Abstract: Various embodiments provide techniques for allowing an application to opt out of system default audio stream behavior, as well as techniques for notifying applications on a computing device that a communication audio stream has been initiated. The techniques may differentiate between communication-related audio streams and audio streams that are not communication-related. In some embodiments, an application may register to receive notification that a communication stream has been initiated. The application may be configured to comply with system default audio stream handling policies, or it can perform custom behavior in response to the audio stream notification. In some embodiments, an application may register for filtered or unfiltered notification. In a filtered notification scenario, an application is notified that a communication stream has been initiated when an audio stream associated with the application has not already been modified in response to the initiation of a different communication stream.

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: In a method and device for low-frequency emphasis, where the spectrum of a sound signal is transformed in a frequency domain and comprises transform coefficients grouped in a number of blocks, a maximum energy for one block having a position index is calculated. Also, a factor having a position index smaller than the position index of the block with maximum energy is calculated for each block. For each block, an energy of the block is calculated, the factor is computed from the calculated maximum energy and the computed energy of the block, and a gain is determined from the factor and applied to the transform coefficients of the block.

Abstract: The present invention provides a computationally efficient technique for compression encoding of an audio signal, and further provides a technique to enhance the sound quality of the encoded audio signal. This is accomplished by including more accurate attack detection and a computationally efficient quantization technique. The improved audio coder converts the input audio signal to a digital audio signal. The audio coder then divides the digital audio signal into larger frames having a long-block frame length and partitions each of the frames into multiple short-blocks. The audio coder then computes short-block audio signal characteristics for each of the partitioned short-blocks based on changes in the input audio signal.

Abstract: To alleviate degradation of sound quality which may be caused by pre-echoes and bit starvation. An acoustic analyzer analyzes an audio signal to calculate perceptual entropy indicating how many bits are required for quantization. A coded bit count monitor monitors the number of coded bits produced from the audio signal and calculates the number of available bits for the current frame. Based on the combination of the perceptual entropy and the number of available bits, a frame division number determiner determines a division number N for dividing a frame of the audio signal into N blocks. An orthogonal transform processor divides a frame by the determined division number and subjects each divided block of the audio signal to an orthogonal transform process, thereby obtaining orthogonal transform coefficients. A quantizer quantizes the orthogonal transform coefficients on a divided block basis.

Abstract: The invention includes several techniques and tools, which can be used in combination or separately. For example, an audio encoder can encode information directly using coding processes that include a windowed overlapped transform, a selective multi-channel transform, scalar quantization and entropy encoding. The audio encoder can also encode information parametrically according to a parametric compression mode that accounts for audibility of distortion according to an auditory model. A corresponding audio decoder can decode first information directly and second information according to the parametric mode.

Abstract: The invention relates to an audio encoder and decoder and methods for audio encoding and decoding. In a preferred encoder embodiment an audio signal is encoded by deterministic encoder means to form a first encoded signal part. A spectrum of the audio signal is determined and represented by an excitation pattern, i.e. spectral values corresponding to human auditory filters, as a second encoded signal part. A masking curve is also extracted based on the excitation pattern, thus improving encoding efficiency in terms of bit rate. In a preferred decoder the first encoded signal part is decoded by deterministic decoder means. A noise generator uses the decoded first signal part together with the second signal part, i.e. the excitation pattern for the original audio signal, to generate a noise signal. The noise signal is then added to the first decoded signal part to form an output audio signal. At the decoder side the masking curve is also extracted based on the second encoded signal part, i.e.

Abstract: An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source.

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: 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 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: Provided are, among other things, systems, methods and techniques for encoding an audio signal, in which is obtained a sampled audio signal which has been divided into frames. The location of a transient within one of the frames is identified, and transform data samples are generated by performing multi-resolution filter bank analysis on the frame data, including filtering at different resolutions for different portions of the frame that includes the transient. Quantization data are generated by quantizing the transform data samples using variable numbers of bits based on a psychoacoustical model, and the quantization data are grouped into variable-length segments based on magnitudes of the quantization data. A code book is assigned to each of the variable-length segments, and the quantization data in each of the variable-length segments are encoded using the code book assigned to such variable-length segment.

Abstract: A method and apparatus is disclosed herein for quantizing data using a perceptually relevant search of multiple quantization patterns. In one embodiment, the method comprises performing a perceptually relevant search of multiple quantization patterns in which one of a plurality of prototype patterns and its associated permutation are selected to quantize the target vector, each prototype pattern in the plurality of prototype patterns being capable of directing quantization across the vector; converting the one prototype pattern, the associated permutation and quantization information resulting from both to a plurality of bits by an encoder; and transferring the bits as part of a bit stream.

