Abstract: Provided are a method and apparatus for encoding and decoding a high frequency signal by using a low frequency signal. The high frequency signal can be encoded by extracting a coefficient by linear predicting a high frequency signal, and encoding the coefficient, generating a signal by using the extracted coefficient and a low frequency signal, and encoding the high frequency signal by calculating a ratio between the high frequency signal and an energy value of the generated signal. Also, the high frequency signal can be decoded by decoding a coefficient, which is extracted by linear predicting a high frequency signal, and a low frequency signal, and generating a signal by using the decoded coefficient and the decoded low frequency signal, and adjusting the generated signal by decoding a ratio between the generated signal and an energy value of the high frequency signal.

Abstract: Disclosed is an apparatus for extending a bandwidth of a sound signal. The apparatus includes a database that stores predetermined training information as a result of at least one of Gaussian mixture model (GMM) training and hidden Markov model (HMM) training; a modified discrete cosine transform (MDCT) transformer that transforms a first band signal through MDCT, a feature extractor that extracts a feature parameter of the first band signal from an MDCT coefficient output from the MDCT transformer; an extender that provides an extended MDCT coefficient for a second band signal based on the MDCT coefficient of the first band signal output from the MDCT transformer, a subband energy estimator that estimates subband energy of the second band signal with reference to information stored in the database based on the feature parameter.

Abstract: A method is provided for detecting a predetermined frequency band in an audio data signal which has previously been coded according to a succession of data blocks, among which at least certain blocks contain respectively at least one set of spectral parameters representing a linear prediction filter. Such a method of detection implements, for a current block among the at least certain blocks and for which at least a plurality of spectral parameters of the set have been previously decoded, acts of: determining, among the plurality of previously decoded spectral parameters, the index of the first spectral parameter closest to a threshold frequency; calculating at least one criterion on the basis of the determined index; and deciding whether the predetermined frequency band is detected in the current block, as a function of the criterion calculated.

Abstract: An apparatus for generating a decorrelated signal including a transient separator, a transient decorrelator, a second decorrelator, a combining unit and a mixer, wherein the transient separator is adapted to separate an input signal into a first signal component and into a second signal component such that the first signal component includes transient signal portions of the input signal and such that the second signal component includes non-transient signal portions of the input signal. The combining unit and the mixer are arranged so that a decorrelated signal from a combination unit is fed into the mixer as an input signal.

Abstract: There is provided a method and device for determining an inter-channel time difference of a multi-channel audio signal having at least two channels. A determination is made, at a number of consecutive time instances, of inter-channel correlation based on a cross-correlation function involving at least two different channels of the multi-channel audio signal. Each value of the inter-channel correlation is associated with a corresponding value of the inter-channel time difference. An adaptive inter-channel correlation threshold is adaptively determined based on adaptive smoothing of the inter-channel correlation in time. A current value of the inter-channel correlation is then evaluated in relation to the adaptive inter-channel correlation threshold to determine whether the corresponding current value of the inter-channel time difference is relevant. Based on the result of this evaluation, an updated value of the inter-channel time difference is determined.

Abstract: An encoding method of an encoder is provided. The encoder generates first MDCT coefficients by transforming an input signal, and generates MDCT indices by quantizing the first MDCT coefficients. The encoder generates second MDCT coefficients by dequantizing the MDCT indices, and calculates MDCT residual coefficients using differences between the first MDCT coefficients and the second MDCT coefficients. The encoder generates a residual index by encoding the MDCT residual coefficients, and generates gain indices corresponding to gains from the first MDCT coefficients and the second MDCT coefficients.

Abstract: A method, apparatus and computer program product are therefore provided according to an example embodiment of the present invention in order to perform categorical analysis and synthesis of a multichannel signal to synthesize binaural signals and extract, separate, and manipulate components within the audio scene of the multichannel signal that were captured through multichannel audio means. In the context of a method, a multichannel signal is received. The method may include computing the spectrum for the multichannel signal, determining tonality of bands within the spectrum, and generating a band structure for the spectrum. The method may also include performing spatial analysis of the bands, performing source filtering using the bands, performing synthesis on the filtered band components, and generating an output signal. A corresponding apparatus and a computer program product are also provided.

Abstract: A server system 500 is provided for receiving video clips having an associated audio/musical track for processing at the server system. The system comprises a first beat tracking module for generating a first beat time sequence from the audio signal using an estimation of the signal's tempo and chroma accent information. A ceiling and floor function is applied to the tempo estimation to provide integer versions which are subsequently applied separately to a further accent signal derived from a lower-frequency sub-band of the audio signal to generate second and third beat time sequences. A selection module then compares each of the beat time sequences with the further accent signal to identify a best match.

