction receiving unit acknowledges an instruction for setting an audio output mode that requires low delay in outputting sound, the buffer control unit reduces the buffer size of the jitter buffer. Further, the buffer control unit controls, in accordance with the instructed setting of the sound output mode, timing for allowing a media buffer to transmit one or more voice packets to the jitter buffer.

Abstract: An audio decoding system including a decoder decoding a first part of audio data, and an audio buffer compressor compressing and storing the decoded first part of audio data in a first time interval and decompressing the stored first part of audio data in a second time interval.

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

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 coding and decoding an audio signal or speech signal and an apparatus adopting the method are provided.

Abstract: According to at least one embodiment, a system for generating a plurality of caption frames is provided. The system comprises a memory storing a plurality of elements generated from transcription information, at least one processor coupled to the memory, and a caption engine component executed by the at least one processor. The caption engine component is configured to identify at least one element sequence as meeting predetermined criteria specifying a plurality of caption characteristics, the at least one element sequence including at least one element of the plurality of elements, and store the at least one element sequence within at least one caption frame. The at least one element sequence may correspond to at least one sentence. The transcription information may be time-coded.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part.

Abstract: Embodiments for audio classification are described. An audio classification system includes at least one device which executes a process of audio classification on an audio signal. The at least one device can operate in at least two modes requiring different resources. The audio classification system also includes a complexity controller which determines a combination and instructs the at least one device to operate according to the combination. For each of the at least one device, the combination specifies one of the modes of the device, and the resources requirement of the combination does not exceed maximum available resources. By controlling the modes, the audio classification system has improved scalability to an execution environment.

Abstract: A purpose is to suppress recognition process delay generated due to load in signal processing. Included is a speech input means 10 that inputs a speech signal, an output evaluation means 20 that evaluates whether or not the speech signal input by the speech input means 10 is the speech signal in a sound section, which is a speech section assuming that a speaker is speaking, and outputs the speech signal as a speech signal to be processed only when evaluated as the speech signal in the sound section, a signal processing means 30 that performs signal processing to the speech signal, which is output by the output evaluation means 20 as the speech signal to be processed, and a speech recognition processing means 40 that performs a speech recognition process to the speech signal which is signal-processed by the signal processing means 30.

Abstract: A method and apparatus to evaluate a quality of an audio signal, in which the number of effective channels is determined for each of a reference signal of a current frame and a test signal indicative of the reference signal that has passed through an audio codec, and an audio quality evaluation score of the current frame is calculated by evaluating an audio quality of the current frame based on the determined number of effective channels for each of the reference signal and the test signal by means of a predetermined evaluator.

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, a device and a system for voice encoding/decoding are disclosed in the present invention. The method includes: assembling an input pulse code modulation signal into one signal according to a designated time slot and assembly manner; and encoding the assembled signal according to a designated encoding manner to output an encoded voice signal. In the present invention, because a process of assembling or splitting the signal may be implemented through software, in the case that hardware in a current network does not need to be replaced, an effect of encoding/decoding voice with a 7 K spectrum may be achieved in the current network.

Abstract: A sound quality correcting apparatus includes: an input module; a feature quantity calculator; a score calculator; a modulation spectrum power calculator; a score corrector; and a signal corrector. The input module receives an input audio signal. The feature quantity calculator calculates feature quantities of the input audio signal for each of a plurality of first intervals having a certain time length. The score calculator calculates a score value for each the first interval based on the feature quantities. The modulation spectrum power calculator calculates a power value, at a certain modulation frequency, of a modulation spectrum of the input audio signal. The score corrector corrects score values in the plurality of first intervals that belong to a second interval if a power value calculated in the second interval is larger than or equal to a certain value. The signal corrector corrects the audio signal based on the score values.

Abstract: A computerized method, software, and system for recognizing emotions from a speech signal, wherein statistical and MFCC features are extracted from the speech signal, the MFCC features are sorted to provide a basis for comparison between the speech signal and reference samples, the statistical and MFCC features are compared between the speech signal and reference samples, a scoring system is used to compare relative correlation to different emotions, a probable emotional state is assigned to the speech signal based on the scoring system and the probable emotional state is communicated to a user.

