Abstract: An encoder, based on a combination of two audio channels, obtains a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid signal. The first combination signal and the prediction residual signal are encoded and written into a data stream together with the prediction information. A decoder generates decoded first and second channel signals using the prediction residual signal, the first combination signal and the prediction information. A real-to-imaginary transform may be applied for estimating the imaginary part of the spectrum of the first combination signal. For calculating the prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

Abstract: An audio decoder has an arithmetic decoder for providing decoded spectral values on the basis of an arithmetically-encoded representation and a frequency-domain-to-time-domain converter for providing a time-domain audio representation. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a numeric current context value describing a current context state. The arithmetic decoder determines the numeric current context value in dependence on a plurality of previously decoded spectral values. The arithmetic decoder evaluates at least one table using an iterative interval size reduction to determine whether the numeric current context value is identical to a table context value described by an entry of the table or lies within an interval described by entries of the table, and derives a mapping rule index value describing a selected mapping table. An audio encoder also uses an iterative interval table size reduction.

Abstract: A speech conversion system facilitates voice communications. A database comprises a plurality of conversion heuristics, at least some of the conversion heuristics being associated with identification information for at least one first party. At least one speech converter is configured to convert a first speech signal received from the at least one first party into a converted first speech signal different than the first speech signal.

Abstract: An audio decoder includes an arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically encoded representation of the spectral values, and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state described by a numeric current context value. The arithmetic decoder determines the numeric current context value in dependence on a plurality of previously decoded spectral values. The arithmetic decoder evaluates a hash table, entries of which define both significant state values and boundaries of intervals of numeric context values, in order to select the mapping rule. A mapping rule index value is individually associated to a numeric context value being a significant state value.

Abstract: System and method to improve intelligibility of coded speech, the method including: receiving an encoded speech signal from a network; extracting an encoded media data stream and one or more control data packets from the encoded speech signal; decoding the encoded media data stream to produce a decoded speech signal; boosting an upper spectral portion of the decoded speech signal to produce a boosted speech signal; and outputting the boosted speech signal. In another embodiment, the method may include: receiving an uncoded speech signal; processing the uncoded speech signal, wherein the processing comprises generating an unencoded data stream from the uncoded speech signal; boosting an upper spectral portion of the unencoded data stream to produce a boosted speech signal; encoding the boosted speech signal to produce an encoded speech signal; and outputting the boosted speech signal.

Abstract: One or more tags are associated with a voice or multimedia message. These tags can be applied to the message based on one or more of an analysis of the message, rules, caller information, presence information, user input and GPS information. Based on the assigned and associated tags, one or more of message handling, classification and one or more actions can be automatically invoked to assist with management of messages. An interface is also provided that allows for the management of the assigned tags as well as the editing and creation of new tags and rules to assist with message management.

Abstract: Disclosed is a speech synthesizing apparatus including a segment selection unit that selects a segment suited to a target segment environment from candidate segments, includes a prosody change amount calculation unit that calculates prosody change amount of each candidate segment based on prosody information of candidate segments and the target segment environment, a selection criterion calculation unit that calculates a selection criterion based on the prosody change amount, a candidate selection unit that narrows down selection candidates based on the prosody change amount and the selection criterion, and an optimum segment search unit than searches for an optimum segment from among the narrowed-down candidate segments.

Abstract: Improved oscillator-based source modeling methods for estimating model parameters, for evaluating model quality for restoring the input from the model parameters, and for improving performance over known in the art methods are disclosed. An application of these innovations to speech coding is described. The improved oscillator model is derived from the information contained in the current input signal as well as from some form of data history, often the restored versions of the earlier processed data. Operations can be performed in real time, and compression can be achieved at a user-specified level of performance and, in some cases, without information loss. The new model can be combined with methods in the existing art in order to complement the properties of these methods, to improve overall performance. The present invention is effective for very low bit-rate coding/compression and decoding/decompression of digital signals, including digitized speech and audio signals.

Abstract: Filter banks may have different structures and different individual output signal domains. Often a translation between different filter bank domains is desirable. Usually, mapping matrices are used that, however, vary over frequency. This requires a significant amount of lookup tables. A method for transforming first data frames of a first filter bank domain to second data frames of a different second filter bank domain, comprises steps of transcoding sub-bands of the first filter bank domain into sub-bands of an intermediate domain that corresponds to said second filter bank domain but has warped phase, and transcoding the sub-bands of the intermediate domain to sub-bands of the second filter bank domain, wherein a phase correction is performed on the sub-bands of the intermediate domain.