Abstract: In general, techniques are described by which to perform spatial masking with respect to spherical harmonic coefficients. As one example, an audio encoding device comprising a processor may perform various aspects of the techniques. The processor may be configured to perform spatial analysis based on the spherical harmonic coefficients describing a three-dimensional sound field to identify a spatial masking threshold. The processor may further be configured to render the multi-channel audio data from the plurality of spherical harmonic coefficients, and compress the multi-channel audio data based on the identified spatial masking threshold to generate a bitstream.

Abstract: A low power sound recognition sensor is configured to receive an analog signal that may contain a signature sound. Sparse sound parameter information is extracted from the analog signal. The extracted sound parameter information is sampled in a periodic manner and a context value is updated to indicate a current environmental condition. The sparse sound parameter information is compared to both the context value and a signature sound parameter database stored locally with the sound recognition sensor to identify sounds or speech contained in the analog signal, such that identification of sound or speech is adaptive to the current environmental condition.

Abstract: A recording method for recording a base video stream and an enhancement video stream. The recording method includes: a first step of generating the base video stream by performing an irreversible conversion on an original image; and a second step of generating the enhancement video stream that includes a shift parameter and picture data. A gradation bit sequence of each pixel constituting the picture data of the enhancement video stream represents a difference between a gradation bit sequence of each pixel constituting picture data of the original image and a gradation bit sequence of each pixel constituting picture data of the base video stream. The shift parameter defines a shift operation that is performed by a playback device when the gradation bit sequence of the base video stream is added to the gradation bit sequence of the enhancement video stream.

Abstract: When a plurality of radio signals having different sub frame lengths are transmitted on the same radio carrier, a radio signal of a short sub frame length is arranged inside a carrier band, and a radio signal of a sub frame length longer than the sub frame length of the short sub frame length signal is arranged outside the carrier band.

Abstract: A user selects an audio sample to be combined with a set of audio samples. The selected sample is automatically combined with the set of samples based on metadata corresponding to the sample and metadata corresponding to the set of samples. The rhythmic content (beat locations) of the sample and/or set of samples is automatically adjusted to increase rhythmic coherence of the sample and the set of samples, and a pitch of the sample and/or set of samples is automatically adjusted to increase harmonic coherence of the sample and the set of samples. The user is thus able to select a sample and a set of samples, and have one or both automatically adjusted so that the combination sounds good together both rhythmically and harmonically. Audio samples can be similarly combined with other audio samples, and sets of audio samples can be similarly combined with other sets of audio samples.

Abstract: A method and device are provided for coding or decoding a digital audio signal by transform using analysis or synthesis weighting windows applied to sample frames. The method includes an irregular sampling of an initial window provided for a transform of given initial size N, to apply a secondary transform of size M different from N.

Abstract: A method and apparatus to encode and decode an audio/speech signal is provided. An inputted audio signal or speech signal may be transformed into at least one of a high frequency resolution signal and a high temporal resolution signal. The signal may be encoded by determining an appropriate resolution, the encoded signal may be decoded, and thus the audio signal, the speech signal, and a mixed signal of the audio signal and the speech signal may be processed.

Abstract: Provided is a signal processing apparatus, including a filter unit that filters an audio signal created by decimating a portion of frequency components by an all-pass filter and outputs a filtering result thereof as improvement components to improve sound quality of the audio signal, and an adder that generates an improved sound in which the sound quality of the audio signal is improved by adding the improvement components to the audio signal.

Abstract: A method of depiction of an object in underwater space is provided. The underwater space is acoustically covered using nonlinear wave packets capable of retaining coherence in shallow waters. The echo from the object is encoded as perturbations on flexible cilia-like transducers that are otherwise undergoing limit cycle oscillations. The perturbed pattern of the cilia-like transducers is recorded as holograms and transmitted. The hologram is decoded and a hologram of the underwater space is created. A small hemispherical drive apparatus actuates the cilia to undergo the limit cycle oscillations. Electromagnets are positioned on a housing of the apparatus. A gimbal with shaft is also attached to the housing. The shaft has a first end within the housing that is proximally separated and remains separated from the electromagnets as the shaft rotates. Generated signals excite electromagnets in sequence to produce an electromagnetic track for the shaft.

