Abstract: A scalable encoding apparatus capable of suppressing the quality degradation of a decoded signal without increasing the bit rate. In this apparatus, a core layer encoding part (101) and an extended layer encoding part (102) encode an input signal for each of audio frames. When a replacement determining part (103) determines that a degree to which the input signal changes between a preceding frame and a current frame is equal to or greater than a predetermined value or that a degree, to which the quality of the decoded signal is improved by an extended layer encoding process in the preceding frame, is equal to less than a predetermined level, a replacing part (105) replaces a part of an extended layer encoded data of the preceding frame by a core layer encoded data of the current frame. That is, a transmitting part (108) transmits, as a backup, the core layer encoded data of the current frame to a decoding end in advance.

Abstract: A method for supporting an encoding of an audio signal is shown, wherein at least a first and a second coder mode are available for encoding a section of the audio signal. The first coder mode enables a coding based on two different coding models. A selection of a coding model is enabled by a selection rule which is based on signal characteristics which have been determined for a certain analysis window. In order to avoid a misclassification of a section after a switch to the first coder mode, it is proposed that the selection rule is activated only when sufficient sections for the analysis window have been received. The invention relates equally to a module in which this method is implemented, to a device and a system comprising such a module and to a software program product including a software code for realizing the proposed method.

Abstract: A speech recognition client sends a speech stream and control stream in parallel to a server-side speech recognizer over a network. The network may be an unreliable, low-latency network. The server-side speech recognizer recognizes the speech stream continuously. The speech recognition client receives recognition results from the server-side recognizer in response to requests from the client. The client may remotely reconfigure the state of the server-side recognizer during recognition.

Abstract: A system and apparatus for establishing whether a received signal frame is an audio signal frame is disclosed. In one embodiment, the system includes a predetermined position in an audio signal frame containing a piece of secondary information for an audio characteristic of the audio data, with a selection device for selecting a succession of bits which is arranged at the predetermined position in the received signal frame. A decision-making device flags the received signal frame as an audio signal frame if the succession of bits represents the piece of secondary information.

Abstract: There is described a method of encoding an input signal (20) to generate a corresponding encoded output signal (30), and also encoders (10) arranged to implement the method. The method comprises steps of: (a) distributing the input signal to sub-encoders (300, 310, 320) of the encoder (10); (b) processing the distributed input signal (20) at the sub-encoders (300, 310, 320) to generate corresponding representative parameter outputs (200, 210, 220) from the sub-encoders (300, 310, 320); and (c) combining the parameter outputs (200, 210, 220) of the sub-encoders (300, 310, 320) to generate the encoded output signal (30). Processing of the input signal (20) in the sub-encoders (300, 310, 320) involves segmenting the input signal (20) for analysis, such segments having associated temporal durations which are dynamically variable at least partially in response to information content present in the input signal (20).

Abstract: A one-step correction mechanism for voice interaction is provided. Correction of a previous state is enabled simultaneously with recognition in a current or subsequent state. An application is decomposed into a set of tasks. Each task is associated with the collection of one piece of information. Each task may be in a different state. At any point during the interaction, while a task/state pair is active, the dialog manager may enable multiple other task/state pairs to be active in latent fashion. The application developer may then use those facilities or resources to the active task/state and the latent task/state pairs depending on contextual condition of the interaction state of the application.

Abstract: A method for playing DVD. A method for simultaneously outputting a DVD movie to multiple channels each having its own playback parameters such as view angle, spoken language and subtitle language, using a single DVD player.

Abstract: An apparatus and method for the event-driven analysis of media contents derived from customer interactions is disclosed. Content analysis is executed exclusively on those segments of the interaction media that are relevant in a given context. The steps of the analysis are carried out either in a static or dynamic manner where less demanding on resources analysis type is performed prior to more demanding analysis type.

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.

Abstract: A method is disclosed that enables the transmission of a digital message along with a corresponding media information signal, such as audio or video. A telecommunications device that is processing the information signal from its user, such as a speech signal, encodes the information signal by using a model-based compression coder. One such device is a telecommunications endpoint. Then, based on an evaluation of the perceptual significance of each encoded bit, or on some other meaningful characteristic of the signal, the endpoint's processor: (i) determines which encoded bits can be overwritten; and (ii) intersperses the digital message bits throughout the encoded signal in place of the overwritten bits. The endpoint then transmits those digital message bits as part of the encoded information signal. In this way, no additional bits are appended to the packet to be transmitted, thereby addressing the issue of compatibility with existing protocols and firewalls.

