Abstract: An electronic device according to the present invention comprises a pair of housings coupled to each other openably and closably by a hinge mechanism, and each of the housings comprises a front cabinet and a back cabinet joined to each other. The back cabinet of at least one of the housings is formed from a material having a greater expansion coefficient than that of a material forming the front cabinet, and the back cabinet of at least one of the housings has a cutting part extending in the width direction perpendicular to the longitudinal direction thereof and longitudinally cutting at least a part of the back cabinet.

Abstract: Methods, devices, and computer program products enable the embedding of forensic marks in a host content that is in compressed domain. These and other features are achieved by preprocessing of a host content to provide a plurality of host content versions with different embedded watermarks that are subsequently compressed. A host content may then be efficiently marked with forensic marks in response to a request for such content. The marking process is conducted in compressed domain, thus reducing the computational burden of decompressing and re-compressing the content, and avoiding further perceptual degradation of the host content. In addition, methods, devices and computer program products are disclosed that obstruct differential analysis of such forensically marked content.

Abstract: An efficient encoded representation of a first and a second input audio signal can be derived using correlation information indicating a correlation between the first and the second input audio signals, when a signal characterization information, indicating at least a first or a second, different characteristic of the input audio signal is additionally considered. Phase information indicating a phase relation between the first and the second input audio signals is derived, when the input audio signals have the first characteristic. The phase information and a correlation measure are included into the encoded representation when the input audio signals have the first characteristic, and only the correlation information is included into the encoded representation when the input audio signals have the second characteristic.

Abstract: A new approach to speech recognition that reacts to concepts conveyed through speech, which shifts the balance of power in speech recognition from straight sound recognition and statistical models to a more powerful and complete approach determining and addressing conveyed concepts. A probabilistically unbiased multi-phoneme recognition process is employed, followed by a phoneme stream analysis process that builds the list of candidate words derived from recognized phonemes, followed by a permutation analysis process that produces sequences of candidate words with high potential of being syntactically valid, and finally, by processing targeted syntactic sequences in a conceptual analysis process to generate the utterance's conceptual representation that can be used to produce an adequate response.

Abstract: A multi-channel audio decoder reconstructs multi-channel audio of more than two physical channels from a reduced set of coded channels based on correlation parameters that specify a full power cross-correlation matrix of the physical channels, or merely preserve a partial correlation matrix (such as power of the physical channels, and some subset of cross-correlations between the physical channels, or cross-correlations of the physical channels with coded or virtual channels).

Abstract: A sound signal correcting apparatus converts an acquired sound signal into a phase spectrum and an amplitude spectrum by an FFT process, compares the amplitude spectrum of the obtained sound signal with a noise model so that a correction coefficient used for correcting the amplitude spectrum of the sound signal is derived, smoothes waveform of the amplitude spectrum of the sound signal using the derived correction coefficient, and converts the sound signal into a sound signal where the amplitude spectrum is corrected by performing an inverse FFT process on the phase spectrum and the smoothed amplitude spectrum.

Abstract: A decoding apparatus that decodes a first encoded data that is encoded into a first time range from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of the audio signal from the low-frequency component and encoded into a second time range, into the audio signal. In the decoding apparatus, a high-frequency component compensating unit that compensates the high-frequency component created from the second encoded data based on the first time range. A decoding unit that decodes into the audio signal by synthesizing the high-frequency component compensated by the high-frequency component compensating unit, and the low-frequency component decoded from the first encoded data.

Abstract: Disclosed is an adaptive sound source vector quantization device capable of reducing deviation of the quantization accuracy of the adaptive sound source vector quantization of each sub-frame when performing an adaptive sound source vector quantization in a sub-frame unit by using a greater information amount in a first sub-frame than in a second sub-frame.

Abstract: A Unified Speech and Audio Codec (USAC) for adjusting an overlap area of a window based on a transition is provided. To increase an encoding efficiency, encoding may be performed by overlapping relatively long windows. Additionally, when a transition is generated between frames, an overlap area of a window may be reduced based on the transition, thereby preventing a noise from occurring due to the transition.

Abstract: Embodiments of the invention provide a method and encoder for encoding a frame in of a communication system. The method includes calculating a first set of parameters associated with the frame, wherein said first set of parameters comprises filter bank parameters. The method further includes selecting, in a first stage, one of a plurality of encoding methods based on the first set of parameters one of modes for encoding, calculating a second set of parameters associated with the frame, selecting, in a second stage, one of the plurality of encoding methods based on the result of the first stage selection and the second set of parameters one of modes for encoding, and encoding the frame using the selected encoding excitation method from the second stage.