Abstract: The Speech Transmission Index (STI) uses the modulation transfer function to characterize speech communication channels with a single index. This index accurately predicts speech intelligibility. In order to measure the STI, a device or software is needed that utilizes an artificial test signal to estimate the modulation transfer function, and to derive the STI from that function. In practice, many STI measurements are found to be invalid and inaccurate, for two reasons: (1) the measurements are carried out in test conditions for which by design the STI is not intended; (2) equipment is used that does not meet specifications and requirements. A procedure is described to automatically verify if conditions for obtaining valid STI results are met. If possible, corrections are applied to compensate for incorrect settings and poorly adjusted measuring equipment. If automatic correction of the problem is not possible, then a warning is generated.

Abstract: An encoding apparatus comprises a frame processor (105) which receives a multi channel audio signal comprising at least a first audio signal from a first microphone (101) and a second audio signal from a second microphone (103). An ITD processor 107 then determines an inter time difference between the first audio signal and the second audio signal and a set of delays (109, 111) generates a compensated multi channel audio signal from the multi channel audio signal by delaying at least one of the first and second audio signals in response to the inter time difference signal. A combiner (113) then generates a mono signal by combining channels of the compensated multi channel audio signal and a mono signal encoder (115) encodes the mono signal. The inter time difference may specifically be determined by an algorithm based on determining cross correlations between the first and second audio signals.

Abstract: A sound reproducing device has a loudspeaker arranged to produce sound from an audio signal provided by an audio signal source. A microphone is positioned to pick up ambient noise and generate a microphone signal which comprises the noise. An ambient noise cancellation (ANC) system receives the microphone signal from the microphone and generates anti-noise corresponding to the ambient noise in the microphone signal. An automatic polarity adaptation (AAP) system monitors the ANC system and, when a decision criterion is fulfilled, causes a switch in polarity for the generated anti-noise.

Abstract: A decoding device is capable of flexibly calculating high-band spectrum data with a high accuracy in accordance with an encoding band selected by an upper-node layer of the encoding side. In this device: a first layer decoder decodes first layer encoded information to generate a first layer decoded signal; a second layer decoder decodes second layer encoded information to generate a second layer decoded signal; a spectrum decoder performs a band extension process by using the second layer decoded signal and the first layer decoded signal up-sampled in an up-sampler so as to generate an all-band decoded signal; and a switch outputs the first layer decoded signal or the all-band decoded signal according to the control information generated in a controller.

Abstract: A parameter decoding apparatus includes a prediction residue decoder that finds a quantized prediction residue based on encoded information included in a current frame subject to decoding and an auto-regressive predictor produces a predicted parameter by multiplying a predictive coefficient with a past decoded parameter. An adder decodes a parameter by adding the quantized prediction residue and the predicted parameter, wherein the prediction residue decoder, when the current frame is erased, finds a current-frame quantized prediction residue from a weighted linear sum of a parameter decoded in the past and a future-frame quantized prediction residue.

Abstract: A method for searching for an input symbol string, includes receiving (B) an input symbol string, proceeding (C) in a trie data structure to a calculation point indicated by the next symbol, calculating (D) distances at the calculation point, selecting (E) repeatedly the next branch to follow (C) to the next calculation point to repeat the calculation (D). After the calculation (G), selecting the symbol string having the shortest distance to the input symbol string on the basis of the performed calculations. To minimize the number of calculations, not only the distances are calculated (D) at the calculation points, but also the smallest possible length difference corresponding to each distance, and on the basis of each distance and corresponding length difference a reference value is calculated, and the branch is selected (E) in such a manner that next the routine proceeds from the calculation point producing the lowest reference value.

Abstract: Provided are methods and systems of managing handoffs in a wireless communication system having different types of vocoders. Some embodiments include translating state memory of a first vocoder to a second vocoder using a state memory transcoder. The state memory may be delayed to align differences in algorithmic delays between the first vocoder and the second vocoder. In one embodiment, a speech signal may be decoded from the first vocoder, delayed, and encoded to the second vocoder. Furthermore, for a period of time during and/or adjacent to the handoff, the first vocoder may output with decreasing amplitude while the second vocoder outputs with increasing amplitude. Such techniques may be used alone or in combination.