Abstract: The method comprises, in the frequency domain: estimating (18), for each frequency band of the spectrum (Y(k,l)) of each current time frame (y(k)), a speech presence probability in the signal (p(k,l)); calculating (16) a spectral gain (GOMLSA(k,l)), proper to each frequency band of each current time frame, as a function i) of an estimation of the noise energy in each frequency band, ii) of the speech presence probability estimated at step c1), and iii) of a scalar minimal gain value; and selectively reducing the noise (14) by applying the calculated gain at each frequency band. The scalar minimal gain value, representative of a parameter of soundproofing hardness, is a value (Gmin(k)) that can be dynamically modulated at each successive time frame, calculated for the current time frame as a function of a global variable linked to this current time frame with application of an increment/decrement to a parameterized nominal value (Gmin) of the minimal gain.

Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: This invention relates to a device and a method of generating a first and a second fingerprint (102,104) usable for synchronisation of at least two signals (101,103) and corresponding method and device for synchronising two or more signals. A fingerprint pair is generated on the basis of a segment of a first signal e.g. an audio signal and of a segment of a second signal e.g. a video signal at each synchronisation time point. The generated fingerprint pair(s) are stored in a database (203) and communicated or distributed to a synchronisation device (303). During synchronisation, fingerprint(s) of the audio signal and fingerprint(s) of the video signal to be synchronised are generated and matched against fingerprints in the database. When a match is found, the fingerprints also determine the synchronisation time point, which is used to synchronise the two signals. In this way, a simple, reliable and efficient way of synchronising at least two signals is obtained.

Abstract: The suppression of off-axis audio in an audio environment is provided. Off-axis audio may be considered audio that does not originate from a region of interest. The off-axis audio is suppressed by comparing a phase difference between signals from two microphones to a target slope of the phase difference between signals originating from the region of interest. The target slope can be adapted to allow the region of interest to move with the location of a human speaker such as a driver.

Abstract: An method and apparatus that modifies static media, such as music files being played to a user of the device, upon the generation or receipt of an alert, notification or message, so that information in the alert, notification or message can be incorporated into the media files then communicated to the user. In a further embodiment, a user's response to the communicated information can be sensed using one or more sensors and transducers so as to provide feedback to the device, and then optionally to a node in a system.

Abstract: The present disclosure introduces a new technique for environmental recognition of audio input using feature selection. In one embodiment, audio data may be identified using feature selection. A plurality of audio descriptors may be ranked by calculating a Fisher's discriminant ratio for each audio descriptor. Next, a configurable number of highest ranking audio descriptors based on the Fisher's discriminant ratio of each audio descriptor are selected to obtain a selected feature set. The selected feature set is then applied to audio data. Other embodiments are also described.

Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: A method for slowing a digital audio signal from the transmitter in order to transmit a slowed-down digital audio signal to a receiver, comprises the conversion of the digital audio signal received from a transmitter into a text made up of a series of words, assigning a timestamp bookmark to each word of the text, identifying words that belong to patterns referenced in a database of patterns to be eliminated, the definition of a rate of slowing, the adaptation of timestamped bookmarks to a slowed-down time frame based on the time of slowing, the slowing of the digital audio signal, the deletion of patterns to be eliminated, and the transmission to the receiver of a slowed-down digital audio signal.

Abstract: Aspects of a method and system for an asynchronous pipeline architecture for multiple independent dual/stereo channel PCM processing are provided. Asynchronously pipeline processing of audio information comprised within a decoded PCM frame may be based on metadata information generated from the decoded PCM frame and an output decoding rate. The asynchronously pipeline processing may comprise mixing a primary audio information portion and a secondary audio information, portion, sample rate converting the audio information, and buffering the audio information. The asynchronously pipeline processing may comprise multiple pipeline stages. Feeding back an output of one of the pipeline stages to an input of a previous one of the pipeline stages may be enabled. The metadata information may comprise a frame start indicator associated with the decoded PCM frame and/or a plurality of mixing coefficients.

Abstract: An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.

Abstract: Methods and apparatus to generate and use content-aware watermarks are disclosed herein. In a disclosed example method, media composition data is received and at least one word present in an audio track of the media composition data is selected. The word is then located in a watermark.

