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.

Abstract: An electronic device configured for adaptively encoding a watermarked signal is described. The electronic device includes modeler circuitry that determines watermark data based on a first signal. The electronic device also includes coder circuitry coupled to the modeler circuitry. The coder circuitry determines a low priority portion of a second signal and embeds the watermark data into the low priority portion of the second signal to produce a watermarked second signal.

Abstract: A method for processing sound data is provided for the reconstruction of multi-channel audio data on the basis at least of data on a reduced number of channels and of spatialization data. A test is carried out to determine whether the spatialization data received are valid. If the test is positive, a spatialization value is predicted according to a per respective model of a plurality of models. A prediction model is chosen on the basis of the spatialization values thus predicted and on the basis of the spatialization data received, to permit, in case of subsequent reception of defective spatialization data, a prediction according to this chosen model of a spatialization value and to use this predicted spatialization value for the reconstruction of the multi-channel audio data.

Abstract: An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

Abstract: A method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals.

Abstract: Automatic measurements are made of audio presence and level in an audio signal by direct processing of an MPEG data stream representing the audio signal, without reconstructing the audio signal. Sub-band data is extracted from the data stream, and the extracted sub-band data is dequantized and denormalized. An audio level for the dequantized and denormalized sub-band data is measured without reconstructing the audio signal. Channel characteristics are used in measuring the audio level of the sub-band data, wherein the channel characteristics are used to weight the measured levels. The measured levels are compared against at least one threshold to determine whether an alarm should be triggered.

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 computing device to determine whether to update using a computer file by generating a file signature for that computer file based on its file header information and comparing the file signature to a collection of file signatures for updates already applied for matches.

Abstract: A representation of an audio signal having a first, a second and a third frame is derived by estimating first warp information for the first and second frames and second warp information for the second and third frames, the warp information describing pitch information of the audio signal. First or second spectral coefficients for first and second frames or second and third frames are derived using first or second warp information and a first or second weighted representation of the first and second frames or second and third frames, the first or second weighted representation derived by applying a first or second window function to the first and second frames or second and third frames, wherein the first or second window function depends on the first or second warp information. The representation of the audio signal is generated including the first and the second spectral coefficients.

Abstract: An apparatus for generating a decorrelated signal having a receiving unit for receiving phase information, a transient separator, a transient decorrelator, a second decorrelator and a combining unit, 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 has transient signal portions of the input signal and such that the second signal component has non-transient signal portions of the input signal. The transient decorrelator is adapted to apply the phase information received by the receiving unit to a transient signal component.

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, and 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. Further on, the loudness of the output signal is scaled towards a fixed loudness level in the perceptual loudness domain. Finally, the loudness of the reference signal is scaled from a loudness level corresponding to the output signal related intensity level towards a loudness level related to the loudness level of the scaled output signal in the perceptual loudness domain.

Abstract: Provided are, among other things, systems, methods and techniques for detecting whether a transient exists within an audio signal. According to one representative embodiment, a segment of a digital audio signal is divided into blocks, and a norm value is calculated for each of a number of the blocks, resulting in a set of norm values for such blocks, each such norm value representing a measure of signal strength within a corresponding block. A maximum norm value is then identified across such blocks, and a test criterion is applied to the norm values. If the test criterion is not satisfied, a first signal indicating that the segment does not include any transient is output, and if the test criterion is satisfied, a second signal indicating that the segment includes a transient is output. According to this embodiment, the test criterion involves a comparison of the maximum norm value to a different second maximum norm value, subject to a specified constraint, within the segment.

Abstract: Methods and devices to enable efficient beat-matched, DJ-style crossfading are provided. For example, such a method may involve determining beat locations of a first audio stream and a second audio stream and crossfading the first audio stream and the second audio stream such that the beat locations of the first audio stream are substantially aligned with the beat locations of the second audio stream. The beat locations of the first audio stream or the second audio stream may be determined based at least in part on an analysis of frequency data unpacked from one or more compressed audio files.

Abstract: A method and an apparatus for encoding and decoding audio signals using adaptive sinusoidal coding are provided. The audio signal encoding method includes the steps of dividing a synthesized audio signal into a plurality of sub-bands, calculating the energy of each sub-band, selecting a predetermined number of sub-bands having a relatively large amount of energy from the sub-bands, and performing sinusoidal coding with regard to the selected sub-bands. Application of sinusoidal coding based on consideration of the amount of energy of each sub-band of the synthesized signal improves the quality of the synthesized signal more efficiently.

