Abstract: The invention discloses a method including: performing in a unit of first timeframe frame length, framing on a continuous voice sample to obtain a plurality of first timeframes, detecting energy of each of the first timeframes, and determining a target first timeframe including a potential abrupt exception of a voice signal by analyzing a relationship between the energy of the plurality of first timeframes; performing, in a unit of second timeframe frame length, framing on the continuous voice sample to obtain a plurality of second timeframes, and processing each of the second timeframes to acquire a tone feature, and determining, by analyzing a tone feature of at least one of the second timeframes including at least one target second timeframe, whether the potential abrupt exception of a voice signal included in the target first timeframe included in the target second timeframe is a real abrupt exception of a voice signal.

Abstract: An ultrasonic diagnosing apparatus includes: a transducer array 1 composed of arrayed transducer elements T1 to T6 for transmitting ultrasound; driving circuits D1A to D6A each provided for transmission channels for driving each of the transducer elements; a transmission trigger generator 2 for generating a trigger pulse for controlling each of the driving circuits; a parallel reception beam former 3 for processing reception signals from the transducer elements; a signal processor 4 for processing an output signal of the parallel reception beam former; and a control unit 5 for controlling the transmission trigger generator, the parallel reception beam former and the signal processor.

Abstract: An encryption device that, when voice or image data or the like being encoded is encrypted using a one-time pad (OTP) cipher and then transmitted, reduces a period of time in which a cipher key for the OTP cipher runs out. A first terminal device determines whether to encode transmission data by a first encoding scheme or a second encoding scheme having a lower bit rate than the first encoding scheme, depending on the number of remaining bits of an OTP cipher key, and encodes the transmission data according to the determined encoding scheme, thereby generating encoded data. The first terminal device encrypts the generated encoded data with the OTP cipher using the OTP cipher key, thereby generating encrypted communication data, and transmits the generated encrypted communication data to a second terminal device.

Abstract: A recognition dictionary creating device includes a user dictionary in which a phoneme label string of an inputted voice is registered and an interlanguage acoustic data mapping table in which a correspondence between phoneme labels in different languages is defined, and refers to the interlanguage acoustic data mapping table to convert the phoneme label string registered in the user dictionary and expressed in a language set at the time of creating the user dictionary into a phoneme label string expressed in another language which the recognition dictionary creating device has switched.

Abstract: A method in a mobile communication terminal includes holding a definition of a sub-sampled codebook identifying precoding matrices to be used for providing precoding feedback by the terminal. The precoding matrices in the sub-sampled codebook are selected from a master codebook that is made-up of a long-term sub-codebook and a short-term sub-codebook. The definition defines a first subset of the long-term sub-codebook and a second subset of the short-term sub-codebook. A Multiple-Input Multiple-Output (MIMO) signal is received in the terminal via multiple receive antennas. Based on the received MIMO signal, a precoding matrix is selected from the sub-sampled codebook for precoding subsequent MIMO signals transmitted to the terminal. The precoding feedback indicating the selected precoding matrix is calculated.

Abstract: A parametric background noise estimate is continuously updated during an active or non-silence phase so that the noise generation may immediately be started with upon the entrance of an inactive phase following the active phase. In accordance with another aspect, a spectral domain is very efficiently used in order to parameterize the background noise thereby yielding a background noise synthesis which is more realistic and thus leads to a more transparent active to inactive phase switching.

Abstract: Disclosed are an audio encoding device and an audio decoding device which reduce degradation of subjective quality of a decoding signal caused by power mismatch of a decoding signal which is generated by a concealing process upon disappearance of a frame. When a frame is lost, a past encoding parameter is used to obtain a concealed LPC of the current frame and a concealed sound source parameter. A normal CELP decoding is performed from the obtained concealed sound source parameter. Correction is performed by using a conceal parameter on the obtained concealed LPC and the concealed sound source signal. The power of the corrected concealed sound source signal is adjusted to match a reference sound source power. A filter gain of the synthesis filter is adjusted so as to adjust the power of a decoded sound signal to the power of a decoded sound signal during an error-free state.

Abstract: A method and system for compressing an audio signal. The method includes receiving a segment of an audio signal and selectively disabling noise suppression for the received segment. The segment is filtered in a noise-suppression module if noise suppression is not disabled. The method also includes calculating an autocorrelation coefficient and an LSP coefficient, predicting a short-term coefficient and long-term coefficients according to the LSP coefficient and calculating one or more bandwidth-expanded correlation coefficients. Further, the method includes determining the type of packet in which to encode the segment. An encoding rate is selected from among a full rate encode, a half-rate encode, and an eight-rate encode if noise suppression is not disabled. An encoding rate is selected from among a full rate encode and a half-rate encode if noise suppression is disabled. Furthermore, the segment is formed into a packet of the determined type and selected rate.

