Abstract: A sound encoding device enabling the amount of delay to be kept small and the distortion between frames to be mitigated. In the sound encoding device, a window multiplication part (211) of a long analysis section (21) multiplies a long analysis frame signal of analysis length M1 by an analysis window, the resultant signal multiplied by the analysis window is outputted to an MDCT section (212), and the MDCT section (212) performs MDCT of the input signal to obtain the transform coefficients of the long analysis frame and outputs it to a transform coefficient encoding section (30). The window multiplication part (221) of a short analysis section (22) multiplies a short analysis frame signal of analysis length M2 (M2<M1) by an analysis window and the resultant signal multiplied by the analysis window is outputted to the MDCT section (222).

Abstract: Disclosed are an encoding device and encoding method capable of improving the quality of a decoded signal under very low bit rate conditions using a small amount of computation. A spectrum correction unit (302) performs correction processing on the subspectrum in each subband in such a manner that samples equal to or greater than a subspectrum average value are left unchanged while samples smaller than the subspectrum average value are replaced by zero. As a result of this, it is possible to significantly reduce the number of bits required to quantize the subspectrums without substantial reduction in quality in a local search unit (303) and in a multi-rate indexing unit (304).

Abstract: A method and an apparatus to screen electronic communications have been disclosed. In one embodiment, the method includes extracting URLs from electronic communication and analyzing the URLs extracted to determine whether the electronic communication is of a first predetermined category. Other embodiments have been claimed and described.

Abstract: Methods and apparatus are disclosed for approximating an MDCT coefficient of a block of windowed sinusoid having a defined frequency, the block being multiplied by a window sequence and having a block length and a block index. A finite trigonometric series is employed to approximate the window sequence. A window summation table is pre-computed using the finite trigonometric series and the defined frequency of the sinusoid. A block phase is computed for each block with the defined frequency, the block length and the block index. An MDCT coefficient is approximated by the dot product of a phase vector computed using the block phase with a corresponding row of the window summation table.

Abstract: A “STAC Codec” provides audio transcoding and decoding by processing an encoded audio signal using a backward-adaptive run-length Golomb-Rice (RLGR) decoder to recover transform coefficients of the encoded audio signal. The transform coefficients are then either transcoded in the transform domain to lossy or other formats, or decoded to the time domain by applying an inverse integer-reversible modulated lapped transform (MLT) to the recovered transform coefficients to recover an uncompressed time domain representation compressed audio signal. In additional embodiments, an inter-block spectral estimation and inverse data sorting strategy is used in recovering the transform coefficients from the encoded audio signal.

Abstract: A technique is described for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system. In accordance with the technique, a first synthesized sub-band audio signal is synthesized, wherein synthesizing the first synthesized sub-band audio signal comprises performing waveform extrapolation based on a stored first sub-band decoded audio signal. A second synthesized sub-band audio signal is also synthesized, wherein synthesizing the second synthesized sub-band audio signal comprises performing waveform extrapolation based on the stored second sub-band decoded audio signal. The first synthesized sub-band audio signal and the second synthesized sub-band audio signal are combined to generate a synthesized full-band output audio signal corresponding to a lost frame.

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: A hardware device for analyzing an audio signal comprises a calculator for calculating a neural activity pattern over time resulting at nerve fibers of an ear model based on the audio signal and a processor for processing the neural activity pattern to obtain a sequence of time information as an analysis representation describing a temporal position of consecutive trajectories, wherein a trajectory includes activity impulses on different nerve fibers based on the same event in the audio signal. A two-dimensional representation of the neural activity pattern is gradually distorted over time, and it is recognized when an approximately straight line is contained in the distorted two-dimensional representation of the neural activity pattern over time. Accordingly, a time information belonging to the trajectory is provided.

Abstract: A “STAC Codec” provides lossless audio compression and decompression by processing an audio signal using integer-reversible modulated lapped transforms (MLT) to produce transform coefficients. Transform coefficients are then encoded using a backward-adaptive run-length Golomb-Rice (RLGR) encoder to produce losslessly compressed audio signals. In additional embodiments, further compression gains are achieved via an inter-block spectral estimation and data sorting strategy. Further, compression in the transform domain allows the bitstream to be partially decoded, using the corresponding RLGR decoder, to reconstruct the frequency-domain coefficients. These frequency-domain coefficients are then directly used to speed up various transform-domain based applications such as transcoding media to lossy or other formats, search, identification, visualization, watermarking, etc.