Abstract: A method is disclosed that enables the transmission of a digital message along with a corresponding information signal, such as audio or video. The supplemental information contained in digital messages can be used for a variety of purposes, such as enabling or enhancing packet authentication. In particular, a telecommunications device that is processing an information signal from its user, such as a speech signal, encrypts the information signal by performing a bitwise exclusive-or of an encryption key stream with the information signal stream. The device, such as a telecommunications endpoint, then intersperses the bits of the digital message throughout the encrypted signal in place of those bits overwritten, in a process referred to as “watermarking.” The endpoint then transmits the interspersed digital message bits as part of a composite signal that also comprises the encrypted information bits. No additional bits are appended to the packet to be transmitted, thereby addressing compatibility issues.

Abstract: A method of mitigating errors in a distributed speech recognition process. The method comprises the steps of identifying a group comprising one or more vectors which have undergone a transmission error, and replacing one or more speech recognition parameters in the identified group of vectors. In one embodiment all the speech recognition parameters of each vector of the group are replaced by replacing the whole vectors, and each respective replaced whole vector is replaced by a copy of whichever of the preceding or following vector without error is closest in receipt order to the vector being replaced.

Abstract: A method, a system, and a media gateway (MG) for reporting media instance information are disclosed. The method for reporting media instance information includes: detecting, by an MG, received media data according to a set media instance detection (MID) event; and reporting, by the MG, the MID event when the media instance information is detected. With the present invention, the MG reports the detected media instance information related to the media data to a media gateway controller (MGC) through a set MID event, so that the MG can detect media instance information related to the media data, and report the detected media instance information related to the media data to the MGC. In this way, the MGC can execute corresponding control operations according to the media instance information related to the media data, extending the applicable scope of media services.

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 method and device for updating statuses of synthesis filters are provided. The method includes: exciting a synthesis filter corresponding to a first encoding rate by using an excitation signal of the first encoding rate, outputting reconstructed signal information, and updating status information of the synthesis filter and a synthesis filter corresponding to a second encoding rate. In the present disclosure, the status of the synthesis filter corresponding to the current rate and the statuses of the synthesis filters at other rates are updated. Thus, synchronization between the statuses of the synthesis filters corresponding to different rates at the encoding terminal may be realized, thereby facilitating the consistency of the reconstructed signals of the encoding and decoding terminals when the encoding rate is switched, and improving the quality of the reconstructed signal of the decoding terminal.

Abstract: An audio decoder which reproduces original signals from a bit stream including (i) a downmix signal of the original signals, and (ii) supplementary information indicating a gain ratio D and phase difference ? between the original signals. The audio decoder includes: a decoding unit extracting the downmix signal; a transformation unit transforming the extracted downmix signal into a frequency domain signal; a phase rotator determination unit determining two phase rotators having, as the phase rotation angles, angles ? and ? respectively obtained by dividing a contained angle by a diagonal of a parallelogram; a separation unit separating the frequency domain signal into two separation signals respectively indicating angles ? and ? as phase differences between the signals and the decoded downmix signal; and an inverse transformation unit inversely transforming the respective two separation signals into time domain signals so as to reproduce the two audio signals.

Abstract: Method and apparatus for encoding/decoding audio data with scalability are provided. The method includes slicing audio data so that sliced audio data corresponds to a plurality of layers, obtaining scale band information and coding band information corresponding to each of the plurality of layers, coding additional information containing scale factor information and coding model information based on scale band information and coding band information corresponding to a first layer, obtaining quantized samples by quantizing audio data corresponding to the first layer with reference to the scale factor information, coding the obtained plurality of quantized samples in units of symbols in order from a symbol formed with most significant bits (MSB) down to a symbol formed with least significant bits (LSB) by referring to the coding model information, and repeatedly performing the steps with increasing the ordinal number of the layer one by one every time, until coding for the plurality of layers is finished.