Abstract: A signature is extracted from the audio of a program received by a tunable receiver such that the signature characterizes the program. In order to extract the signature, blocks of the audio are converted to corresponding spectral moments. At least one of the spectral moments is then converted to the signature. Also, a test audio signal from a receiver is correlated to a reference audio signal by converting the test audio signal and the reference audio signal to corresponding test and reference spectra, determining test slopes corresponding to coefficients of the test spectrum and reference slopes corresponding to coefficients of the reference spectrum, and comparing the test slopes to the reference slopes in order to determine a match between the test audio signal and the reference audio signal.

Abstract: Video signal relative to an imaged audience and audio signal according to voices from the audience are generated in an input unit. A characteristic amount detection unit detects information on a movement amount, movement periodicity, a volume, voice periodicity of the audience, and a frequency component of voices from the audience based on the video signal or the audio signal. An estimation unit estimates an audience state based on the detected result. An output unit outputs the estimated result of the audience state. The audience state can be easily estimated without observing the audience state by a person.

Abstract: A method and system for audibly outputting text messages includes: setting a vocalizing function for audibly outputting text messages, searching a character speech library for each character of a received text message, and acquiring pronunciation data of each character of the received text message. The method and the system further includes vocalizing the pronunciation data of each character of the received text message, generating a voice message, and audibly outputting the generated voice message.

Abstract: A method and associated apparatus for using visual separators to indicate additional character combination choices from a disambiguation function on a handheld electronic device.

Abstract: A method of communicating speech comprising time-warping a residual low band speech signal to an expanded or compressed version of the residual low band speech signal, time-warping a high band speech signal to an expanded or compressed version of the high band speech signal, and merging the time-warped low band and high band speech signals to give an entire time-warped speech signal. In the low band, the residual low band speech signal is synthesized after time-warping of the residual low band signal while in the high band, an unwarped high band signal is synthesized before time-warping of the high band speech signal. The method may further comprise classifying speech segments and encoding the speech segments. The encoding of the speech segments may be one of code-excited linear prediction, noise-excited linear prediction or ? frame (silence) coding.

Abstract: The invention concerns a method for spectral enhancement and a device therefor. The inventive method is a method for enhancing spectral content of a signal having an incomplete spectrum including a first spectral band, the method including the following steps: at least transposing the spectral content of the first band into a second spectral band not included in the spectrum to generate a transposed spectrum signal, with spectrum limited to the second spectral band; transforming the spectrum of the transposed spectrum signal to obtain an enhancing signal; combining the incomplete spectrum signal and the enhancing signal to produce a spectrum enhanced signal. The invention is characterized in that the spectral content is subjected to a whitening step.

Abstract: A distributed call recording method and system. A recording agent performs recording of a phone call, preparing of the recorded phone call for being subsequently archived, and transferring of the prepared phone call to a central archiving server through a secure connection. The recording agent is a software application runs on a computer system, which may be a generic personal computer or a phone for Voice/video Over Internet Protocol (VOIP) communication. The secure connection utilizes a connection-oriented transport protocol and a security application, providing Quality of Service and data integrity, respectively. A remote management server may be used for monitoring, configuring, and controlling operations of a recording agent through a management interface.

Abstract: A method for processing audio data includes determining a first common scalefactor value for representing quantized audio data in a frame. A second common scalefactor value is determined for representing the quantized audio data in the frame. A line equation common scalefactor value is determined from the first and second common scalefactor values.

Abstract: According to one embodiment, an improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal.

Abstract: A maintenance system of an image forming apparatus is composed of an image forming apparatus and a central management apparatus connected to it via a communication line. The image forming apparatus is composed of an image forming unit for forming an image, an information output unit for outputting intra-machine information of the image forming unit, a voice information input unit for inputting voice information of a user of the image forming unit, and a first communication controller for outputting the intra-machine information and voice information via the communication line.

Abstract: A method is provided for coding a source audio signal. The method includes the following steps: coding a quantization profile of coefficients representative of at least one transform of the source audio signal, according to at least to distinct coding techniques, delivering at least two sets of data representative of a quantization profile; selecting one of the sets of data representative of a quantization profile, as a function of a predetermined selection criterion; transmitting and/or storing the set of data representative of a selected quantization profile and an indicator representative of the corresponding coding technique.