Abstract: An adaptive sound source vector quantization device includes a first pitch cycle instructor, a search range calculator, and a second pitch cycle instructor. The first pitch cycle instructor successively instructs pitch cycle search candidates in a predetermined search range having a search resolution which transits over a predetermined pitch cycle candidate for the first sub-frame. The search range calculator calculates a predetermined range before and after the pitch cycle of the first sub-frame as the pitch cycle search range for the second sub-frame, if the predetermined range includes the predetermined pitch cycle search candidate. In the predetermined range, the search resolution transits over a boundary defined by the predetermined pitch cycle. The second pitch cycle instructor successively instructs the pitch cycle search candidates in the search range for the second sub-frame.

Abstract: An apparatus for synthesizing a rendered output signal having a first audio channel and a second audio channel includes a decorrelator stage for generating a decorrelator signal based on a downmix signal, and a combiner for performing a weighted combination of the downmix signal and a decorrelated signal based on parametric audio object information, downmix information and target rendering information. The combiner solves the problem of optimally combining matrixing with decorrelation for a high quality stereo scene reproduction of a number of individual audio objects using a multichannel downmix.

Abstract: A modulation device including: a modulation unit for modulating a carrier in an audible sound range by an encoded transmission signal to generate a modulated signal; a masker sound generation unit for generating a masker signal outputted as a masker sound for making the modulated signal harder to hear when transmitted with the modulated signal; and an acoustic signal generation unit for inserting the masker signal in the modulated signal to generate an acoustic signal.

Abstract: Systems and methods are disclosed for streaming of audio data of separate streams in at least two different formats. In some embodiments handheld game devices are in wireless communication and a first of the handheld game devices streams audio data during game play to a second of the handheld game devices. In some embodiments the audio data includes audio data from a plurality of streams of audio data. In some embodiments the streams of audio data include streams of audio data in different formats, generally different compressed formats, some of which may be selected based on whether a device includes circuitry specifically configured to decompress audio data in a specific data format.

Abstract: A computerized method and system is provided for automatically selecting from a digitized sound sample a segment of the sample that is optimal for the purpose of measuring clinical metrics for voice and speech assessment. A quality measure based on quality parameters of segments of the sound sample is applied to candidate segments to identify the highest quality segment within the sound sample. The invention can optionally provide feedback to the speaker to help the speaker increase the quality of the sound sample provided. The invention also can optionally perform sound pressure level calibration and noise calibration. The invention may optionally compute clinical metrics on the selected segment and may further include a normative database method or system for storing and analyzing clinical measurements.

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 a 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; a quantization unit for quantizing a transform domain signal; a long term prediction unit for determining an estimation of the frame of the filtered input signal based on a reconstruction of a previous segment of the filtered input signal; and a transform domain signal combination unit for combining, in the transform domain, the long term prediction estimation and the transformed input signal to generate the transform domain signal.

Abstract: A method of transmitting speech data to a remote device in a distributed speech recognition system, includes the steps of: dividing an input speech signal into frames; calculating, for each frame, a voice activity value representative of the presence of speech activity in the frame; grouping the frames into multiframes, each multiframe including a predetermined number of frames; calculating, for each multiframe, a voice activity marker representative of the number of frames in the multiframe representing speech activity; and selectively transmitting, on the basis of the voice activity marker associated with each multiframe, the multiframes to the remote device.

Abstract: A system and method for modeling speech in such a way that both voiced and unvoiced contributions can co-exist at certain frequencies. In various embodiments, three spectral bands (or bands of up to three different types) are used. In one embodiment, the lowest band or group of bands is completely voiced, the middle band or group of bands contains both voiced and unvoiced contributions, and the highest band or group of bands is completely unvoiced. The embodiments of the present invention may be used for speech coding and other speech processing applications.

Abstract: A first party creates and edits a phonetic-alphabet representation of its name. The phonetic representation is conveyed to a second party as “caller-identification” information by messages that set up a call between the parties. The phonetic representation of the name is displayed to the second party, converted to speech, and/or converted to an alphabet of a language of the second party and then displayed to the second party.