Abstract: Methods and apparatus for coordinating audio data processing and network communication processing in a communication device. In an exemplary method lower and upper threshold values for use by a network communication processing circuit are set, the lower and upper threshold values defining a window of timing offsets relative to each of a series of periodic network communications frame boundaries. A series of encoded audio data frames are sent to the network communication processing circuit for transmission over the network communications link. The delivery of encoded audio data to the network communication processing circuit outside of the corresponding time window defined by the threshold values will trigger an event report. This event report is received from the network communication processing circuit by the audio data processing circuit, and, in response, timing is adjusted for the sending of one or more of the encoded audio data frames.

Abstract: Encoding a sequence of digital speech samples into a bit stream includes dividing the digital speech samples into one or more frames and computing a set of model parameters for the frames. The set of model parameters includes at least a first parameter conveying pitch information. The voicing state of a frame is determined and the first parameter conveying pitch information is modified to designate the determined voicing state of the frame, if the determined voicing state of the frame is equal to one of a set of reserved voicing states. The model parameters are quantized to generate quantizer bits which are used to produce the bit stream.

Abstract: The transient problem may be sufficiently addressed, and for this purpose, a further delay on the side of the decoding may be reduced if a new SBR frame class is used wherein the frame boundaries are not shifted, i.e. the grid boundaries are still synchronized with the frame boundaries, but wherein a transient position indication is additionally used as a syntax element so as to be used, on the encoder and/or decoder sides, within the frames of these new frame class for determining the grid boundaries within these frames.

Abstract: In one embodiment, a method for providing voice quality assurance is provided. The method determines voice information for an end point in a voice communication system. The voice information may be from an ingress microphone. The method determines if the voice quality is considered degraded based on an analysis of the voice information. For example, the voice information may indicate that it is distorted, too loud, too soft, is subject to an external noise, etc. Feedback information is determined if the voice quality is considered degraded where the feedback information designed to improve voice quality at an ingress point for a user speaking. The feedback information is then outputted at the end point to the user using the end point.

Abstract: By individuals in an organization possessing a portable node and acquiring sound information, an electronic device and a system that analyze human behavior, group formation, and an activity level of the organization and a system is provided. In an electronic device that has a radio communication unit and a microcomputer, sound is converted into an electrical signal by a microphone, and sound waveform information and characteristic information of sound are obtained using an amplifier, a filter, and an envelope generating circuit. An envelope signal being characteristic information of sound is compared with a reference value by a comparator. Characteristic information of sound when the signal does not reach the reference value, and sound waveform information when greater than the reference value are transmitted from the radio communication unit. The server learns human behavior, group formation; and an activity level of an organization by receiving and analyzing these pieces of information.

Abstract: There is provided an information processing device including a storage unit that stores music data for playing music and lyrics data indicating lyrics of the music, a display control unit that displays the lyrics of the music on a screen, a playback unit that plays the music and a user interface unit that detects a user input. The lyrics data includes a plurality of blocks each having lyrics of at least one character. The display control unit displays the lyrics of the music on the screen in such a way that each block included in the lyrics data is identifiable to a user while the music is played by the playback unit. The user interface unit detects timing corresponding to a boundary of each section of the music corresponding to each displayed block in response to a first user input.

Abstract: A speech signal processing system comprises an audio processor (103) for providing a first signal representing an acoustic speech signal of a speaker. An EMG processor (109) provides a second signal which represents an electromyographic signal for the speaker captured simultaneously with the acoustic speech signal. A speech processor (105) is arranged to process the first signal in response to the second signal to generate a modified speech signal. The processing may for example be a beam forming, noise compensation, or speech encoding. Improved speech processing may be achieved in particular in an acoustically noisy environment.

Abstract: There is provided a method for use by a speech encoder to encode an input speech signal.