Abstract: Stereo to mono voice conferencing conversion is performed during a voice conference. Conferencing equipment receives audio for right and left channels and filters each of the channels into a plurality of bands. For each band of each channel, the equipment determines an energy level and compares each energy level for each band of the right channel to each energy level for each corresponding band of the left channel. Based on the comparison, the equipment determines which channel has more audio resulting from speech. Based on the determination, the equipment adjusts delivery of the audio from the right and left channels to a mono channel for transmission to endpoints only capable of mono audio in the voice conference.

Abstract: A sensation of presence of voice chat in a virtual space is enhanced. A user speech synthesizer used in a virtual space sharing system where information processing devices share the virtual space. The user speech synthesizer comprises a speech data acquiring section (60) for acquiring speech data representing a speech uttered by the user of one of the information processing devices, an environment sound storage section (66) for storing an environment sound associated with one or more regions defined in the virtual space, a region specifying section (64) for specifying a region corresponding to the user in the virtual space, and an environment sound synthesizing section (68) for acquiring the environment sound associated with the specified region from the environment sound storage section (66), combining the acquired environment sound and the speech data and synthesizing synthesized speech data.

Abstract: A device for suppressing ambient sounds from speech received by a microphone array is provided. One embodiment of the device comprises a microphone array, a processor, an analog-to-digital converter, and memory comprising instructions stored therein that are executable by the processor.

Abstract: Frames containing audio data may be received, the audio data having been derived from a microphone array, at least some of the frames containing residual acoustic echo after having acoustic echo partially removed therefrom. Probability distribution functions are determined from the frames of audio data. A probability distribution function comprises likelihoods that respective directions are directions of sources of sounds. An active speaker may be identified in frames of video data based on the video data and based on audio information derived from the audio data, where use of the audio information as a basis for identifying the active speaker is controlled by determining whether the probability distribution functions indicate that corresponding audio data includes residual acoustic echo.

Abstract: An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.

Abstract: An encoder/decoder for multi-channel audio data, and in particular for audio reproduction through wave field synthesis. The encoder comprises a two-dimensional filter-bank to the multi-channel signal, in which the channel index is treated as an independent variable as well as time, and and the resulting spectral coefficient are quantized according to a two-dimensional psychoacoustic model, including masking effect in the spatial frequency as well as in the temporal frequency. The coded spectral data are organized in a bitstream together with side information containing scale factors and Huffman codebook identifiers.

Abstract: Systems and methods that, among other things, analyze and monitor the performance of a call center including performance of the interactive voice response (IVR) systems, call center agents, and other components of the call center. The systems and methods record characteristics of the call such as the audio data, and analyze that record to identify the events and the actions that take place during the call. A call center administrator may also identify a set of metrics, such as the number of dropped calls that occur during a day that may be monitored by the systems described herein. The data collected about these events and the resulting metrics may be stored in a database and provided to a call center administrator through a user interface that allows the administrator to browse through the collected metric and recorded call data and directly review relevant portions of a call.

Abstract: In a voice processing device, a male voice index calculator calculates a male voice index indicating a similarity of the input sound relative to a male speaker sound model. A female voice index calculator calculates a female voice index indicating a similarity of the input sound relative to a female speaker sound model. A first discriminator discriminates the input sound between a non-human-voice sound and a human voice sound which may be either of the male voice sound or the female voice sound. A second discriminator discriminates the input sound between the male voice sound and the female voice sound based on the male voice index and the female voice index in case that the first discriminator discriminates the human voice sound.

Abstract: An arrangement is disclosed for offset compensation of a time-interleaved analog-to-digital converter, having a plurality of computing channels and being adapted to convert a signal from an analog domain to a digital domain. The arrangement comprises the time-interleaved analog-to-digital converter, an analog offset estimation and compensation unit adapted to estimate a mean offset for the plurality of computing channels, a digital offset estimation and compensation unit adapted to estimate a residual computing channel specific offset for each of the plurality of computing channels, and offset compensation means. The offset compensation means are adapted to perform offset compensation in the analog domain of each of the plurality of channels based on the estimated mean offset in the analog domain, and to perform offset compensation in the digital domain of each of the plurality of channels based on respective residual computing channel specific offset.

Abstract: A method, computer program product, and system for variant string matching. A computer implemented method for variant string matching may comprise comparing with a computing device two unidentical strings in a training variant string pair. The two unidentical strings may represent the same item from training data, which may be stored in a memory. The two unidentical strings may be compared to determine if they include an identical substring pair, and a first unidentical substring pair. The computer implemented method may also determine if the first unidentical substring pair includes a first unidentical substring and a second unidentical substring. The computer implemented method may further determine if the first unidentical substring pair is in the training data. The first unidentical substring pair may be entered into the training data as a first variant string pair if it is not in the training data.