Abstract: A method for encoding of an audio signal comprises performing (214) of a transform of the audio signal. An energy offset is selected (216) for each of the first subbands. An energy measure of a first reference band within a low band of an encoding of a synthesis signal is obtained (212). The first high band is encoded (220) by providing quantization indices representing a respective scalar quantization of a spectrum envelope in the first subbands of the first high band relative to the energy measure of the first reference band by use of the selected energy offset. An encoder apparatus comprises means for carrying out the steps of the method. Corresponding decoder methods and apparatuses are also described.

Abstract: Described is an encoder (50) for encoding a parametric spectral representation (f) of auto-regressive coefficients that partially represent an audio signal. The encoder includes a low-frequency encoder (10) configured to quantize elements of a part of the parametric spectral representation that correspond to a low-frequency part of the audio signal. It also includes a high-frequency encoder (12) configured to encode a high-frequency part (fH) of the parametric spectral representation (f) by weighted averaging based on the quantized elements (fL) flipped around a quantized mirroring frequency (fm), which separates the low-frequency part from the high-frequency part, and a frequency grid determined from a frequency grid codebook (24) in a closed-loop search procedure. Described are also a corresponding decoder, corresponding encoding/decoding methods and UEs including such an encoder/decoder.

Abstract: An acoustic communication method and device are provided that filter an audio signal to attenuate a high frequency section of the audio signal; generate a residual signal which corresponds to a difference between the audio signal and the filtered signal; generate a psychoacoustic mask for the audio signal based on a predetermined psychoacoustic model; generate a psychoacoustic spectrum mask by combining the residual signal with the psychoacoustic mask; generate an acoustic communication signal by modulating digital data according to the acoustic signal spectrum mask; and combine the acoustic communication signal with the filtered signal.

Abstract: The present proposes new methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.

Abstract: This invention introduces apparatus and methods to efficiently encode the quantization parameters of split multi-rate lattice vector quantization. In this invention, by doing spectral analysis on the split multi-rate vector quantized spectrum, the spectrum is split to null vectors region and non-null vectors region. For the null vectors region, instead of transmitting series of indication for null vectors, an indication of null vectors region and the quantized value of index of the ending vector in the null vectors region (or the number of the null vectors in the null vectors region) are transmitted. The indication of null vectors region can be designed in many ways, the only requirement is the indication should be distinguishable in the decoder side. The ending index or the number of null vectors can be quantized by an adaptively designed codebook. By applying of the invented method, some bits can be saved from the codebook indications.

Abstract: A system and method for limiting amplitude of a signal, such as limiting the loudness of an audio signal. Embodiments of the present invention allow for the most aggressive limiting by using an advanced psychoacoustic model to intelligently determine the amount of limiting that can be done to the incoming signal before producing distortion that is detectable to the human ear.

Abstract: A method of delivering an audio and/or visual media file including, for example, one or more of full or partial master recordings of songs, musical compositions, ringtones, videos, films, television shows, personal recordings, animation and combinations thereof, over the air wirelessly, from one or more servers to an electronic device with or without an Internet connection, said method comprising transmitting and audio and/or visual media file in compressed format to said electronic device, and wherein the electronic device is effective to receive said audio and/or visual file and playback said audio and/or visual content on demand by a user.

Abstract: A coding device includes: a pitch contour detection unit which detects a pitch contour of an input audio signal; a dynamic time warping unit which determines the number of pitch nodes based on the pitch contour and generates a first time warping parameter including information indicating the determined number of pitch nodes, a pitch change position, and a pitch change ratio; a first encoder which codes the first time warping parameter; a time warping unit which corrects pitch, using the information obtained from the first time warping parameter, to approximate the pitches of the number of pitch nodes to a predetermined reference value; a second encoder which codes the input audio signal at the corrected pitch; and a multiplexer which multiplexes the coded time warping parameter and the coded audio signal to generate a bitstream.

Abstract: An audio fingerprint is extracted from an audio sample, where the fingerprint contains information that is characteristic of the content in the sample. The fingerprint may be generated by computing an energy spectrum for the audio sample, resampling the energy spectrum, transforming the resampled energy spectrum to produce a series of feature vectors, and computing the fingerprint using differential coding of the feature vectors. The generated fingerprint can be compared to a set of reference fingerprints in a database to identify the original audio content.

Abstract: An audio codec supporting both, time-domain and frequency-domain coding modes, having low-delay and an increased coding efficiency in terms of iterate/distortion ratio, is obtained by configuring the audio encoder such that same operates in different operating modes such that if the active operative mode is a first operating mode, a mode dependent set of available frame coding modes is disjoined to a first subset of time-domain coding modes, and overlaps with a second subset of frequency-domain coding modes, whereas if the active operating mode is a second operating mode, the mode dependent set of available frame coding modes overlaps with both subsets, i.e. the subset of time-domain coding modes as well as the subset of frequency-domain coding modes.