Abstract: Apparatus for converting an audio signal into a parameterized representation, has a signal analyzer for analyzing a portion of the audio signal to obtain an analysis result; a band pass estimator for estimating information of a plurality of band pass filters based on the analysis result, wherein the information on the plurality of band pass filters has information on a filter shape for the portion of the audio signal, wherein the band width of a band pass filter is different over an audio spectrum and depends on the center frequency of the band pass filter; a modulation estimator for estimating an amplitude modulation or a frequency modulation or a phase modulation for each band of the plurality of band pass filters for the portion of the audio signal using the information on the plurality of band pass filters; and an output interface for transmitting, storing or modifying information on the amplitude modulation, information on the frequency modulation or phase modulation or the information on the plurality o

Abstract: A method, device, system, and computer program product calculate a gradient index as a sum of magnitudes of gradients of speech signals from a received frame at each change of direction; and provide an indication that the frame contains babble noise if the gradient index, energy information, and background noise level exceed pre-determined thresholds or a voice activity detector algorithm and sound level indicate babble noise.

Abstract: The invention proposes extracting one or more speech signals (151-154) as well as one or more ambient signals (131) from sound signals captured by microphones, wherein each of the speech signals corresponds to a different speaker. The invention proposes to transmit both the one or more speech signals (151-154) and the one or more ambient signals (131) to a rendering side, as opposed to sending only speech signals. This enables to reproduce the speech and ambient signals in a spatially different way at the rendering side. By reproducing the ambient signals a feeling of “being together” is created. In an embodiment, the invention enables reproducing two or more speech signals spatially from each other and from the ambient signals so that speech intelligibility is increased despite the presence of the ambient signals.

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: Systems and methods are disclosed for encoding audio in a set-top box that is invoked by a user when listening to a broadcast audio signal from a radio, TV, streaming or other audio device. A detection and identification system comprising an audio encoder is integrated in a set-top box, where detection and identification of media is realized. The encoding automatically identifies characteristics of the media (e.g., the source of a particular piece of material) by embedding an inaudible code within the content. This code contains information about the content that can be decoded by a machine, but is not detectable by human hearing. The embedded code may be used to provide programming information to the view or audience measurement date to the provider.

Abstract: Provided is a frequency band extension apparatus and method, an encoding apparatus and method, a decoding apparatus and method, and a program. Band-pass filters obtain a plurality of lowband subband signals from an input signal. A frequency envelope extracting circuit extracts a frequency envelope from the plurality of lowband subband signals obtained by the plurality of band-pass filters. A highband signal generating circuit generates highband signal components on the basis of the frequency envelope obtained by the frequency envelope extracting circuit, and the plurality of subband signals obtained by the band-pass filters. A frequency band extension apparatus extends the frequency band of the input signal on the basis of the highband signal components generated by the highband signal generating circuit.

Abstract: For determining a quantizer step size for quantizing a signal including audio or video information, a first quantizer step size as well as an interference threshold are provided. Then, the actual interference introduced by the first quantizer step size is determined and compared with the interference threshold. Despite the fact that the comparison reveals that the actually introduced interference exceeds the threshold, a second, coarser quantizer step size is nevertheless used, which will then be used for quantization if it turns out that the interference introduced by the coarser, second quantizer step size falls below the threshold or falls below the interference introduced by the first quantizer step size. Thus, the quantization interference is reduced while the quantization is coarsened and, thus, the compression gain is increased.

Abstract: The present invention applies spectral flatness characteristic values to simplify psychoacoustic analysis of a sound signal. If the sound signal comprises a plurality of frames, the present invention calculates the energy of the sound signal in a frequency domain, calculates a plurality of spectral flatness, and decides to use a short-block or a long-block Modified Discrete Cosine Transform accordingly. If the sound signal comprises left and right channel signals, the present invention performs psychoacoustic analysis on the sound signal to count energy of the left and right channel signals in a frequency domain, counts spectral flatness of the left and right channel signals, and decides to use middle/side transform or left and right channel encoding to transform the left and right channel signals accordingly.