Abstract: In lossy data compression of a signal using ADPCM, an adaptive decorrelation or “prediction” filter is used to reduce the amplitude of the signal, the spectral dynamic range of the signal also being reduced. This latter reduction is effected in a nonuniform manner, if known techniques are used, with regions of high spectral density being compressed more than regions of low spectral density. The present invention recognises that using a uniform compression ratio results in a better tradeoff between compression and robustness to transmission channel errors. A method is described for obtaining a uniform compression ratio by adjusting coefficients of the decorrelation filter in dependence on coefficients of an adaptive training filter that is fed from the output of the decorrelation filter. A reverse method is also provided along with encoder, decoder and codec implementing the techniques.

Abstract: A converter and conversion method are disclosed for converting N channel audio input channels into M channel audio output channels, wherein a processor is used for applying a transfer function to a signal received on an input channel to obtain reverberation components of a calculated output channel, wherein said transfer function is a simplified transfer function matching a selected subset of a set of local maxima of a measured reverberation when applied to a corresponding stimulus.

Abstract: An apparatus including at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to select at least two single frequency components; generate an indicator, the indicator being configured to represent the at least two single frequency components and is configured to be dependent on the frequency separation between the two single frequency components.

Abstract: A method for concatenating a first frame of samples and a subsequent second frame of samples, the method comprising applying a phase filter adapted to minimizing a discontinuity at a boundary between the first and second frames of samples.

Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: An apparatus for encoding an audio signal having a stream of audio samples has: a windower for applying a prediction coding analysis window to the stream of audio samples to obtain windowed data for a prediction analysis and for applying a transform coding analysis window to the stream of audio samples to obtain windowed data for a transform analysis, wherein the transform coding analysis window is associated with audio samples within a current frame of audio samples and with audio samples of a predefined portion of a future frame of audio samples being a transform-coding look-ahead portion, wherein the prediction coding analysis window is associated with at least the portion of the audio samples of the current frame and with audio samples of a predefined portion of the future frame being a prediction coding look-ahead portion, wherein the transform coding look-ahead portion and the prediction coding look-ahead portion are identically to each other or are different from each other by less than 20%; and an enc

Abstract: A method of noise-robust speech classification is disclosed. Classification parameters are input to a speech classifier from external components. Internal classification parameters are generated in the speech classifier from at least one of the input parameters. A Normalized Auto-correlation Coefficient Function threshold is set. A parameter analyzer is selected according to a signal environment. A speech mode classification is determined based on a noise estimate of multiple frames of input speech.

Abstract: A coding method, a decoding method, a coder, and a decoder are disclosed herein. A coding method includes: obtaining the pulse distribution, on a track, of the pulses to be encoded on the track; determining a distribution identifier for identifying the pulse distribution according to the pulse distribution; and generating a coding index that includes the distribution identifier. A decoding method includes: receiving a coding index; obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track; determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier; and reconstructing the pulse order on the track according to the pulse distribution.

Abstract: The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal.

Abstract: A multi-carrier signal is typically comprised of many equidistant sub-carriers. This results in periodicity of spectrum within the bandwidth of such a multi-carrier signal. An unknown multi-carrier signal with equidistant sub-carriers can thus be sensed together with its sub-carrier spacing by finding a discernable local maximum in the cepstrum (Fourier transform of the log spectrum) of the multi-carrier signal.

Abstract: A multi-carrier signal is typically comprised of many equidistant sub-carriers. This results in periodicity of spectrum within the bandwidth of such a multi-carrier signal. An unknown multi-carrier signal with equidistant sub-carriers can thus be sensed together with its sub-carrier spacing by finding a discernible local maximum in the cepstrum (Fourier transform of the log spectrum) of the multi-carrier signal.

Abstract: A wave resynthesis method and system comprises receiving input wave form, processing received data to create an enhanced wave form, identifying the enhanced wave form, transmitting the identified wave form to a receiving unit, identifying the received wave form, resynthesizing the received wave form and outputting the resynthesized wave form. Identifying the enhanced wave form includes sampling the waveform and measuring the angle of the samples at two or more points in the waveform. The enhancing of voice audio input includes the parallel processing the input audio by a module that is a low pass filter with dynamic offset, an envelope controlled band-pass filter, a high pass filter and adding an amount of dynamic synthesized sub bass to the audio. The four processed audio signals are combined in a summing mixer with the original audio. The receiving unit has a complete set of encrypted tables for accurate resynthesizing/reproduction.

Abstract: A method for coding and decoding an audio signal or speech signal and an apparatus adopting the method are provided.