Abstract: A transient noise removal system removes or dampens undesired transients from speech. When the transient noise removal system receives a speech frame, the system performs a wavelet transform analysis. The speech frame may be represented by one or more wavelet coefficients across one or more wavelet levels. For a given wavelet level, the transient noise-removal system may determine a wavelet threshold. The transient noise removal system may compare the threshold corresponding to a wavelet level to the wavelet coefficients within that level. The transient noise removal system may attenuate each wavelet coefficient based on a comparison to a threshold.

Abstract: Method for processing speech signal data. A speech signal is divided into frames. Each frame is characterized by a frame number T representing a unique interval of time. Each speech signal is characterized by a power spectrum with respect to frame T and frequency band ?. A speech segment and a reverberation segment of the speech signal is determined. L filter coefficients W(k) (k=1, 2, . . . , L) respectively corresponding to L frames immediately preceding frame T are computed such that the L filter coefficients minimize a function ? that is a linear combination of sum of squares of a residual speech power in the reverberation segment and a sum of squares of a subtracted speech power in the speech segment. The computed L filter coefficients are stored within storage media of the computing apparatus.

Abstract: Remote configuration and utilization of a virtual tape management system with creation and management options. At least one security administrator CPU is communicably attached to a virtual tape management CPU. A pair of disk drives is communicably attached to the virtual tape management CPU and to the security administrator. First software within the virtual tape management CPU validates authorized remote access to said disk drives and encrypts the data. Second software facilitates remote configuration and utilization of the virtual tape management CPU. Third software provides tape image file processing including inspecting each remote data storage to determine if a file is present, opening the file, reading tape related information thereon, and reading from or writing to the disk arrays. Fourth software provides checker support for tape image files for enumerating tape image files on the disk drives, opening the tape image files, and reading the tape related information.

Abstract: A smoothing method for suppressing fluctuating artifacts in the reduction of interference noise includes the following steps: providing short-term spectra for a sequence of signal frames, transforming each short-term spectrum by way of a forward transformation which describes the short-term spectrum using transformation coefficients that represent the short-term spectrum subdivided into its coarse and fine structures; smoothing the transformation coefficients with the respective same coefficient indices by combining at least two successive transformed short-term spectra; and transforming the smoothed transformation coefficients into smoothed short-term spectra by way of a backward transformation.

Abstract: A spectral representation of an audio signal having consecutive audio frames can be derived more efficiently, when a common time warp is estimated for any two neighboring frames, such that a following block transform can additionally use the warp information. Thus, window functions required for successful application of an overlap and add procedure during reconstruction can be derived and applied, the window functions already anticipating the re-sampling of the signal due to the time warping. Therefore, the increased efficiency of block-based transform coding of time-warped signals can be used without introducing audible discontinuities.

Abstract: A method generates envelope spectra and harmonic spectra from an input broad-band training acoustic signal. Corresponding non-negative envelope bases are trained for the envelope spectra and non-negative harmonic bases are trained for the harmonic spectra using convolutive non-negative matrix factorization. Higher-band frequencies are generated for an input lower-band acoustic signal according to the non-negative envelope bases and the non-negative harmonic bases. Then, the input lower-band acoustic signal is combined with the higher-band frequencies to produce an output broad-band acoustic signal.

Abstract: A voice and musical tone coding apparatus is provided that can perform high-quality coding by executing vector quantization taking the characteristics of human hearing into consideration. In this voice and musical tone coding apparatus, a quadrature transformation processing section (201) converts a voice and musical tone signal from time components to frequency components. An auditory masking characteristic value calculation section (203) finds an auditory masking characteristic value from a voice and musical tone signal. A vector quantization section (202) performs vector quantization changing a calculation method of a distance between a code vector found from a preset codebook and a frequency component based on an auditory masking characteristic value.

Abstract: A method detects components of a non-stationary signal. The non-stationary signal is acquired and a non-negative matrix of the non-stationary signal is constructed. The matrix includes columns representing features of the non-stationary signal at different instances in time. The non-negative matrix is factored into characteristic profiles and temporal profiles.

Abstract: A storage controller, cooperable with host computer apparatus, and a plurality of controlled storage apparatus, comprises a host write component operable to write a data object to a source data image at one of the plurality of controlled storage apparatus; a first copy component responsive to a first metadata state and operable to control copying of the data object to a first target data image at one of the plurality of controlled storage apparatus; a second copy component responsive to a second metadata state and operable to perform one of: controlling copying of the data object to a second target data image at one of the plurality of controlled storage apparatus; and causing the first copy component to perform copying of the second target data image to the first target data image.