Abstract: A system and method for adaptive rate control in audio processing is provided. The process could include receiving uncompressed audio data from an input and generating MDCT spectrum for each frame of the uncompressed audio data using a filterbank. The process could also include estimating masking thresholds for current frame to be encoded based on the MDCT spectrum. The masking thresholds reflect a bit budget for the current frame. The process could also include performing quantization of the current frame based on the masking thresholds. After the quantization of the current frame, the bit budget for next frame is updated for estimating the masking thresholds of the next frame. The process could also include encoding the quantized audio data.

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: In an embodiment, a method of frequency domain post-processing is disclosed. The method includes applying adaptive modification gain factor to each frequency coefficient, and determining gain factors based on Local Masking Magnitude and Local Masked Magnitude.

Abstract: The present invention is a detection system of a segment including specific sound signal which detects a segment in a stored sound signal similar to a reference sound signal, including: a reference signal spectrogram division portion which divides a reference signal spectrogram into spectrograms of small-regions; a small-region reference signal spectrogram coding portion which encodes the small-region reference signal spectrogram to a reference signal small-region code; a small-region stored signal spectrogram coding portion which encodes a small-region stored signal spectrogram to a stored signal small-region code; a similar small-region spectrogram detection portion which detects a small-region spectrogram similar to the small-region reference signal spectrograms based on a degree of similarity of a code; and a degree of segment similarity calculation portion which uses a degree of small-region similarity and calculates a degree of similarity between the segment of the stored signal and the reference signal

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

Abstract: An apparatus and method for encoding and decoding a voice signal. The apparatus includes an encoder configured to generate an output bitstream signal from an input voice signal. The output bitstream signal is associated with at least a first standard of a first plurality of CELP voice compression standards. Additionally, the apparatus includes a decoder configured to generate an output voice signal from an input bitstream signal. The input bitstream signal is associated with at least a first standard of a second plurality of CELP voice compression standards. The CELP encoder includes a plurality of codec-specific encoder modules. Additionally, the CELP encoder includes a plurality of generic encoder modules. The CELP decoder includes a plurality of codec-specific decoder modules. Additionally, the CELP decoder includes a plurality of generic decoder modules.

Abstract: The perceptual scalable audio coding/decoding technique lies in the use of a psychoacoustic mask to guide residue coding in enhancement layer coders. At the encoder, a psychoacoustic mask is calculated for the enhancement layer coders or is simply extracted from the coded base layer bitstream. One can also decode the coded base layer bitstream into the audio waveform, and calculate the psychoacoustic mask from the decoded base layer waveform. Furthermore, a predictive technology can be used to refine the psychoacoustic mask derived from the base layer bitstream to form a more accurate psychoacoustic mask of the enhancement layer. In addition, one can calculate the enhancement layer psychoacoustic mask from the original audio, and send the difference between the enhancement layer psychoacoustic mask and the base layer psychoacoustic mask as side information to the decoder. This psychoacoustic mask may then be used for the perceptual coding and decoding of the residue.

Abstract: An apparatus and method of low bit rate encoding and reproducing. The method includes transforming input audio signals in a time domain into spectral signals in a frequency domain, extracting important-spectrum components from the spectral signals in the frequency domain, and quantizing the important-spectrum components, extracting residual-spectrum components other than the important-spectrum components from the spectral signals in the frequency domain, and calculating and quantizing a noise level of the residual-spectrum components, and encoding the quantized important-spectrum components and the quantized noise level losslessly, and outputting encoded bitstreams.

Abstract: An encoding device (1) and method convert a set of signals (l, r) into a dominant signal (m) containing most signal energy, a residual signal (s) containing a remainder of the signal energy, and signal parameters (IID, ICC) associated with the conversion. The dominant signal (m) and selected parts of the residual signal (s) are encoded. Selecting parts of the residual signal involves a residual signal (s?) passing perceptually relevant parts of the residual signal (s), attenuating perceptually less relevant parts of the residual signal and suppressing least relevant parts of the residual signal. An associated decoding device (2) and method decode the encoded dominant signal and the encoded residual signal so as to produce a decoded dominant signal (m?u) and a decoded residual signal (s?mod) respectively. A synthetic residual signal (s?Syn) is derived from the decoded dominant signal (m?u) and is attenuated so as to produce an attenuated synthetic residual signal (S?Syn,mod).