Abstract: A down-sampler 101 down-samples the sampling rate of an input signal from sampling rate FH to sampling rate FL. A base layer coder 102 encodes the sampling rate FL acoustic signal. A local decoder 103 decodes coding information output from base layer coder 102. An up-sampler 104 raises the sampling rate of the decoded signal to FH. A subtracter 106 subtracts the decoded signal from the sampling rate FH acoustic signal. An enhancement layer coder 107 encodes the signal output from subtracter 106 using a decoding result parameter output from local decoder 103.

Abstract: An apparatus and method for selecting a set of channels from a plurality channels in a signal processor, the method comprising sampling each one of a plurality of channels and obtaining a binary representation of each one of the samples, arranging each one of the binary representations of samples into a series of bit planes from a most significant bit plane containing the most significant bit of each binary representation, to a least significant bit plane containing the least significant bit of each binary representation, determining those bit planes having binary representations that conform to a predetermined value criteria, and selecting a set of channels by summing bits from each one of those determined bit planes that conform to the predetermined value criteria.

Abstract: A memory optimization method for a MP3 decoder. In a pipeline structure for speeding matrix calculation in Mp3 decoding, an output sequence of IMDCT calculation is altered so that matrix calculation is activated before completing the IMDCT calculation. A decoding control method allows pipeline processing in MP3 decoding, with decoding procedures for subsequent granules activated while the current granule is still being processing in the matrix calculation.

Abstract: An active stream is selected from one of a plurality of audio streams generated in a common acoustic environment by obtaining, for each stream obtaining, at a series of measurement instants tn, where n=1 . . . N, a final performance metric that is representative of the stream's goodness for representing near-end speech at each measurement instant tn. The final performance metrics for each stream are accumulated to determine an overall performance score. A switch to a new stream as the active stream occurs when the stream with the best overall performance score exceeds the overall performance score of the currently active stream by a threshold amount.

Abstract: Methods and mobile devices are provided for asymmetric independent processing of audio streams in a system on a chip (SOC). More specifically, independent audio paths are provided for processors performing audio processing on the SOC and mixing of decoded audio samples from the processors is performed digitally on the SOC by a hardware digital mixer.

Abstract: Associating a message identifier with a message is disclosed. The message is specified in a program code. It is determined whether the message specified in the program code matches an entry of a message catalog. A message identifier is associated with the message automatically based at least in part on the determination.

Abstract: Disclosed is an adaptive excitation vector quantization device capable of improving quantization accuracy of adaptive excitation vector quantization while suppressing increase of the calculation amount in CELP encoding which performs encoding in sub-frame units. An adaptive excitation vector generator cuts out an adaptive excitation vector of a frame length (n) from an adaptive excitation codebook. An impulse response matrix former forms a n×n impulse response matrix using impulse response matrixes of sub-frames inputted from a synthesis filter. A target vector generator adds a linear prediction residual vector of each sub-frame to form a target vector of frame length (n). An evaluation measure calculator calculates an evaluation measure of the adaptive excitation vector quantization by using the adaptive excitation vector, the impulse response matrix, and the target vector.

Abstract: A system extends the high-frequency spectrum of a narrowband audio signal in the time domain. The system extends the harmonics of vowels by introducing a non linearity in a narrow band signal. Extended consonants are generated by a random-noise generator. The system differentiates the vowels from the consonants by exploiting predetermined features of a speech signal.

Abstract: The invention relates to a method for determining a quality indicator representing a perceived quality of an output signal of an audio system with respect to a reference signal. The reference signal and the output signal are processed and compared. The processing includes dividing the reference signal and the output signal into mutually corresponding time frames. Additionally, the processing includes scaling the intensity of the reference signal towards a fixed intensity level, and then performing measurements on time frames within the scaled reference signal for determining reference signal time frame characteristics. The intensity of the reference signal is then scaled from the fixed intensity level towards an intensity level related to the output signal. Further on in the method, the loudness of the output signal is scaled towards a fixed loudness level in the perceptual loudness domain. This scaling action uses the reference signal time frame characteristics.

Abstract: A coding apparatus reduces a circuit scale and the amount of coding processing calculation. A frequency domain conversion section performs a frequency analysis of the signal sampled at a sampling rate Fx with an analysis length of 2·Na and calculates first spectrum S1(k)(0?k<Na). A band extension section extends the effective frequency band of first spectrum S1(k) to 0?k<Nb so that a new spectrum can be assigned to the extended area following to the frequency k=Na of first spectrum S1(k). An extended spectrum assignment section assigns extended spectrum S1?(k)(Na?k<Nb) input to the extended frequency band from the outside. A spectral information specification section outputs information necessary to specify extended spectrum S1?(k) out of the spectrum given from the extended spectrum assignment section as a code.