Abstract: An audio decoding device of the present invention includes: a decoding unit decoding a stream to a spectrum coefficient, and outputting stream information when a frame included in the stream cannot be decoded; an orthogonal transformation unit transforming the spectrum coefficient to a time signal; a correction unit generating a correction time signal based on an output waveform within a reference section that is in a section that overlaps between an error frame section to which the stream information is outputted and an adjacent frame section and that is a section in the middle of the adjacent frame section, when the decoding unit outputs the stream information: and an output unit generating the output waveform by synthesizing the correction time signal and the time signal.

Abstract: There is provided a speech post-processor for enhancing a speech signal divided into a plurality of sub-bands in frequency domain. The speech post-processor comprises an envelope modification factor generator configured to use frequency domain coefficients representative of an envelope derived from the plurality of sub-bands to generate an envelope modification factor for the envelope derived from the plurality of sub-bands, where the envelope modification factor is generated using FAC=?ENV/Max+(1??), where FAC is the envelope modification factor, ENV is the envelope, Max is the maximum envelope, and a is a value between 0 and 1, where ? is a different constant value for each speech coding rate. The speech post-processor further comprises an envelope modifier configured to modify the envelope derived from the plurality of sub-bands by the envelope modification factor corresponding to each of the plurality of sub-bands.

Abstract: An apparatus for processing a speech signal includes a receiver, a speech signal decoder, a speech rate conversion information detector, and a speech rate converting processor. The receiver receives multiplexed signal of information concerning controls and programs, including speech packets through a transmission line. The decoder decodes the speech signal of packets out of the received signals. The detector detects speech rate conversion execution information in the received signals. The processor subjects the decoded speech signal to a speech rate conversion process if the speech rate conversion execution information indicates that the speech signal has not been subjected to the speech rate conversion process on the transmitting end, and which does not subject the decoded speech signal to the speech rate conversion process if the speech rate conversion execution information indicates that the speech signal has been subjected to the speech rate conversion process on the transmitting end.

Abstract: A method and system separates components in individual signals, such as time series data streams. A single sensor acquires concurrently multiple individual signals. Each individual signal is generated by a different source. An input non-negative matrix representing the individual signals is constructed. The columns of the input non-negative matrix represent features of the individual signals at different instances in time. The input non-negative matrix is factored into a set of non-negative bases matrices and a non-negative weight matrix. The set of bases matrices and the weight matrix represent the individual signals at the different instances of time.

Abstract: A method of sending a voice message via a mobile communication device, said method involving: receiving an utterance from a user of the mobile communication device; generating a non-text representation of the received utterance; inserting the non-text representation into a body of a text message; and sending the text message over a wireless messaging channel from the mobile communication device to a recipient's device.

Abstract: A storage controller, cooperable with host computer apparatus, and a plurality of controlled storage apparatus, comprises a host write component operable to write a data object to a source data image at one of the plurality of controlled storage apparatus; a first copy component responsive to a first metadata state and operable to control copying of the data object to a first target data image at one of the plurality of controlled storage apparatus; a second copy component responsive to a second metadata state and operable to perform one of: controlling copying of the data object to a second target data image at one of the plurality of controlled storage apparatus; and causing the first copy component to perform copying of the second target data image to the first target data image.

Abstract: A method and apparatus is disclosed in which a storage controller cooperable with a host and a plurality of controlled storage is provided to localize an impact of a failure to a target disk in an affected segment. The storage controller includes a host write component to write a data object to a source image storage; a first copy component responsive to a first metadata state to control copying of the data object to a first target storage; a second copy component responsive to a second metadata state to perform either: copying the data object to a second target or causing the first copy component to copy the second target to the first target; and a third copy component to control cascaded copying of the data object to a third target storage. Either the second or the third copy component controls cascaded copying of a delimited data image subsequence responsive to a metadata state indicating currency of a data grain in either the second or the third target.

Abstract: A method for discriminating noise from signal in a noise-contaminated signal involves decomposing a frame of samples of the signal into decorrelated components, and using a difference between probability distributions of the noise contributions and the signal contributions to identify signal and noise. A Gaussian distribution is used to determine whether the components are only noise whereas a Laplacian distribution is used to determine whether the components contain the signal. Such discrimination may be used in speech enhancement or voice activity detection apparatus.