Abstract: Topics of potential interest to a user, useful for purposes such as targeted advertising and product recommendations, can be extracted from voice content produced by a user. A computing device can capture voice content, such as when a user speaks into or near the device. One or more sniffer algorithms or processes can attempt to identify trigger words in the voice content, which can indicate a level of interest of the user. For each identified potential trigger word, the device can capture adjacent audio that can be analyzed, on the device or remotely, to attempt to determine one or more keywords associated with that trigger word. The identified keywords can be stored and/or transmitted to an appropriate location accessible to entities such as advertisers or content providers who can use the keywords to attempt to select or customize content that is likely relevant to the user.

Abstract: The present invention is directed to system(s), method(s), and apparatus for accurate fast forward rate when performing trick play with variable distance between frames. In one embodiment, there is presented a circuit for providing a fast forward video sequence. The circuit comprises a system time clock for providing a time reference, said time reference incremented at a predetermined fast forward rate; a comparator for comparing the time reference with timing information associated with a picture; and a controller for determining whether to display the picture based at least in part on the comparison between the timing information and the time reference.

Abstract: A method for encoding an audio signal including: processing a selected subset of a lower series of samples forming a lower frequency spectral band of the audio signal and a higher series of samples forming a higher frequency spectral band of the audio signal to parametrically encode the higher series of samples forming the higher frequency spectral band by identifying a sub-series of the lower series of samples.

Abstract: Noise robust template matching may be performed. First features of a first signal may be computed. Based at least on a portion of the first features, second features of a second signal may be computed. A new signal may be generated based on at least another portion of the first features and on at least a portion of the second features.

Abstract: A Yule-Walker based, low-complexity voice activity detector (VAD) is disclosed. An input signal is typically noisy speech (i.e., corrupted with, for example, babble noise). In one embodiment, a first initialization stage of the VAD computes an occurrence of a silent period within the input signal and the AR parameters. The VAD could accordingly compute a tentative adaptive threshold and output hypothesis H1 (which means speech is present) during this stage. During the second initialization stage, the VAD generally builds a database of associated values and computes the adaptive threshold accordingly. The second initialization stage could also output tentative VAD decisions based on the tentative threshold computed in the first initialization stage. Finally, the VAD periodically retrains or updates AR parameters, threshold values and/or the database and outputs VAD decisions accordingly.

Abstract: A sound encoder for efficiently encoding stereophonic sound. A prediction parameter analyzer determines a delay difference D and an amplitude ratio g of a first-channel sound signal with respect to a second-channel sound signal as channel-to-channel prediction parameters from a first-channel decoded signal and a second-channel sound signal. A prediction parameter quantizer quantizes the prediction parameters, and a signal predictor predicts a second-channel signal using the first decoded signal and the quantization prediction parameters. The prediction parameter quantizer encodes and quantizes the prediction parameters (the delay difference D and the amplitude ratio g) using a relationship (correlation) between the delay difference D and the amplitude ratio g attributed to a spatial characteristic (e.g., distance) from a sound source of the signal to a receiving point.

Abstract: In some embodiments, a method includes measuring a disparity between two speech samples by segmenting both a reference speech sample and a student speech sample into speech units. A duration disparity can be determined for units that are not adjacent to each other in the reference speech sample. A duration disparity can also be determined for the corresponding units in the student speech sample. A difference can then be calculated between the student speech sample duration disparity and the reference speech sample duration disparity.

Abstract: Embodiments of the disclosure can include systems, methods, and apparatus for assigning three-dimensional spatial data to sounds and audio files. In one embodiment, a method can include receiving at least one audio signal, receiving sonic spatial data, associating the sonic spatial data with the at least one audio signal, associating the at least one audio signal and sonic spatial data with a time code, and storing the sonic spatial data, the at least one audio signal, and time code in an encoded sound file.