Abstract: The invention provides an efficient technique for encoding a multi-channel audio signal. The invention relies on the principle of encoding (S1) a signal representation of one or more of the multiple channels in a first encoding process, and encoding another signal representation of one or more channels in a second, filter-based encoding process. A basic idea according to the invention is to select (S2), for the second encoding process, a combination of i) frame division configuration of an overall encoding frame into a set of sub-frames, and ii) filter length for each sub-frame, according to a predetermined criterion. The second signal representation is then encoded (S3) in each sub-frame of the overall encoding frame according to the selected combination. The possibility to select frame division configuration and at the same time adjust the filter length for each sub-frame provides added degrees of freedom, and generally results in improved performance.

Abstract: An audio information retrieval method, medium, and system that can rapidly retrieve audio information, even in noisy environments, by extracting a modulation spectrum that is robust against noise, converting features of the extracted modulation spectrum into hash bits, and using a hash table. The audio information retrieval method may include extracting a modulation spectrum from audio data of a compressed domain, converting the extracted modulation spectrum into fingerprint bits, arranging the fingerprint bits in a form of a hash table, converting a received query into an address by a hash function corresponding to the query, and retrieving the audio information by referring to the hash table.

Abstract: There is described a method of encoding input signals (CHI to CH3; 400 to 450) in a multi-channel encoder (5; 15) to generate corresponding output data comprising down-mix output signals (610, 620) together with complementary parametric data (600). The method includes a first step of down-mixing input signals (CHI to CH3; 400 to 450) to generate the corresponding down-mix output signals (610, 620), and a second step of processing the input signals (CHI to CH3; 400 to 450) during down-mixing to generate said parametric data (600) complementary to the down-mix output signals (610, 620). Processing of the input signals (CHI to CH3; 400 to 450) involves including information in the down-mix signals (610, 620) which is useable during subsequent decoding of the down-mix output signals (610, 620) and the parametric data (600) to determine at least some parameter data and thereby enabling representations of the input signals (CHI to CH3; 400 to 450) to be subsequently regenerated.

Abstract: A method and system for processing messages within the framework of an integrated message system. Recipients of messages in an integrated messaging system are provided with an authentic impression of the received message. In a first step, a message received within the framework of an integrated messaging system is automatically translated. Language detection and dictation system is provided. The message contents of the incoming message as well as its segments and parameters are simultaneously utilized to generate additional information regarding the sender and the information, which is suitable to give the recipient an impression of the received message in the most authentic form possible.

Abstract: A method and device for processing signals representing speech or audio via a plurality of filters that approximate behaviors of the basilar membrane of human cochlea. Each of the plurality of filters is formed from a mother filter via the dilation and a shift in time and has the similar impulse response of the basilar membrane to the frequency band for which the filter represents. Any process can be conducted and any feature can be extracted in the domain of the filters' outputs for applications, such as noise reduction, speech synthesis, coding, and speech and speaker recognition.

Abstract: A technique for computing perceptual noise in an audio signal that is computationally efficient. In one example embodiment, the technique includes computing perceptual noise in an input audio signal. The steps involve pre-computing NER (noise-to-excitation ratio) values associated with critical bands within a frame by zeroing out associated spectral coefficient values before the quantization loop, and also assuming bands with lower spectral energy than the band under consideration are zeroed out during quantization. When a critical band is zeroed out during quantization, the associated NER values which have been pre-computed are used in computing an overall perceptual distortion of the frame.

Abstract: A data-compressed audio waveform is temporally modified without requiring complete decompression of the audio signal. Packets of compressed audio data are first unpacked, to remove scaling that was applied in the formation of the packets. The unpacked data is then temporally modified, using one of a number of different approaches. This modification takes place while the audio information remains in a data-compressed format. New packets are then assembled from the modified data, to produce a data-compressed output stream that can be subsequently processed in a conventional manner to reproduce the desired sound. The assembly of the new packets employs a technique for inferring an auditory model from the original packets, to requantize the data in the output packets.

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 either a fixed number of bits or a variable number of bits based on the data structure type as indicated by the data structure type indicator.

Abstract: An audio encoder in which two or more preferably different encoders cooperate to generate a joint encoded audio signal. Encoding parameters of the two or more encoders are optimized in response to a measure of distortion of the joint encoded audio signal in accordance with a predetermined criterion. The distortion. measure is preferably a perceptual distortion measure. In one encoder embodiment comprising a sinusoidal and a waveform encoder, a constant total bit rate for each audio frame is distributed between the two encoders so as to minimize perceptual distortion for both the first and the second encoder. Other embodiments consider a set of encoding parameters that is larger than only those that minimize the perceptual distortion of the first encoder. In some embodiments, perceptual distortion may be minimized by optimizing encoding via optimizing entire encoding templates, i.e. a complex set of encoding parameters, for the separate encoders.