Abstract: A method and apparatus for generating a graphical display report. The method includes representing a plurality of events as one or more icons; and positioning the icons inside two or more concentric circles defining at least an outer circle and an inner circle disposed inside the outer circle. The inner circle is configured to contain icons corresponding to events having a first status. The outer circle is configured to contain icons corresponding to events having a second status. The outer circle defines a plurality of tick marks disposed at a periphery of the outer circle, wherein the tick marks represent an attribute.

Abstract: A system for managing message distributions in multi-messaging system includes a server, at least one messaging request system and at least one messaging delivery system connected to the server. The server includes a message managing module which is programmed for receiving at least one messaging request from the at least one messaging request system, creating a message assignment for each messaging request, and sending each massage assignment to a corresponding massage receiving system. A related method is also disclosed. Utilizing the system and method can manage and maintain system messages in an unified way.

Abstract: A signal encoding apparatus (10) limits an inputted time series signal to a low frequency band signal having a certain cut-off frequency or less to include the low frequency band signal into code train for outputting encoded low frequency band code train. In addition, the signal encoding apparatus (10) adaptively determines aliasing frequency fa, shift frequency fsh or tone•noise synthesis information r used for generation of high frequency band signal at the decoding side to include these information into code train outputted along with high frequency band spectrum envelope information as high frequency band generation information.

Abstract: A quantizer finds a quantization threshold using a quantization loop with a heuristic approach. Following the heuristic approach reduces the number of iterations in the quantization loop required to find an acceptable quantization threshold, which instantly improves the performance of an encoder system by eliminating costly compression operations. A heuristic model relates actual bit-rate of output following compression to quantization threshold for a block of a particular type of data. The quantizer determines an initial approximation for the quantization threshold based upon the heuristic model. The quantizer evaluates actual bit-rate following compression of output quantized by the initial approximation. If the actual bit-rate satisfies a criterion such as proximity to a target bit-rate, the quantizer sets accepts the initial approximation as the quantization threshold. Otherwise, the quantizer adjusts the heuristic model and repeats the process with a new approximation of the quantization threshold.

Abstract: A signal processing device includes a biorthogonal filter bank that processes a finite length signal including a left boundary and a right boundary. The biorthogonal filter bank includes an analysis filter bank. The analysis filter bank includes one or more left boundary filters, one or more right boundary filters, and one or more steady-state analysis filters. Each left boundary filter and each right boundary filter includes a row vector.

Abstract: A system and method for detecting beats in a compressed audio domain is disclosed where a beat detector functions as part of an error concealment system in an audio decoding section used in audio information transfer and audio download-streaming system terminal devices such as mobile phones. The beat detector includes a MDCT coefficient extractor, a band feature value analyzer, a confidence score calculator; and a converging and storage unit. The method provides beat detection by means of beat information obtained using both MDCT coefficients as well as window-switching information. A baseline beat position is determined using MDCT coefficients obtained from the audio bitstream which also provides a window-switching pattern. A window-switching beat position is compared with the baseline beat position and, if a predetermined condition is satisfied, the window-switching beat position is validated as a detected beat.

Abstract: This invention is directed toward improving the subjective voice quality of a mobile to mobile phone call having tandem vocoder processing by modifying the spectrum of the voice signal before it is processed by the second vocoder to compensate for digital distortion which is generated by the second vocoder. An adaptive filter can be used to modify the spectrum of the voice signal. With this invention, the voice quality of a call from a first mobile phone to a second mobile phone has a quality that is substantially similar to the voice quality of a call from a mobile phone to a desk phone.

Abstract: An application server generates and maintains a server-side data record, also referred to as a “brownie”, that includes application state information and user attribute information for multiple users within a single session controlled by a web-based browser. The brownie includes a session identifier that uniquely identifies the session, and a subsession identifier that uniquely identifies each corresponding user of the application session. As each new user is added to the session, for example by initiating a call to the new user, the application server stores the subsession identifier and corresponding application state information for the new user in the same brownie. In response to receiving a second web page request from the browser that includes the session identifier, the application server initiates a new web application instance, and recovers the brownie from the memory based on the session identifier included in the second page request.

Abstract: Partial waveform data representative of a waveform shape variation are extracted from supplied waveform data, and the extracted partial waveform data are stored along with time position information indicative of their respective time positions. In reproduction, the partial waveform data and time position information are read out, then the partial waveform data are arranged on the time axis in accordance with the time position information, and a waveform is produced on the basis of the waveform data arranged on the time axis. In another implementation, sets of sample identification information and time position information are obtained in accordance with a performance tone waveform to be reproduced, and sample data are obtained from a database in accordance with the sample identification information. The thus-obtained sample data are arranged on the time axis in accordance with the time position information, and the desired waveform is produced on the basis of the sample data arranged on the time axis.