Abstract: Provided are systems, methods and techniques for processing frame-based data. A frame of data, an indication that a transient occurs within the frame, and a location of the transient within the frame are obtained. Based on the indication of the transient, a block size is set for the frame, thereby effectively defining a plurality of equal-sized blocks within the frame. In addition, different window functions are selected for different ones of the plurality of equal-sized blocks based on the location of the transient, and the frame of data is processed by applying the selected window functions.

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: The present invention relates to a postfilter and a postfilter control to be associated with a postfilter for improving perceived quality of speech reconstructed at a speech decoder. The postfilter control comprises means for measuring stationarity of a speech signal reconstructed at a decoder, means for determining a coefficient to a postfilter control parameter based on the measured stationarity, and means for transmitting the determined coefficient to a postfilter, such that the postfilter can process the reconstructed speech signal by applying the determined coefficient to the postfilter control parameter to obtain an enhanced speech signal.

Abstract: An apparatus for providing one or more adjusted parameters for a provision of an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information includes a parameter adjuster. The parameter adjuster is configured to receive one or more input parameters and to provide, on the basis thereof, one or more adjusted parameters. The parameter adjuster is configured to provide the one or more adjusted parameters in dependence on the one or more input parameters and the object-related parametric information, such that a distortion of the upmix signal representation caused by the use of non-optimal parameters is reduced at least for input parameters deviating from optimal parameters by more than a predetermined deviation.

Abstract: Provided are audio encoding and decoding apparatuses capable of recovering a high-quality audio signal at a low bit rate. The audio encoding method includes: detecting at least one sinusoidal wave from an input audio signal; calculating components of additional basis vectors based on residual audio signals and the additional basis vectors of the sinusoidal wave; determining transmission of components of the additional basis vectors; and at least one of (a) encoding frequencies and (b) phases and amplitudes of the sinusoidal waves when the transmission of the components of the additional basis vectors are determined, wherein the residual audio signals are obtained by excluding the detected sinusoidal waves from the input audio signal.

Abstract: Provided is an encoding device which can obtain a sound quality preferable for auditory sense even if the number of information bits is small. The encoding device includes a shape quantization unit (111) having: a section search unit (121) which searches for a pulse for each of bands into which a predetermined search section is divided; and a whole search unit (122) which performs search for a pulse over the entire search section. The shape of an input spectrum is quantized by a small number of pulse positions and polarities. A gain quantization unit (112) calculates a gain of the pulse searched by the shape quantization unit (111) and quantizes the gain for each of the bands.

Abstract: Speaker identification techniques are described. In one or more implementations, sample data is received at a computing device of one or more user utterances captured using a microphone. The sample data is processed by the computing device to identify a speaker of the one or more user utterances. The processing involving use of a feature set that includes features obtained using a filterbank having filters that space linearly at higher frequencies and logarithmically at lower frequencies, respectively, features that model the speaker's vocal tract transfer function, and features that indicate a vibration rate of vocal folds of the speaker of the sample data.

Abstract: The invention concerns an encoder for an input audio signal (S(z)) comprising a combination module combining the input audio signal with an intermediate counter-reaction signal forming a modified input signal and a quantification module scalable for the rate (91) of said modified input signal, delivering a binary raster of quantification indexes of a predetermined rate.

Abstract: The invention consists of a method and computing system for recording and analyzing the voice which allows a series of parameters of phonation to be calculated. These transmit relevant information regarding effects caused by organic disorders (which affect the physiology of the larynx) or neurological disorders (which affect the cerebral centers of speech). The classification methods are also considered an essential part of the invention which allow estimations of the existing dysfunction to be obtained and for the allocation of personality. The usefulness of the invention lies in the possibility of applying the dysfunction estimation in primary care service centers for patient screening to specialist care centers, simplifying examination protocols, saving costs and reducing waiting lists. This methodology can also be used for detecting the personality of a speaker by their voice, allowing access to installations or services.

Abstract: A method, device, system, and computer program product expand narrowband speech signals to wideband speech signals. The method includes determining signal type information from a signal, obtaining characteristics for forming an upper band signal using the determined signal type information, determining signal noise information, using the determined signal noise information to modify the obtained characteristics for forming the upper band signal, and forming the upper band signal using the modified characteristics.