Abstract: The invention relates to methods and units supporting a multichannel audio extension. In order to allow an efficient extension requiring a low computational complexity, it is proposed that at an encoding end, at least state information is provided as side information for a provided mono audio signal (M) generated out of a multichannel audio signal. The state information indicates for each of a plurality of frequency bands how a predetermined or equally provided gain value is to be applied in the frequency domain to the mono audio signal (M) for obtaining first and a second channel signals (L,R) of a reconstructed multichannel audio signal.

Abstract: The invention relates to a method for quantifying components, wherein certain components are each determined based on a plurality of audio signals and can be calculated by the application of a linear conversion on the audio signals, said method comprising: determining a quantification function to be applied to the components by testing a condition relative to an audio signal and depending on a comparison made between a psycho-acoustic masking threshold relative to the audio signal and a value determined based on the reverse linear conversion and quantification errors of the components by the function.

Abstract: In one aspect, the invention divides an audio signal into auditory events, each of which tends to be perceived as separate and distinct, by calculating the spectral content of successive time blocks of the audio signal, calculating the difference in spectral content between successive time blocks of the audio signal, and identifying an auditory event boundary as the boundary between successive time blocks when the difference in the spectral content between such successive time blocks exceeds a threshold. In another aspect, the invention generates a reduced-information representation of an audio signal by dividing an audio signal into auditory events, each of which tends to be perceived as separate and distinct, and formatting and storing information relating to the auditory events. Optionally, the invention may also assign a characteristic to one or more of the auditory events. Auditory events may be determined according to the first aspect of the invention or by another method.

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.

Abstract: A method and apparatus to disturb a voice signal by attenuating and masking the voice signal are provided. The method includes; receiving a voice signal from a wired or wireless network; obtaining a masked voice signal by dividing the received voice signal into a plurality of segments of the same size; outputting the received voice signal and receiving a feedback signal of the output voice signal; obtaining an attenuated voice signal by performing a first sound attenuation operation on the feedback signal; and combining the attenuated voice signal and the masked voice signal and outputting the result of the combination as disturbing sound.

Abstract: A method and an apparatus for encoding/decoding a digital signal are provided. First, a digital input signal is transformed into samples to remove redundant information among signals. Then, a lookup table corresponding to a characteristic of the input signal is selected among a plurality of lookup tables that indicate different numbers of bits allocated for each quantization unit depending on different characteristics of input signals, and the number of bits allocated for each quantization unit is acquired from the selected lookup table. Next, a distribution of samples within each quantization unit is divided into a predetermined number of sections, and the samples are linearly quantized using the allocated number of bits on a section-by-section basis. Thereafter, a bitstream comprised of frames is produced from the quantized samples and predetermined side information so that information about a frame length is stored in the end of frame.

Abstract: An audio encoding apparatus that encodes audio signals of a plurality of channels, includes an adaptive bit allocation control unit that adaptively controls a number of encoding bits assigned to the audio signal of each channel in accordance with perceptual entropy of the audio signal of each of the channels, a fixed bit allocation control unit that fixedly controls the number of encoding bits assigned to the audio signal of each of the channels in predetermined allocations, and a channel encoding unit that encodes the audio signal of each of the channels based on the number of adaptive allocation bits assigned by the adaptive bit allocation control unit and the number of fixed allocation bits assigned by the fixed bit allocation control unit.

Abstract: A method of providing a quality measure for an output voice signal generated to reproduce an input voice signal, the method comprising: partitioning the input and output signals into frames; for each frame of the input signal, determining a disturbance relative to each of a plurality of frames of the output signal; determining a subset of the determined disturbances comprising one disturbance for each input frame such that a sum of the disturbances in the subset set is a minimum; and using the set of disturbances to provide the measure of quality.

Abstract: A bi-phase decoder suitable for use in a broadcast router and an associated method for extracting subframes of digital audio data from a stream of digital audio data. Logical circuitry within the bi-phase decoder extracts subframes of the digital audio data by constructing a transition window from an estimated bit time, sampling the stream of digital audio data using a fast clock and applying the sampled stream of digital audio data to the transition window to identify transitions indicative of preambles of the subframes of digital audio data.