Abstract: A parameter decoding device performs a parameter compensation process so as to suppress degradation of a main observation quality in a prediction quantization. The parameter decoding device includes first amplifiers which multiply inputted quantization prediction residual vectors by a weighting coefficient. A further amplifier multiplies the preceding frame decoding LSF vector yn?1 by the weighting coefficient. An additional amplifier multiplies the code vector xn+1 outputted from a codebook by the weighting coefficient ?0. An adder calculates the total of the vectors outputted from the amplifiers, the further amplifier, and the additional amplifier. A selector switch selects the vector outputted from the adder if the frame erasure coding Bn of the current frame indicates that ‘the n-th frame is an erased frame’ and the frame erasure coding Bn+1 of the next frame indicates that ‘the n+1-th frame is a normal frame’.

Abstract: In a method of smoothing background noise in a telecommunication speech session; receiving and decoding S1O a signal representative of a speech session, the signal comprising both a speech component and a background noise component. Subsequently, determining LPC parameters S20 and an excitation signal S30 for the received signal. Thereafter, synthesizing and outputting (S40) an output signal based on the determined LPC parameters and excitation signal. In addition, modifying S35 the determined excitation signal by reducing power and spectral fluctuations of the excitation signal to provide a smoothed output signal.

Abstract: The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilizing high frequency reconstruction (HFR). It utilizes a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit.

Abstract: Methods and apparatus are disclosed for controlling a buffer in a communication system, such as a digital audio broadcasting (DAB) communication system. A more consistent perceptual quality over time provides for a more pleasing auditory experience to a listener. The disclosed bit allocation process determines, for each frame, a distortion d[k] at which the frame is to be encoded. The distortion d[k] is determined to minimize (i) the probability for a buffer overflow, and (ii) the variation of perceived distortion over time. A buffer level is controlled by partitioning a signal into a sequence of successive frames; estimating a distortion rate for a number of frames; and selecting a distortion such that the variance of the buffer level is bounded by a specified value.

Abstract: An apparatus and method for adaptive sub-band allocation of spectral coefficients are disclosed. The sizes of sub-bands are determined according to the distribution of spectral coefficients transformed from an input speech/audio signal to perform more elaborate quantization in units of sub-bands. Thus, quantization noise of the spectral coefficients is reduced, and sound quality in a frequency region is enhanced, thereby improving the quality of the signal.

Abstract: Disclosed are apparatus and methods for generating synthesized utterances. A computing device can receive speech data corresponding to spoken utterances of a particular speaker. Textual elements of an input text corresponding to the speech data can be recognized. Confidence levels associated with the recognized textual elements can be determined. Speech-synthesis parameters of decision trees can be adapted based on the speech data, recognized textual elements, and associated confidence levels. Each adapted decision tree can map individual elements of a text to individual of the speech-synthesis parameters. A second input text can be received. The second input text can be mapped to speech-synthesis parameters using the adapted decision trees. A synthesized spoken utterance can be generated corresponding to the second input text using the speech-synthesis parameters. At least some of the speech-synthesis parameters are configured to simulate the particular speaker.

Abstract: The present invention relates to speech coding in wireless and wireline communication systems. The present invention provides a method of saving bandwidth by a controlled dropping of speech frames at an encoder in a sending communication device. The dropping is controlled in a manner to minimize the effects on the speech quality after the decoding in the receiving communication device, by assuring that the state mismatch between the encoder and the decoder is removed or at least significantly reduced. This is achieved by letting the encoder run an ECU algorithm with a similar behavior as the one running in the decoder in the receiving communication device.

Abstract: An encoder capable of reducing the degradation of the quality of the decoded signal in the case of band expansion in which the high band of the spectrum of an input signal is estimated from the low band. In this encoder, a first layer encoder encodes an input signal and generates first encoded information, a first layer decoder decodes the first encoded information and generates a first decoded signal, a characteristic judger analyzes the intensity of the harmonic structure of the input signal and generates harmonic characteristic information representing the analysis result, and a second layer encoder changes, on the basis of the harmonic characteristic information, the numbers of bits allocated to parameters included in second encoded information created by encoding the difference between the input signal and the first decoded signal before creating the second information.