Abstract: A method for implementing a noise suppressor in a speech recognition system comprises a filter bank for separating source speech data into discrete frequency sub-bands to generate filtered channel energy, and a noise suppressor for weighting the frequency sub-bands to improve the signal-to-noise ratio of the resultant noise-suppressed channel energy. The noise suppressor preferably includes a noise calculator for calculating background noise values, a speech energy calculator for calculating speech energy values for each channel of the filter bank, and a weighting module for applying calculated weighting values to the projected channel energy to generate the noise-suppressed channel energy.

Abstract: A trellis filtering of a digital signal is disclosed. A method is provided of analysis filtering of an original digital signal including original samples representing physical quantities. Original samples of the digital signal are transformed by successive calculation steps into high and low frequency output samples. Any sample calculated at a given step is calculated by a predetermined function of original samples, and/or previously calculated samples, the samples being ordered in increasing rank. The signal is processed by successive series of samples, the calculations made on any series not taking into account the samples in a following series, and in that the any series terminates in a low-frequency sample.

Abstract: A method and a device by which original digital signals are analysis-filtered, where the original digital signals include original samples representing physical quantities, and where the original samples are transformed by successive calculation steps into high and low frequency output samples. Any sample calculated at a given step is calculated by a predetermined function of the original samples and/or previously calculated samples, where the samples are ordered by increasing rank. The signal is processed by successive input blocks of samples, where the calculations made on an input block under consideration take into account only the original or calculated samples belonging to the input block under consideration, and where the input block under consideration and the following input block overlap over a predetermined number of original samples. Output blocks are formed, where each output block corresponds respectively to an input block.

Abstract: A speech compression system with a special fixed codebook structure and a new search routine is proposed for speech coding. The system is capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech. The codebook structure uses a plurality of subcodebooks. Each subcodebook is designed to fit a specific group of speech signals. A better way is used to calculate a criterion value, minimizing an error signal in a minimization loop as part of the coding system. An external signal sets a maximum bitstream rate for delivering encoded speech into a communications system. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. Each codec is selectively activated to encode and decode the speech signals at different bit rates to enhance overall quality of the synthesized speech at a limited average bit rate.

Abstract: An apparatus and a method for encoding an input signal on the time base through orthogonal transform involves removing the correlation of signal waveform on the basis of the parameters obtained by means of linear predictive coding (LPC) analysis and pitch analysis of the input signal on the time base prior to the orthogonal transform. The time base input signal from input terminal is sent to a normalization circuit section and a LPC analysis circuit. The normalization circuit section removes the correlation of the signal waveform and takes out the residue by an LPC inverse filter and pitch inverse filter and sends the residue to an orthogonal transform circuit section. The LPC parameters from the LPC analysis circuit and the pitch parameters from the pitch analysis circuit are sent to a bit allocation calculation circuit.

Abstract: A central processing unit reads management information corresponding to encoded content information to be reproduced from a storage medium and analyzes the management information, thereby creating read control information. Then, the central processing unit supplies the read control information to a digital signal processing unit via a bus. According to the read control information supplied from the central processing unit, the digital signal processing unit accesses a memory directly via the memory interface and reads out the encoded content information. Then, the digital signal processing unit decodes the read-out encoded content information and outputs the decoded information.

Abstract: The invention provides a perceptual audio signal compression system and method. One aspect of the invention described herein includes transforming the signal into a plurality of bi-orthogonal modified discrete cosine transform (BMDCT) frequency coefficients using a bi-orthogonal modified discrete cosine transform; quantizing the BMDCT frequency coefficients to produce a set of integer numbers which represent the BMDCT frequency coefficients; and encoding the set of integer numbers to lower the number of bits required to represent the BMDCT frequency coefficients.

Abstract: Efficiently encoded signals wherein the compression ration can be high without a corresponding degradation of signal quality. The range of values of subband-based signal components for quantization to zero, to which a short codelength is given, is set so as to be larger. Consequently the ratio of the subband-based signal components quantized to values of a shorter codelength will become higher than if the subband-based signal components are quantized in such a manner as to minimize the total quantization error energy in each frequency band.