Abstract: Methods of, apparatuses for, and non-transitory computer readable media having instructions thereon that when executed cause carrying out methods of determining and modifying the perceived loudness of a frequency domain audio signal where the frequency resolution, and corresponding temporal coverage of the frequency domain information is not constant. The frequency (and thus temporal) resolution of the perceived loudness processing is maintained constant at the longest block size. One method includes a block combiner and a loudness modification interpolator.

Abstract: A method of noise-robust speech classification is disclosed. Classification parameters are input to a speech classifier from external components. Internal classification parameters are generated in the speech classifier from at least one of the input parameters. A Normalized Auto-correlation Coefficient Function threshold is set. A parameter analyzer is selected according to a signal environment. A speech mode classification is determined based on a noise estimate of multiple frames of input speech.

Abstract: Systems and methods for detecting features in spoken speech and processing speech sounds based on the features are provided. One or more features may be identified in a speech sound. The speech sound may be modified to enhance or reduce the degree to which the feature affects the sound ultimately heard by a listener. Systems and methods according to embodiments of the invention may allow for automatic speech recognition devices that enhance detection and recognition of spoken sounds, such as by a user of a hearing aid or other device.

Abstract: A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end.

Abstract: A method for embedding digital information into an audio signal, is provided. The method includes dividing the digital information into low-priority data and high-priority data; dividing the audio signal into first and second signal parts; embedding at least one echo signal into the first signal part; embedding a communication signal modulated with low-priority data, which has a spectrum according to psychoacoustic analysis of the second signal part, into the second signal part; and combining the embedded first and second signal parts.

Abstract: A method for processing an audio signal is disclosed. The method for processing an audio signal includes frequency-transforming an audio signal to generate a frequency-spectrum, deciding a weighting per band corresponding to energy per band using the frequency spectrum, receiving a masking threshold based on a psychoacoustic model, applying the weighting to the masking threshold to generate a modified masking threshold, and quantizing the audio signal using the modified masking threshold.

Abstract: Systems and methods are provided herein relating to audio matching. A compact digest can be generated based on sets of triples, where triples are groupings of three interest points that meet threshold criteria. The compact digest can be used in identifying a potential audio match. A full digest can then be used in verifying the potential match. By using a compact digest to perform audio matching, the audio matching system can be scaled to encompass millions or billions of reference audio samples while still using the full digest to maintain accuracy.

Abstract: An exemplary noise reduction system and method processes a speech signal that is delivered in a noisy channel or with ambient noise. Some exemplary embodiments of the system and method use filters to extract speech information, and focus on a subset of harmonics that are least corrupted by noise. Some exemplary embodiments disregard signal harmonics with low signal-to-noise ratio(s), and disregard amplitude modulations that are inconsistent with speech. An exemplary system and method processes a signal that focuses on a subset of harmonics that are least corrupted by noise, disregards the signal harmonics with low signal-to-noise ratio(s), and disregards amplitude modulations that are inconsistent with speech.

Abstract: An objective quality assessment method for obtaining an improved estimate of a perceptual quality degradation of a processed signal, and an arrangement for executing such a method, is provided, which is executed on a processed signal and an associate reference signal. Both signals are split up into associated frame-pairs after which either all or selected frame-pairs are processed further, by creating a reference residual signal and a processed residual signal for each frame-pair, calculating separate ratios of p-norms on both residual signals, and by calculating and storing a per-frame quality estimate on the basis of the ratios of p-norms for each selected frame-pair. An objective per-signal quality estimate that is proportional to the perceptual quality degradation is then provided by aggregating the calculated per-frame-pair quality estimates.

Abstract: A method of operating an audio processing device to improve a user's perception of an input sound includes defining a critical frequency fcrit between a low frequency range and a high frequency range, receiving an input sound by the audio processing device, and analyzing the input sound in a number of frequency bands below and above the critical frequency. The method also includes defining a cut-off frequency fcut below the critical frequency fcrit, identifying a source frequency band above the cut-off frequency fcut, and extracting an envelope of the source band. Further, the method identifying a corresponding target band below the critical frequency fcrit, extracting a phase of the target band, and combining the envelope of the source band with the phase of the target band.

Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Abstract: A method of communicating with an electronic device. The method includes providing an electronic device having an audible sound receiving and generating sub-system including a microphone, transmitting from a source at least one acoustic signal encoded with information, receiving said at least one acoustic signal by said microphone and determining a spatial position, distance or movement of the microphone relative to the source, responsive to the received at least one signal.