Abstract: A method and device for coding an input sound signal in at least one lower layer and at least one upper layer of an embedded codec comprises, in the at least one lower layer, coding the input sound signal to produce coding parameters, wherein coding the input sound signal comprises producing a synthesized sound signal. An error signal is computed as a difference between the input sound signal and the synthesized sound signal and a spectral mask is calculated as a function of a minima of a spectrum related to the input sound signal. In the at least one upper layer, the error signal is coded to produce coding coefficients, the spectral mask is applied to the coding coefficients, and the masked coding coefficients are quantized. Applying the spectral mask to the coding coefficients reduces the quantization noise produced upon quantizing the coding coefficients.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: A method, device and system for signal encoding and decoding are disclosed. The method includes: encoding a core layer signal to obtain a core layer signal code; selecting an enhancement sample point that requires enhancement layer signal encoding according to the core layer signal code and the number of bits that can be used by an enhancement layer; obtaining an enhancement layer signal code of the enhancement sample point; and outputting a bit stream, where the bit stream includes the core layer signal code and the enhancement layer signal code. In embodiments of the present invention, according to the number of bits that can be used by the enhancement layer, the enhancement sample point that requires enhancement layer signal encoding is selected; the enhancement layer signal of the selected enhancement sample point is encoded and decoded; when no sufficient bits are available for the enhancement layer, the enhancement quality of the core layer can be improved.

Abstract: An encoder improves inter-channel prediction (ICP) performance in scalable stereo sound encoding using an ICP. In the encoder, ICP analysis units use, as reference signal candidates, a frequency coefficient in the low-band portion of a side residual signal, a frequency coefficient in each sub-band portion of a monaural residual signal, and a frequency coefficient in the low-band portion of the monaural residual signal, respectively, and perform an ICP analysis between the these respective candidates and a frequency coefficient in each sub-band portion of the side residual signal to generate first, second, and third ICP coefficients.

Abstract: A system for coding a hierarchical audio signal, comprising, at least, a core layer using parametric coding by analysis by synthesis in a first frequency band, a band extension layer for widening said first frequency band into a second frequency band, or wideband. The system also comprises a wideband audio coding quality enhancement layer based on transform coding using a spectral parameter obtained from said band extension layer. Application to transmitting speech and/or audio signals over packet networks.

Abstract: Apparatus and methods are provided for measuring perceptual quality of a signal transmitted over a communication network, such as a circuit-switching network, packet-switching network, or a combination thereof. In accordance with one embodiment, a distributed apparatus is provided for measuring perceptual quality of a signal transmitted over a communication network. The distributed apparatus includes communication ports located at various locations in the network. The distributed apparatus may also include a signal processor including a processor for providing non-intrusive measurement of the perceptual quality of the signal. The distributed apparatus may further include recorders operatively connected to the communication ports and to the signal processor, wherein at least one of the recorders processes the signal at one of the communication ports and the recorder sends the signal to the signal processor to measure the perceptual quality of the signal.

Abstract: An encoding device capable of realizing a scalable CODEC of a high performance. In this encoding device, an LPC analyzing unit (551) analyzes an input voice (301) efficiently with a synthesized LPC parameter obtained from a core decoder (305), to acquire an encoded LPC coefficient. An adaptive code note (552) is stored with its sound source codes, as acquired from the core decoder (305). The adaptive code note (552) and a stochastic code note (553) send sound source samples to a gain adjusting unit (554). This gain adjusting unit (554) multiplies the individual sound source samples by an amplification based on the gain parameters acquired from the core decoder (305), and then adds the products to acquire sound source vectors. These vectors are sent to an LPC synthesizing unit (555). This LPC synthesizing unit (555) filters the sound source vectors acquired at the gain adjusting unit (554), with the LPC parameter, to acquire a synthetic signal.

Abstract: Provided is an audio encoding device which can detect an optimal pitch pulse when using pitch pulse information as redundant information.

Abstract: A bandwidth extension module, and an associated method and computer-readable medium, suitable for use in artificially extending the bandwidth of a lowband speech signal. The bandwidth extension module comprises a band-pass filter configured to produce a band-pass signal from the lowband speech signal; at least one carrier frequency modulator, each carrier frequency modulator configured to pitch-synchronously modulate the band-pass signal about a respective carrier frequency, the at least one carrier frequency modulator collectively producing a highband speech signal component; a synthesis filter configured to determine a highband speech signal based on the highband speech signal component; and a summation module configured to combine the lowband speech signal with the highband speech signal to obtain a bandwidth-extended speech signal.