Abstract: A method and corresponding apparatus for coded-domain acoustic echo control is presented. An echo control problem is considered as that of perceptually matching an echo signal to a reference signal. A perceptual similarity function that is based on the coded spectral parameters produced by the speech codec is defined. Since codecs introduce a significant degree of non-linearity into the echo signal, the similarity function is designed to be robust against such effects. The similarity function is incorporated into a coded-domain echo control system that also includes spectrally-matched noise injection for replacing echo frames with comfort noise. Using actual echoes recorded over a commercial mobile network, it is shown herein that the similarity function is robust against both codec non-linearities and additive noise. Experimental results further show that the echo-control is effective at suppressing echoes compared to a Normalized Least Mean Squared (NLMS)-based echo cancellation system.

Abstract: A method of processing a decoded speech (DS) signal including successive DS frames, each DS frame including DS samples. The method comprises: adaptively filtering the DS signal to produce a filtered signal; gain-scaling the filtered signal with an adaptive gain updated once a DS frame, thereby producing a gain-scaled signal; and performing a smoothing operation to smooth possible waveform discontinuities in the gain-scaled signal.

Abstract: A method for managing an interaction of a calling party to a communication partner is provided. The method includes automatically determining if the communication partner expects DTMF input. The method also includes translating speech input to one or more DTMF tones and communicating the one or more DTMF tones to the communication partner, if the communication partner expects DTMF input.

Abstract: In one embodiment, a method includes receiving at a communication device an audio communication and a transcribed text created from the audio communication, and generating a mapping of the transcribed text to the audio communication independent of transcribing the audio. The mapping identifies locations of portions of the text in the audio communication. An apparatus for mapping the text to the audio is also disclosed.

Abstract: A method and apparatus for audio biometric data capture are provided. The apparatus includes in interactive head-mounted eyepiece worn by a user that includes an optical assembly through which a user views a surrounding environment and displayed content. The optical assembly comprises a corrective element that corrects the user's view of the surrounding environment and an integrated processor for handling content to the user. An integrated optical sensor captures visual biometric data when the eyepiece is positioned so that a nearby individual is proximate to the eyepiece. Audio biometric data is captured using multiple microphones mounted in an endfire array in the eyepiece. The remote processing facility interprets the captured audio biometric data and generates display content based on the interpretation. This display content is delivered to the eyepiece and displayed to the user.

Abstract: A system is disclosed for facilitating speech recognition and transcription among users employing incompatible protocols for generating, transcribing, and exchanging speech. The system includes a system transaction manager that receives a speech information request from at least one of the users. The speech information request includes formatted spoken text generated using a first protocol. The system also includes a speech recognition and transcription engine, which communicates with the system transaction manager. The speech recognition and transcription engine receives the speech information request from the system transaction manager and generates a transcribed response, which includes a formatted transcription of the formatted speech. The system transmits the response to the system transaction manager, which routes the response to one or more of the users. The latter users employ a second protocol to handle the response, which may be the same as or different than the first protocol.

Abstract: The present invention relates to methods and devices for encoding and decoding digital audio signals, e.g. a speech signal. An audio coder and a decoder are provided wherein a modeller adds a first distribution model obtained from model parameters of past segments of the digital audio signal and a fixed distribution model, each of the models being multiplied by a weighting coefficient, for obtaining a combined distribution model. The weighting coefficients are selected to minimize a code length of a current segment of the digital audio signal. As the combined distribution model is a sum of several distribution models, wherein at least some of the models is based on the model parameters, flexibility is introduced in the signal model used to encode the digital audio signal. Thus, an audio coder and decoder providing a low bit rate in average, low bit rate variations and low error propagation are provided.

Abstract: Aspects of a method and system for decoding GSM speech data using redundancy are provided. A decoding algorithm in a frame process may be utilized to generate a bit-sequence for GSM speech data received via a burst process. The decoding algorithm may be a modified Viterbi algorithm, for example. The frame process may comprise verifying a CRC for the bit-sequence and/or decrypting the bit-sequence. In some instances, estimates of the bit-sequence may be fed back to the decoding algorithm. A speech stream that satisfies speech constraints may be generated based on the generated bit-sequence. The speech constraints may comprise gain and/or pitch continuity, for example. The generated speech stream may be decoded via a voice decoder that supports full rate (FR), adaptive multi-rate (AMR), and/or enhanced full-rate (EFR) speech coding. Frame process results may be fed back to the burst process to improve decoding of received GSM speech data.