Abstract: A device and a method for quantizing, in a super-frame including a sequence of frames, LPC filters calculated during the frames of the sequence. The LPC filter quantizing device and method comprises: an absolute quantizer for first quantizing one of the LPC filters using absolute quantization; and at least one quantizer of the other LPC filters using a quantization mode selected from the group consisting of absolute quantization and differential quantization relative to at least one previously quantized filter amongst the LPC filters. For inverse quantizing, at least the first quantized LPC filter is received and an inverse quantizer inverse quantizes the first quantized LPC filter using absolute inverse quantization. If any quantized LPC filter other than the first quantized LPC filter is received, an inverse quantizer inverse quantizes this quantized LPC filter using one of absolute inverse quantization and differential inverse quantization relative to at least one previously received quantized LPC filter.

Abstract: The signal processing is based on the concept of using a time-domain aliased frame as a basis for time segmentation and spectral analysis, performing segmentation in time based on the time-domain aliased frame and performing spectral analysis based on the resulting time segments. The time resolution of the overall “segmented” time-to-frequency transform can thus be changed by simply adapting the time segmentation to obtain a suitable number of time segments based on which spectral analysis is applied. The overall set of spectral coefficients, obtained for all the segments, provides a selectable time-frequency tiling of the original signal frame.

Abstract: In a first aspect, arbitrary shaping of the temporal envelope of noise is provided in spectral domain coding systems without the need of side-information. In the encoding, a filtered measure of quantization error is applied as a feedback signal to the frequency-domain representation of a discrete time-domain signal prior to quantization, so that the filtering parameters of said filtering affect the shaping of quantization noise in the time domain of the quantized frequency-domain representation with unchanged quantization of the discrete time-domain signal when it is inversely transformed from the frequency domain back to the time domain in decoding. This may be accomplished with respect to each of a plurality of frequency bins or groups of bins. In another aspect, frequency-domain noise-feedback quantizing in digital audio encoding is provided.

Abstract: A method for dynamically adjusting the spectral content of an audio signal, which increases the harmonic content of said audio signal, said method comprising translating an encoded digital signal into data bands, creating a psychoacoustic model to identify sections of said data bands that are deficient in harmonic quality, analyzing the fundamental frequency and amplitude of said harmonically deficient data bands, creating additional higher order harmonics for said harmonically deficient data bands, adding said higher order harmonics back to said encoded digital signal to form a newly enhanced signal, inverse filtering said newly enhanced signal, and converting said inverse filtered signal to an analog waveform for consumption by the listener.

Abstract: One provides (101) a digital audio signal having a corresponding signal bandwidth, and then provides (102) an energy value that corresponds to at least an estimate of out-of-signal bandwidth energy as corresponds to that digital audio signal. One then uses (103) the energy value to simultaneously determine both a spectral envelope shape and a corresponding suitable energy for the spectral envelope shape for out-of-signal bandwidth content as corresponds to the digital audio signal. By one approach, if desired, one then combines (104) (on, for example, a frame by frame basis) the digital audio signal with the out-of-signal bandwidth content to provide a bandwidth extended version of the digital audio signal to be audibly rendered to thereby improve corresponding audio quality of the digital audio signal as so rendered.

Abstract: There is disclosed an encoding device capable of improving similarity between the high frequency band spectrum of the original signal and a new spectrum to be generated while realizing a low bit rate when encoding a wide-band signal spectrum. The encoding device has sub-band amplitude calculation units for calculating the amplitude of the respective sub-bands for the high frequency band spectrum obtained from the wide-band signal. A search unit and a gain codebook select some sub-bands from a plurality of sub-bands and only the gain of the selected sub-bands is subjected to encoding. An interpolation unit expresses the gain of the sub-band not selected, by mutually interpolating the selected gains.

Abstract: A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.

Abstract: The present disclosure relates to a signal analyzer for processing an overlapped input signal frame comprising 2N subsequent input signal values. The signal analyzer comprises: a windower adapted to window the overlapped input signal frame to obtain a windowed signal, wherein the windower is adapted to zero M+N/2 subsequent input signal values of the overlapped input signal frame, wherein M is equal or greater than 1 and smaller than N/2; and a transformer adapted to transform the remaining 3N/2?M subsequent windowed signal values of the windowed signal using N?M sets of transform parameters to obtain a transformed-domain signal comprising N?M transformed-domain signal values.

Abstract: An audio encoder, an audio decoder or an audio processor includes a filter for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal, the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.