Abstract: A filter bank device for generating a complex spectral representation of a discrete-time signal includes a generator for generating a block-wise real spectral representation, which, for example, implements an MDCT, to obtain temporally successive blocks of real spectral coefficients. The output values of this spectral conversion device are fed to a post-processor for post-processing the block-wise real spectral representation to obtain an approximated complex spectral representation having successive blocks, each block having a set of complex approximated spectral coefficients, wherein a complex approximated spectral coefficient can be represented by a first partial spectral coefficient and by a second partial spectral coefficient, wherein at least one of the first and second partial spectral coefficients is determined by combining at least two real spectral coefficients.

Abstract: The present invention relates to a spectral enhancement method and to an apparatus carrying out this method. The method of the invention enhanced the spectral content of a signal having an incomplete spectrum including a first spectral frequency band, the method comprising the following stages: at least one spectral content transposition of said first frequency band into a second spectral frequency band not included in said spectrum for the purpose of generating a transposed spectrum signal having a spectrum limited to said second spectral frequency band, shaping the spectrum of the transposed spectrum signal for the purpose of producing an enhanced signal, combining an incomplete spectrum signal and the enhanced signal for the purpose of producing an enhanced spectrum signal, characterized in that said spectral content is subject to a stage of whitening.

Abstract: An encoder (100) for encoding a multi-channel audio signal comprises a prediction processor (101) for generating two residual signals for two signal components of the multi-channel signal by linear prediction which is associated with psycho-acoustic characteristics and which specifically uses psycho-acoustic prediction filters; a rotation processor (105) for rotating the combined signal of the two residual signals to generate a main signal and a side signal, in which the energy of the main signal is maximized and the energy of the side signal is minimized; an encoding processor (109) for encoding the main and preferably the side signal; and an output processor (111) for generating an output signal data, prediction parameters and rotation parameters.

Abstract: In accordance with a specific implementation of the disclosure, a stream of audio frames is received and compressed using psycho-acoustical processing. The signal-to-mask ratio table generated by the psycho-acoustical algorithm is updated using only a portion of the received audio frames.

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: The invention for processing speech that is described herein measures the periodic changes of multiple acoustic features in a digitized utterance without regard for lexical, sublexical, or prosodic features. These measurements of periodic, simultaneous changes of multiple acoustic features are assembled into transformational structures. Various types of transformational structures are identified, quantified, and displayed by the invention. The invention is useful for the study of such speaker characteristics as cognitive, emotional, linguistic, and behavioral functioning, and may be employed in the study of other phenomena of interest to the user.

Abstract: A method for reducing a computational complexity of an m-stage adaptive filter is provided by updating recursively forward and backward error prediction square terms for a first portion of a length of the adaptive filter, and keeping the updated forward and backward error prediction square terms constant for a second portion of the length of the adaptive filter.

Abstract: The central idea of the present invention is that the prior procedure, namely interpolation relative to the filter coefficients and the amplification value, for obtaining interpolated values for the intermediate audio values starting from the nodes has to be dismissed. Coding containing less audible artifacts can be obtained by not interpolating the amplification value, but rather taking the power limit derived from the masking threshold, for each node, i.e. for each parameterization to be transferred, and then performing the interpolation between these power limits of neighboring nodes, such as, for example, a linear interpolation. On both the coder and the decoder side, an amplification value can then be calculated from the intermediate power limit determined such that the quantizing noise caused by quantization, which has a constant frequency before post-filtering on the decoder side, is below the power limit or corresponds thereto after post-filtering.

Abstract: A method and system for enhancing copyright revenue generation is disclosed. The method includes receiving a copyrighted media recording, creating an independent work of authorship by generating a simulation of the copyrighted media recording, such that the independent work of authorship is entitled to a copyright and utilizing the simulation in place of the copyrighted media recording, such that use of the independent work of authorship may be entitled to copyright royalties thereon as opposed to requiring copyright royalties for use of the copyrighted media recording.

Abstract: An MPEG-1 layer 3 audio encoder, including a scalefactor generator for determining first scalefactors for encoding a block of audio data if a temporal masking transient is not detected in said block of audio data; and for selecting the maximum of said scalefactors for encoding said block of audio data if a temporal masking transient is detected in said block of audio data to enable greater compression of said audio data. Increases in quantization error, due to use of the maximum scalefactor are pre-masked or post-masked by the temporal masking transient. In cases where the last portion of a block includes a temporal masking transient that masks the preceding portions of the block, the maximum scalefactor is only used to encode the block if the resulting increase in quantization error is less than 30% of the quantization error for the block.