Abstract: Scaling, by a desired amount sm, the overall perceived loudness Lm of a multichannel audio signal, wherein perceived loudness is a nonlinear function of signal power P, by scaling the perceived loudness of each individual channel Lc by an amount substantially equal to the desired amount of scaling of the overall perceived loudness of all channels sm, subject to accuracy in calculations and the desired accuracy of the overall perceived loudness scaling sm. The perceived loudness of each individual channel may be scaled by changing the gain of each individual channel, wherein gain is a scaling of a channel's power. Optionally, in addition, the loudness scaling applied to each channel may be modified so as to reduce the difference between the actual overall loudness scaling and the desired amount of overall loudness scaling.

Abstract: The invention relates to a method for outputting a speech signal. Speech signal frames are received and are used in a predetermined sequence in order to produce a speech signal to be output. If one speech signal frame to be received is not received, then a substitute speech signal frame is used in its place, which is produced as a function of a previously received speech signal frame. According to the invention, in the situation in which the previously received speech signal frame has a voiceless speech signal, the substitute speech signal frame is produced by means of a noise signal.

Abstract: Techniques implemented as systems, methods, and apparatuses, including computer program products, for logging multi-channel audio signals.

Abstract: Techniques implemented as systems, methods, and apparatuses, including computer program products, for obfuscating sensitive content in an audio source representative of an interaction between a contact center caller and a contact center agent. The techniques include performing, by an analysis engine of a contact center system, a context-sensitive content analysis of the audio source to identify each audio source segment that includes content determined by the analysis engine to be sensitive content based on its context; and processing, by an obfuscation engine of the contact center system, one or more identified audio source segments to generate corresponding altered audio source segments each including obfuscated sensitive content.

Abstract: Audio coding and decoding apparatuses and methods which support fine granularity scalability (FGS) using harmonic information of a high-band audio signal or wideband error audio signal when performing wideband audio coding and decoding, and recording mediums on which the methods are stored. The audio coding method includes detecting harmonics of a high-band audio signal or wideband error audio signal of an input audio signal; determining an order of the detected harmonics; and coding the detected harmonics based on the determined order.

Abstract: An audio decoding system performs frame loss concealment (FLC) when portions of a bit stream representing an audio signal are lost within the context of a digital communication system. The audio decoding system employs two different FLC methods: one designed to perform well for music, and the other designed to perform well for speech. When a frame is deemed lost, the audio decoding system analyzes a previously-decoded audio signal corresponding to previously-decoded frames of an audio bit-stream. Based on the results of the analysis, the lost frame is classified as either speech or music. Using this classification, other signal analysis, and knowledge of the employed FLC methods, the audio decoding system selects the appropriate FLC method which then performs FLC on the lost frame.

Abstract: In one embodiment, the method includes receiving an audio data frame having at least first and second channels. The first and second channels are synchronously subdivided into blocks such that the lengths of the blocks into which the second channel is subdivided correspond to the lengths of the blocks into which the first channel is subdivided if the first and second channels are correlated with each other and difference coding is used. The first and second channels are decoded and the subdivided blocks of the first and second channels are interleaved if the first and second channels are synchronously subdivided.

Abstract: There is disclosed a speech switching device capable of improving quality of a decoded signal. In the device, a weighted addition unit outputs a mixed signal of a narrow-band speech signal and a wide-band speech signal when switching the speech signal band. A mixing unit formed by an extended layer decoded speech amplifier and an adder mixes the narrow-band speech signal with the wide-band speech signal while changing the mixing ratio of the narrow-band speech signal and the wide-band speech signal as the time elapses, thereby obtaining a mixed signal. An extended layer decoded speech gain controller variably sets the degree of change of the mixing ratio by the time.

Abstract: A method and an apparatus for transmitting a speech signal are provided. A speech signal transmitter includes a quadrature mirror filter, a base sub-band encoder, an enhancement sub-band encoder, and a network connector. The quadrature mirror filter receives a speech signal, divides the speech signal into an enhancement band speech signal and a base band speech signal, and outputs the enhancement band speech signal and the base band speech signal. The base sub-band encoder receives and encodes the base band speech signal. The enhancement sub-band encoder receives and encodes the enhancement band speech signal. The network connector multiplexes the encoded enhancement band speech signal and the encoded base band speech signal based on the kinds of networks over which speech signals are transmitted, and transmits the multiplexed signals to the networks. A speech signal is multiplexed and transmitted by various methods based on the kinds of networks. Thus, the speech signal can be efficiently transmitted.