Abstract: An encoding device improves the sound quality of a stereo signal while maintaining a low bit rate. The encoding device includes: an LP inverse filter which LP-inverse-filters a left signal L(n) by using an inverse quantization linear prediction coefficient AdM(z) of a monaural signal; a T/F conversion unit which converts the left sound source signal Le(n) from a temporal region to a frequency region; an inverse quantizer which inverse-quantizes encoded information Mqe; spectrum division units which divide a high-frequency component of the sound source signal Mde(f) and the left signal Le(f) into a plurality of bands; and scale factor calculation units which calculate scale factors ai and ssi by using a monaural sound source signal Mdeh,i(f), a left sound source signal Leh,i(f), Mdeh,i(f), and right sound source signal Reh,i(f) of each divided band.

Abstract: In a device configurable to encode speech performing an closed loop re-decision may comprise representing a speech signal by amplitude components and phase components for a current frame and a past frame. In a first closed loop stage, a first set of compressed components and a first set of uncompressed components for a current frame may be generated. A first set of features may be generated by comparing current and past frame amplitude and/or phase components. In a second closed loop stage, a second set of compressed components for the current frame may be generated by compressing the first set of compressed components and compressing the first set of uncompressed components. Generation of a second set of features may be based on the second set of compressed components from the current frame and a combination of amplitude and/or phase components from the past frame.

Abstract: Audio encoding method and device comprising the transmission, in addition to the data representing a frequency-limited signal, of information relating to a temporal filter that is to be applied to the entire enhanced signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter for reshaping the reconstituted high-frequency part and the correction of compression artefacts present in the transmitted low-frequency part. In this way, the application of the temporal filter, simple and inexpensive, to all or part of the reconstituted signal, makes it possible to obtain a signal of good perceived quality.

Abstract: A multi-reference quantization device and method for quantizing an input LPC filter, comprises a plurality of differential quantizers using respective, different references, and a selector of a reference amongst the different references of the differential quantizers using a reference selection criterion. The input LPC filter is differentially quantized by the differential quantizer using the selected reference. A device and method for inverse quantizing a multi-reference differentially quantized LPC filter extracted from a bitstream, comprises an extractor from the bitstream of information about a reference amongst a plurality of possible references used for quantizing the multi-reference differentially quantized LPC filter, and a differential inverse quantizer using the reference corresponding to the extracted reference information to inverse quantize the multi-reference differentially quantized LPC filter.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed, by which the audio signal can be efficiently processed. The present invention includes obtaining start position information of a sub-frame from a header of the main frame and processing an audio signal based on the start position information of the sub-frame, wherein the main frame includes a plurality of sub-frames.

Abstract: For audio encoding and decoding, in order to enhance coded audio signals, the audio signal is divided into at least a low frequency band and a high frequency band, the high frequency band is divided into at least two high frequency sub-band signals, and parameters are generated that refer at least to the low frequency band signal sections which match best with high-frequency sub-band signals.

Abstract: An information extraction unit extracts spectral envelope information of L-dimension from each frame of speech data by discrete Fourier transform. The spectral envelope information is represented by L points. A basis storage unit stores N bases (L>N>1). Each basis is differently a frequency band having a maximum as a peak frequency in a spectral domain having L-dimension. A value corresponding to a frequency outside the frequency band along a frequency axis of the spectral domain is zero. Two frequency bands of which two peak frequencies are adjacent along the frequency axis partially overlap. A parameter calculation unit minimizes a distortion between the spectral envelope information and a linear combination of each basis with a coefficient for each of L points of the spectral envelope information by changing the coefficient, and sets the coefficient of each basis from which the distortion is minimized to a spectral envelope parameter of the spectral envelope information.

Abstract: A post-filtering apparatus and method for speech enhancement in a modified discrete cosine transform (MDCT) domain are disclosed. In the apparatus and method, previous and current MDCT coefficients are used for obtaining a speech spectrum coefficient similar to a real speech spectrum, and a convex function is used for transforming the speech spectrum coefficient and obtaining a post-filter coefficient so that difference can increase in the case where the speech spectrum coefficient is small but decrease in the case where the coefficient is large. Then, the post-filter coefficient is applied to the MDCT coefficient. With this configuration, both the current and previous MDCT values are used, so that it is possible to obtain a spectrum coefficient similar to the real speech spectrum and to obtain a more accurate filter coefficient. Further, the coefficient is adaptively transformed through the convex function, thereby enhancing speech quality.