Abstract: A method for optimizing multiple psychoacoustic effects in a sound system includes synthesizing a high-frequency restored version of a input signal; adding the high-frequency restored version of the input signal to the input signal to create a second signal; synthesizing a third signal having enhanced spatialization from the second signal; synthesizing a fourth signal having virtual bass from the second signal; and, adding the third and fourth signals, or second, third and fourth signals, together to create an output signal.

Abstract: A method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals.

Abstract: Provided are an apparatus and method for coding and decoding a multi object audio signal with multi channel. The apparatus includes: a multi channel encoding means for down-mixing an audio signal including a plurality of channels, generating a spatial cue for the audio signal including the plurality of channels, and generating first rendering information including the generated spatial cue; and a multi object encoding unit for down-mixing an audio signal including a plurality of objects, which includes the down-mixed signal from the multi channel encoding unit, generating a spatial cue for the audio signal including the plurality of objects, and generating second rendering information including the generated spatial cue, wherein the multichannel encoding unit generates a spatial cue for the audio signal including the plurality of objects regardless of a Coder-DECoder (CODEC) scheme the limits the multi channel encoding unit.

Abstract: An audio decoder (100) comprising: effect means, decoding means, and rendering means. The effect means (500) generate modified down-mix audio signals from received down-mix audio signals. Said received down-mix audio signals comprise a down-mix of a plurality of audio objects. Said modified down-mix audio signals are obtained by applying effects to estimated audio signals corresponding to audio objects comprised in said received down-mix audio signals. Said estimated audio signals are derived from the received down-mix audio signals based on received parametric data. Said received parametric data comprise a plurality of object parameters for each of the plurality of audio objects. Said modified down-mix audio signals based on a type of the applied effect are decoded by decoding means or rendered by rendering means or combined with the output of rendering means.

Abstract: A communication device includes memory, an input interface, a processing module, and a transmitter. The processing module receives a digital signal from the input interface, wherein the digital signal includes a desired digital signal component and an undesired digital signal component. The processing module identifies one of a plurality of codebooks based on the undesired digital signal component. The processing module then identifies a codebook entry from the one of the plurality of codebooks based on the desired digital signal component to produce a selected codebook entry. The processing module then generates a coded signal based on the selected codebook entry, wherein the coded signal includes a substantially unattenuated representation of the desired digital signal component and an attenuated representation of the undesired digital signal component. The transmitter converts the coded signal into an outbound signal in accordance with a signaling protocol and transmits it.

Abstract: Provided are, among other things, systems, methods and techniques for detecting whether a transient exists within an audio signal. According to one representative embodiment, a segment of a digital audio signal is divided into blocks, and a norm value is calculated for each of a number of the blocks, resulting in a set of norm values for such blocks, each such norm value representing a measure of signal strength within a corresponding block. A maximum norm value is then identified across such blocks, and a test criterion is applied to the norm values. If the test criterion is not satisfied, a first signal indicating that the segment does not include any transient is output, and if the test criterion is satisfied, a second signal indicating that the segment includes a transient is output. According to this embodiment, the test criterion involves a comparison of the maximum norm value to a different second maximum norm value, subject to a specified constraint, within the segment.

Abstract: The invention relates to a method of identifying and correcting errors in a noisy binary mask. An object of the present invention is to provide a scheme for improving a binary mask representing speech. The problem is solved in that the method comprises a) providing a noisy binary mask comprising a binary representation of the power density of an acoustic signal comprising a target signal and a noise signal at a predefined number of discrete frequencies and a number of discrete time instances; b) providing a statistical model of a clean binary mask representing the power density of the target signal; and c) using the statistical model to detect and correct errors in the noisy binary mask. This has the advantage of providing an alternative and relatively simple way of improving an estimate of a binary mask representing a speech signal. The invention may e.g. be used for speech processing, e.g. in a hearing instrument.

Abstract: An encoder for compressing a plurality of independent mono audio channels into a recording and generating a restricted set of additional parameters used to master an audio track of a storage device is described. The plurality of independent mono audio channels are constructed such that the storage device can be played using solid state disk player so that in a first mode all of the plurality of independent mono audio channels are played as the recording and in a second mode the original channels are reconstructed using a higher sample rate. A corresponding decoder and an audio system comprising such encoder and decoder are also described.