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 voice or audio signal processor for processing received network packets received over a communication network to provide an output signal, the voice or audio signal processor comprising a jitter buffer being configured to buffer the received network packets, a voice or audio decoder being configured to decode the received network packets as buffered by the jitter buffer to obtain a decoded voice or audio signal, a controllable time scaler being configured to amend a length of the decoded voice or audio signal to obtain a time scaled voice or audio signal as the output voice or audio signal, and an adaptation control means being configured to control an operation of the time scaler in dependency on a processing complexity measure.

Abstract: A response storage unit stores a response, a watching degree relative to a display unit, and an output form of the response to a speaker and the display unit. An extracting unit extracts a request from a speech recognition result. A response determining unit determines a response based on the extracted request. A direction detector detects a viewing direction based on sensing information received from a transmitter mounted on a user. A watching-degree determining unit determines a watching degree based on the viewing direction. An output controller obtains an output form corresponding to the response and the determined watching degree from the response storage unit, and outputs the response to the speaker and the display unit according to the obtained output form.

Abstract: Techniques disclosed herein include using a Maximum A Posteriori (MAP) adaptation process that imposes sparseness constraints to generate acoustic parameter adaptation data for specific users based on a relatively small set of training data. The resulting acoustic parameter adaptation data identifies changes for a relatively small fraction of acoustic parameters from a baseline acoustic speech model instead of changes to all acoustic parameters. This results in user-specific acoustic parameter adaptation data that is several orders of magnitude smaller than storage amounts otherwise required for a complete acoustic model. This provides customized acoustic speech models that increase recognition accuracy at a fraction of expected data storage requirements.

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: Recorded or synthesized speech segments of text-to-speech (TTS) systems may be compressed though the use of both time domain compression and perceptual compression techniques. The twice-compressed recording may be separated into speech segments corresponding to words or subword units for use in a TTS system. The compression rate of time domain compression, and the ratio of time domain compression to perceptual compression, may be modified for any speech segment. The compression amount or ratio may be determined based on linguistic or acoustic features of the word or subword unit that the speech segment represents. Differing compression amounts and ratios may be applied to portions of a single speech segment.

Abstract: The present invention relates to means and methods of automated difference recognition between speech and music signals in voice communication systems, devices, telephones, and methods, and more specifically, to systems, devices, and methods that automate control when either speech or music is detected over communication links. The present invention provides a novel system and method for monitoring the audio signal, analyze selected audio signal components, compare the results of analysis with a pre-determined threshold value, and classify the audio signal either as speech or music.

Abstract: A method and apparatus for speech analysis, comprising detecting an at least one temporal characteristic of an at least one speech of an at least one speaker, and deducing an at least one quantitative score from the at least one temporal characteristic, where the at least one quantitative score indicates an at least one extent of an at least one behavioral aspect of the at least one speaker.

Abstract: There is provided a voice transmitting device, including a band limiting unit that performs band limitation on an input time-series signal, a coding unit that encodes a time-series signal output from the band limiting unit, a transmitting unit that transmits a code string output from the coding unit, and a control unit that controls a band limitation operation in the band limiting unit.

Abstract: A method for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder receives encoded frames of compressed speech information transmitted from an encoder. The method determines whether an encoded frame has been lost, corrupted in transmission, or erased, synthesizes properly received frames, and decides on an overlap-add window to use in combining a portion of the synthesized speech signal with a subsequent speech signal resulting from a received and decoded packet, where the size of the overlap-add window is based on the unavailability of packets. If it is determined that an encoded frame has been lost, corrupted in transmission, or erased, the method performed an overlap-add operation on the portion of the synthesized speech signal and the subsequent speech signal, using the decided-on overlap-add window.

Abstract: The equipment comprises two microphones, sampling means, and de-noising means. The de-noising means are non-frequency noise reduction means comprising a combiner having an adaptive filter performing an iterative search seeking to cancel the noise picked up by one of the microphones on the basis of a noise reference given by the other microphone sensor. The adaptive filter is a fractional delay filter modeling a delay that is shorter than the sampling period. The equipment also has voice activity detector means delivering a signal representative of the presence or the absence of speech from the user of the equipment. The adaptive filter receives this signal as input so as to enable it to act selectively: i) either to perform an adaptive search for the parameters of the filter in the absence of speech; ii) or else to “freeze” those parameters of the filter in the presence of speech.