Abstract: The present proposes new methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.

Abstract: Systems and methods for reconstructing decomposed audio signals are presented. In exemplary embodiments, a decomposed audio signal is received. The decomposed audio signal may include a plurality of frequency sub-band signals having successively shifted group delays as a function of frequency from a filter bank. The plurality of frequency sub-band signals may then be grouped into two or more groups. A delay function may be applied to at least one of the two or more groups. Subsequently, the groups may be combined to reconstruct the audio signal, which may be outputted accordingly.

Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Abstract: An encoding apparatus includes a first layer encoder that encodes a signal, a first layer decoder that decodes first layer encoded data, a first layer error transform coefficient calculator that transforms a first layer error signal into a frequency domain and a second layer encoder that encodes the first layer error transform coefficient to acquire second layer encoded data. The second layer encoder includes a band determiner that determines a band to be encoded by the second layer encoder, and a first shape vector encoder that refers the first layer error transform coefficient included in the band to generate a first shape vector and first shape encoded information, a target gain calculator calculates target gain per subband, a gain vector generator generates a gain vector using a plurality of target gains, and a gain vector encoder encodes the gain vector to acquire gain encoded information.

Abstract: An encoding device, a decoding device, and related methods are provided that eliminate the loss of synchronization of the adaptive filters of a terminal at the encoding end and a terminal at the decoding end caused by transmission errors. Deterioration of the sound quality is suppressed when a multiple channel signal is encoded with high efficiency using an adaptive filter. In the terminal at the encoding end, a buffer stores updated filter coefficients. When packet loss detection information indicating whether there is any packet loss in the terminal at the decoding end indicates that there is packet loss, a switch outputs the past filter coefficients of the previous (NX+1) frames from the buffer to an adaptive filter. The adaptive filter uses the past filter coefficients of the previous (NX+1) frames to conduct filtering.

Abstract: Disclosed is an encoding apparatus that can efficiently encode a signal that is a broad or extra-broad band signal or the like, thereby improving the quality of a decoded signal. This encoding apparatus includes a band establishing unit (301) that generate, based on the characteristic of the input signal, band establishment information to be used for dividing the band of the input signal to establish a first band part of lower frequency side and a second band part of higher frequency side; a lower frequency encoding unit (302) for encoding, based on the band establishment information, the input signal of the first band part to generate encoded lower frequency part information; and a higher frequency encoding unit (303) for encoding, based on the band establishment information, the input signal of the second band part to generate encoded higher frequency part information.

Abstract: Methods of, apparatuses for, and computer readable media having instructions thereon that when executed cause carrying out methods of determining and modifying the perceived loudness of a frequency domain audio signal where the frequency resolution, and corresponding temporal coverage of the frequency domain information is not constant. The frequency (and thus temporal) resolution of the perceived loudness processing is maintained constant at the longest block size. One method includes a block combiner and a loudness modification interpolator.

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: Psychoacoustic Bass Enhancement (PBE) is integrated with one or more other audio processing techniques, such as active noise cancellation (ANC), and/or receive voice enhancement (RVE), leveraging each technique to achieve improved audio output. This approach can be advantageous for improving the performance of headset speakers, which often lack adequate low-frequency response to effectively support ANC.

Abstract: The invention discloses a method and an encoder for processing a digital audio stereo signal. A digital audio encoder for coding such audio signal comprises a predictive Temporal Noise Shaping (TNS) filter, a Mid-/Side (M/S) coding unit, a control unit for determining a first prediction gain related to the unmodified L/R signal processed by the TNS filter and for determining a second prediction gain related to the M/S-coded L/R signal processed by the TNS filter, wherein the control unit is adapted to disable TNS-filtering—i.e. to bypass the TNS filter—for a current signal frame, if the first and second prediction gains differ by more than a pre-determined mismatch range.

Abstract: Quality of industrial products is evaluated by evaluating non-stationary operation sound, which is a kind of operation sound, from an aspect of tone, using closely simulated evaluation levels of evaluation of non-stationary sound by used of a human sense of hearing. An operation sound of a conforming product sample is converted into sound waveform data by a sound collecting unit, and the sound waveform data is input into a computer via an A-D converter, and then converted into psychoacoustic parameters. Data of pseudo conforming products is additionally obtained from the psychoacoustic parameters of a plurality of conforming product samples by making use of deviation in the data of the conforming product samples. Threshold data is obtained using thresholds and masking data for evaluation calculated from psychoacoustic parameters of data of the conforming product samples and the pseudo conforming products by a statistical technique.