Abstract: A method for estimating a pitch frequency of an audio signal includes computing a first transform of the signal to a frequency domain over a first time interval, and computing a second transform of the signal to the frequency domain over a second time interval, which contains the first time interval. A line spectrum of the signal is found, based on the first and second transforms, the spectrum including spectral lines having respective line amplitudes and line frequencies. A utility function that is periodic in the frequencies of the lines in the spectrum is then computed. This function is indicative, for each candidate pitch frequency in a given pitch frequency range, of a compatibility of the spectrum with the candidate pitch frequency. The pitch frequency of the speech signal is estimated responsive to the utility function.

Abstract: An encoder which encodes a voice in accordance with LD-CELP (Low-Delay Code Excited Linear Prediction) of the ITU-T Recommendation G.728. When a vibration wave is encoded by vector quantization, the code is secretly combined with other data. The encoder stores dividing key data kidx by which 128 types of representative vector data (waveform codes) yj; j=0, 1, . . . , 127 are labeled with 0 or 1 in order from the uppermost bit. If the bit is “0”, the vectors are quantized by using only the waveform codes yj corresponding to the bit “0” of the dividing key data kidx as the selection objects. If the bit is “1”, the vectors are quantized by using only the waveform codes yj corresponding to the bit “1” of the dividing key data kidx as the selection objects. Thus, the outputted voice code is combined with another datum bit.

Abstract: A homophonic neologisms generator can include a dictionary table (10) having one or more entries including an orthography and an associated pronunciation including one or more phonemes; and a weightings table (14) having one or more entries specifying a cluster including one or more letters, a cluster pronunciation including one or more phonemes, and a weighting for the pronunciation of the cluster. A user interface can receive a word for which neologisms can be generated. A clustering mechanism can divide the pronunciation into a plurality of phonemes having one or more orthographic representations. Each orthographic representation can include one or more graphemes. Orthographic representations of the pronunciation can be ordered according to the associated weightings of the cluster graphemes in the weightings table and the dictionary can be searched to check that a generated well-formed orthography does not exist.

Abstract: A signal processing method and apparatus is disclosed, which is capable of reproducing a coded audio signal by decoding it while shifting its pitch, and reproducing, from an original sound, a sound having a sufficiently higher pitch than the original sound with few operations and less cost for the decoder used in the signal processing apparatus, and an information serving medium for serving a program which implements the signal decoding and pitch shifting. In one embodiment, the method of providing a signal processing method for decoding a coded signal for reading, includes setting a pitch for the coded signal, decoding only a low frequency portion of the coded signal according to the set pitch, and shifting the pitch of the decoded read signal based on the set pitch.

Abstract: A method and apparatus to connect a group of users via a communications network for a conference call. An initializing signal from an initializing user is detected and the initializing user is allowed to select a predefined user group. The user group comprises a list of users and their associated phone numbers that is created by a user using an user interface of a web portal and stored in a web portal database. After the selected user group is retrieved the availability of each user listed in the user group, for a conference call, is determined by connecting to each user via a communications network, announcing the initializing user to the connected user, and requesting a response from the connected user to determine if the connected user is available for the conference call. The available and connected users are interconnected with one another to allow the users to communicate in a conference call.

Abstract: The invention comprises an efficient system and method for performing the modified discrete cosine transform (MDCT) in support of time-domain aliasing cancellation (TDAC) perceptive encoding compression of digital audio. In one embodiment, an AC-3 encoder performs a required time-domain to frequency-domain transformation via a MDCT. The AC-3 specification presents a non-optimized equation for calculating the MDCT. In one embodiment of the present invention, an MDCT transformer is utilized which produces the same results as carrying out the calculations directly as in the AC-3 equation, but requires substantially lower computational resources. Because the TDAC scheme requires MDCT calculations on differing block sizes, called the long and short blocks, one embodiment of the present invention utilizes complex-valued premultiplication and postmultiplication steps which prepare and arrange the data samples so that both the long and short block transforms may be computed with a computationally efficient FFT.

Abstract: A speech encoding comb codebook structure for providing good quality reproduced low bit-rate speech signals in a speech encoding system. The codebook structure requires minimal training, if any, and allows for reduced complexity and memory requirements. The codebook includes a first and at least one additional sub-codebooks, each having a plurality of code-vectors. The codebook may be randomly populated. All even elements may be set to zero in a first codebook, and all odd elements may be set to zero on a second codebook. The resulting comb codebook includes code-vector combination of the code-vectors from the sub-codebooks. In certain embodiments, the code-vectors of the sub-codebooks may contain zero valued elements. In other embodiments where the code-vectors of the sub-codebooks contain only non-zero elements, zero valued elements may be inserted in between the non-zero elements of the sub-codebooks during the forming of the resultant comb codebook.