Abstract: A non-intrusive signal quality assessment apparatus includes a feature vector calculator that determines parameters representing frames of a signal and extracts a collection of per-frame feature vectors (?;(n)) representing structural information of the signal from the parameters. A frame selector preferably selects only frames (?\with a feature vector (?;(n)) lying within a predetermined multi-dimensional window (?). Means determine a global feature set (?) over the collection of feature vectors (?;(n)) from statistical moments of selected feature vector components ((1^,02, . . . O11). A quality predictor predicts a signal quality measure (Qj from the global feature set (?)).

Abstract: A unified filter bank for performing signal conversions may include an interface that receives signal conversion commands in relation to multiple types of compressed audio bitstreams. The unified filter bank may also include a reconfigurable transform component that performs a transform as part of signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include complementary modules that perform complementary processing as part of the signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include an interface command controller that controls the configuration of the reconfigurable transform component and the complementary modules.

Abstract: A system for managing voice and data communications coupled to a wide area telecommunications network. Video data is coupled between the system and a video unit under control of a processor. Audio data is coupled between the system and a microphone unit under control of the processor. The audio data are coupled to a TDM bus in the system operating under control of the processor. The video and audio data are coupled to a digital transmission link of the wide area telecommunications network. A user participates in a video call with the video data, while concurrently the user participates in a telephone call with the audio data. Voice communications of the telephone call that stay in a circuit-switched form are provided via the TDM bus. The video call and the telephone call provide a conference call.

Abstract: The present invention relates to an alarm management system intended to be carried onboard an aircraft. More precisely, the invention is aimed at improving the certainty level relating to the integrity of the announcements of faults or information intended for the pilot and based on voice syntheses. For this purpose, the present invention proposes a device and a method for detecting the digital origin of an analog signal providing a validity signal (VAL) enabling the voice announcements made to the pilot to be rendered secure.

Abstract: A device for improving the intelligibility of a signal arising from a source subjected to a noisy environment, said source marking the signal with a specific signature, the device comprising a processing circuit receiving the signal; and means for analyzing the signal and parameterizing the processing circuit according to characteristics of the signature present in the signal. A first channel with low distortion conveys the signal from the source to the means for analyzing, and a second channel, susceptible to introduce a distortion, conveys the signal from the source to the processing circuit.

Abstract: The invention relates to the processing of a signal that is compression encoded (COD) according to a predetermined encoding type applying a quantification operation (Q) and then decoded (DEC) so that the quantification noise is present in the decoded signal (S*). The signal processing of the invention comprises applying a quantification noise reduction (TBQ) to the decoded signal (S), preferably in the following manner: first obtaining information (INF) on the type of compression encoding, selecting a model for the reduction of the quantification noise adapted to said information by estimating the quantification noise (BQ) that the encoding may have generated; and applying to the decoded signal (S*) a processing for reducing the quantification noise (FIL) according to the selected model.

Abstract: A method and system for improving speech quality may include estimating at least one component of a distorted portion of a speech signal from at least one component of an undistorted portion of the speech signal and reinforcing the component of the distorted portion based on the estimating. The components may include the pitch, spectral envelope and spectral energy of the speech signal. The undistorted portion of the speech signal may be delayed and the components of the distorted portion may be interpolated from the components of a delayed undistorted portion and a current undistorted portion of the speech signal. The components of the distorted portion of the speech signal may be extrapolated from a current undistorted portion of the speech signal. Components of the distorted portion of the speech signal may be estimated from frequency bands other than the frequency band affected by the distortion.

Abstract: Apparatus and method for generating high quality synthesized speech having smooth waveform concatenation. The apparatus includes a pitch frequency calculation section, a pitch synchronization position calculation section, a unit waveform storage, a unit waveform selection section, a unit waveform generation section, and a waveform synthesis section. The unit waveform generation section includes a conversion ratio calculation section, a sampling rate conversion section, and a unit waveform re-selection section. The conversion ratio calculation section calculates a sampling rate conversion ratio from the pitch information and the position of pitch synchronization, and the sampling rate conversion section converts the sampling rate of the unit waveform, delivered as input, based on the sampling rate conversion ratio.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with a fixed number of bits or a variable number of bits based on the data structure type.