Abstract: The present invention discloses a solution for providing a phonetic representation for a content item along with a content item delivered to a speech enabled computing device. The phonetic representation can be specified in a manner that enables it to be added to a speech recognition grammar of the speech enabled computing device. Thus, the device can recognize speech commands using the newly added phonetic representation that involve the content item. Current implementations of speech recognition systems of this type rely internal generation of speech recognition data that is added to the speech recognition grammar. Generation of speech recognition data can, however, be resource intensive, which can be particularly problematic when the speech enabled device is resource limited. The disclosed solution offloads the task of providing the speech recognition data to an external device, such as a relatively resource rich server or a desktop device.

Abstract: The present invention provides a system and method for representing quasi-periodic (“qp”) waveforms comprising, representing a plurality of limited decompositions of the qp waveform, wherein each decomposition includes a first and second amplitude value and at least one time value. In some embodiments, each of the decompositions is phase adjusted such that the arithmetic sum of the plurality of limited decompositions reconstructs the qp waveform. These decompositions are stored into a data structure having a plurality of attributes. Optionally, these attributes are used to reconstruct the qp waveform, or patterns or features of the qp wave can be determined by using various pattern-recognition techniques. Some embodiments provide a system that uses software, embedded hardware or firmware to carry out the above-described method. Some embodiments use a computer-readable medium to store the data structure and/or instructions to execute the method.

Abstract: A scalable encoder enabling improvement of the encoding efficiency in the second layer and improvement of the quality of the original signal decoded using the encoding signal in the second layer. A predictive coefficient encoder of the scalable encoder has a predictive coefficient codebook where candidates of the predictive coefficient are recorded. After searching the predictive coefficient codebook, the scale factor of the first layer decoded signal inputted from a scale factor calculator is multiplied, and a predictive coefficient which most approximates the multiplication result to the scale factor of the original signal inputted from the scale factor calculator is determined and encoded, and the coded code is inputted to a multiplexer.

Abstract: Packet loss concealment (PLC) systems and methods are described that use time-warping to merge a concealment signal generated to replace one or more bad frames of an audio signal with a received signal representing one or more subsequent good frames of the audio signal in a manner that avoids signal discontinuity and audible artifacts resulting therefrom. Prediction-based PLC systems and methods are also described that use time-warping to conceal the loss of one or more frames containing a transition region in a manner that will not result in an audible artifact.

Abstract: A data embedding device for embedding data in a speech code obtained by encoding a speech in accordance with a speech encoding method based on a voice generation process of a human being, includes an embedding judgment unit, every speech code, judging whether or not data should be embedded in the speech code, and an embedding unit embedding data in two or more parameter codes of a plurality of parameter codes constituting the speech code for which it is judged by the embedding judgment unit that the data should be embedded.

Abstract: A normalization factor for a current frame of a signal may be determined. The normalization factor may depend on an amplitude of the current frame of the signal. The normalization factor may also depend on values of states after one or more operations were performed on a previous frame of a normalized signal. The current frame of the signal may be normalized based on the normalization factor that is determined. The states' normalization factor may be adjusted based on the normalization factor that is determined.

Abstract: An audio recording and reproducing apparatus includes a controller for controlling the entire behaviors, hard disc for write and read of audio data, audio compression/expansion circuit for expanding compressed audio data, and external I/O port. The audio recording and reproducing apparatus is connected to a network service center to obtain desired music data from storage of the network service center and to store it in the hard disc.

Abstract: Estimates of spectral magnitude and phase are obtained by an estimation process using spectral information from analysis filter banks such as the Modified Discrete Cosine Transform. The estimation process may be implemented by convolution-like operations with impulse responses. Portions of the impulse responses may be selected for use in the convolution-like operations to trade off between computational complexity and estimation accuracy. Mathematical derivations of analytical expressions for filter structures and impulse responses are disclosed.