Abstract: An encoding apparatus includes a first layer encoder that encodes an input signal, a first layer decoder that decodes the first layer encoded data, a weighting filter that filters a first layer error signal to acquire a weighted first layer error signal, a first layer error transform coefficient calculator that transforms the weighted first layer error signal into a frequency domain, and a second layer encoder that encodes the first layer error transform coefficient. The second layer encoder includes a first shape vector encoder that refers the first layer error transform coefficient to generate a first shape vector and first shape encoded information. A target gain calculator calculates a target gain using the first layer error transform coefficient and the first shape vector, a gain vector generator generates a gain vector, and a gain vector encoder encodes the gain vector to acquire gain encoded information.

Abstract: An encoding apparatus includes a first layer encoder that encodes a signal, a first layer decoder that decodes first layer encoded data, a first layer error transform coefficient calculator that transforms a first layer error signal into a frequency domain and a second layer encoder that encodes the first layer error transform coefficient to acquire second layer encoded data. The second layer encoder includes a band determiner that determines a band to be encoded by the second layer encoder, and a first shape vector encoder that refers the first layer error transform coefficient included in the band to generate a first shape vector and first shape encoded information, a target gain calculator calculates target gain per subband, a gain vector generator generates a gain vector using a plurality of target gains, and a gain vector encoder encodes the gain vector to acquire gain encoded information.

Abstract: By monitoring the wind noise in a location in which a cellular telephone is operating and by applying noise reduction and/or cancellation protocols at the appropriate time via analog and/or digital signal processing, it is possible to significantly reduce wind noise entering into a communication system.

Abstract: A method for encoding and decoding a digital audio signal is provided, said method comprising the steps of: encoding a first sequence of samples of the digital signal according to a transform encoding; encoding a second sequence of samples of the digital signal according to a predictive encoding; wherein the second sequence starts before the end of the first sequence, a subsequence common to the first and second sequences being thus encoded both by predictive encoding and by transform encoding.

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

Abstract: An electronic device for determining a set of pitch cycle energy parameters is described. The electronic device includes a processor and executable instructions stored in memory. The electronic device obtains a frame, a set of filter coefficients and a residual signal based on the frame and the set of filter coefficients. The electronic device determines a set of peak locations based on the residual signal and segments the residual signal such that each segment includes one peak. The electronic device determines a first set of pitch cycle energy parameters based on a frame region between two consecutive peak locations and maps regions between peaks in the residual signal to regions between peaks in a synthesized excitation signal to produce a mapping. The electronic device determines a second set of pitch cycle energy parameters based on the first set of pitch cycle energy parameters and the mapping.

Abstract: A framing method and apparatus are disclosed to overcome inconsistency of gains between sub-frames caused by simple average framing in the prior art. The method includes: obtaining the Linear Prediction Coding (LPC) order and the pitch of the signal; removing the samples inapplicable to Long-Term Prediction (LTP) synthesis according to the LPC prediction order and the pitch; and splitting the remaining samples of the signal into several sub-frames. The technical solution under the present invention is applicable to the multimedia speech coding field.

Abstract: Disclosed is an apparatus for playing and producing realistic object audio. The apparatus for playing realistic object audio includes: a deformatter unit individually separating scene description (SD) compression data and object audio compression data from inputted audio files; an SD decoding unit decoding the SD compression data to restore SD information; an object audio decoding unit decoding the object audio compression data to restore object audio signals which are respective audio signals of a plurality of objects; and an object audio effect unit adding an audio effect for each object to the object audio signals according to SD information for each object corresponding to the object audio signals among the SD information to produce a realistic object audio signal corresponding to each of the object audio signals.

Abstract: Provided is a system, method, and computer program product for improving the quality of speech reproduction in wireless applications where the received speech frames are subject to transmission and packet losses. The speech decoding process is dynamically delayed by at least one frame period in order to perform additional error correction and concealment techniques during times when the wireless link quality if below a predetermined threshold. The wireless link is monitored and if the link quality falls below a predetermined threshold, the decoding process is delayed by at least one frame period so that one or more error correcting techniques can be performed to increase the quality of the reconstructed speech.

Abstract: Codebook Arrangement for use in coding an input sound signal includes First and Second Codebook Stages. First Codebook Stage includes one of a time-domain CELP codebook and a transform-domain codebook. Second Codebook Stage follows the first codebook stage and includes the other of the time-domain CELP codebook and the transform-domain codebook. Codebook Stage includes an adaptive codebook may be provided before First Codebook Stage. A selector may be provided to select an order of the time-domain CELP codebook and the transform-domain codebook in First and Second Codebook Stages, respectively, as a function of characteristics of the input sound signal. The selector may also be responsive to both the characteristics of the input sound signal and a bit rate of the codec using Codebook Arrangement to bypass Second Codebook Stage. Codebook Arrangement can be used in a coder of an input sound signal.

Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: The invention relates to a method for speech signal analysis, modification and synthesis comprising a phase for the location of analysis windows by means of an iterative process for the determination of the phase of the first sinusoidal component and comparison between the phase value of said component and a predetermined value, a phase for the selection of analysis frames corresponding to an allophone and readjustment of the duration and the fundamental frequency according to certain thresholds and a phase for the generation of synthetic speech from synthesis frames taking the information of the closest analysis frame as spectral information of the synthesis frame and taking as many synthesis frames as periods that the synthetic signal has. The method allows a coherent location of the analysis windows within the periods of the signal and the exact generation of the synthesis instants in a manner synchronous with the fundamental period.

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

Abstract: In an embodiment, a method of receiving a digital audio signal, using a processor, includes generating a high band time domain signal; generating low band time domain signal; estimating an energy ratio between the high band and the low band from a last good frame; keeping the energy ratio for following frame-erased frames by applying an energy correction scaling gain to a high band signal segment by segment in the time domain; and combining the low band signal and the high band signal into a final output.

Abstract: An encoding method includes: extracting core layer characteristic parameters and enhancement layer characteristic parameters of a background noise signal, encoding the core layer characteristic parameters and enhancement layer characteristic parameters to obtain a core layer codestream and an enhancement layer codestream. The disclosure also provides an encoding device, a decoding device and method, an encapsulating method, a reconstructing method, an encoding-decoding system and an encoding-decoding method. By describing the background noise signal with the enhancement layer characteristic parameters, the background noise signal can be processed by using more accurate encoding and decoding method, so as to improve the quality of encoding and decoding the background noise signal.

Abstract: A sound encoder for efficiently encoding stereophonic sound. A prediction parameter analyzer determines a delay difference D and an amplitude ratio g of a first-channel sound signal with respect to a second-channel sound signal as channel-to-channel prediction parameters from a first-channel decoded signal and a second-channel sound signal. A prediction parameter quantizer quantizes the prediction parameters, and a signal predictor predicts a second-channel signal using the first decoded signal and the quantization prediction parameters. The prediction parameter quantizer encodes and quantizes the prediction parameters (the delay difference D and the amplitude ratio g) using a relationship (correlation) between the delay difference D and the amplitude ratio g attributed to a spatial characteristic (e.g., distance) from a sound source of the signal to a receiving point.

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

Abstract: A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.

Abstract: Disclosed herein are systems, computer-implemented methods, and tangible computer-readable media for handling missing speech data. The computer-implemented method includes receiving speech with a missing segment, generating a plurality of hypotheses for the missing segment, identifying a best hypothesis for the missing segment, and recognizing the received speech by inserting the identified best hypothesis for the missing segment. In another method embodiment, the final step is replaced with synthesizing the received speech by inserting the identified best hypothesis for the missing segment. In one aspect, the method further includes identifying a duration for the missing segment and generating the plurality of hypotheses of the identified duration for the missing segment. The step of identifying the best hypothesis for the missing segment can be based on speech context, a pronouncing lexicon, and/or a language model. Each hypothesis can have an identical acoustic score.

Abstract: A method for overlap-adding signals useful for performing frame loss concealment (FLC) in an audio decoder as well as in other applications. The method uses a dynamic mix of windows to overlap two signals whose normalized cross-correlation may vary from zero to one. If the overlapping signals are decomposed into a correlated component and an uncorrelated component, they are overlap-added separately using the appropriate window, and then added together. If the overlapping signals are not decomposed, a weighted mix of windows is used. The mix is determined by a measure estimating the amount of cross-correlation between overlapping signals, or the relative amount of correlated to uncorrelated signals.

Abstract: A method and system for analysis-by-synthesis encoding of an information signal is provide. The encoder (400) can include the steps of generating a first synthetic signal based on a first pitch-related codebook (402), generating a second synthetic signal based on a second pitch-related codebook (404), selecting a codebook configuration parameter based on the reference signal and the first and second synthetic signals, and conveying the codebook configuration for use in reconstructing an estimate of the input signal. The encoder can include an error expression having an error bias (506) and a prediction gain having a prediction gain bias (508) for determining the codebook configuration. The encoder can employ variable length coding and combinatorial subframe coding (600) for efficiently compressing the codebook configuration parameter and codebook related parameters for one or more subframes.

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

Abstract: An image reproduction system comprising: an image data storage unit that stores a plurality of image data, an image display unit that successively displays a plurality of images based on the plurality of image data, an audio input unit that receives as input sound of a surrounding environment, an audio determination unit that determines whether the sound received as input by the audio input unit is in a predetermined state, and a control unit that controls a display time of the image displayed on the image display unit at the time of determination by the audio determination unit based on the results of determination by the audio determination unit when continuously displaying the plurality of images on the image display unit while successively switching them.

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 a moving-average predictor produces a predicted parameter by multiplying a predictive coefficient with a past quantized prediction residue. 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: Methods and systems for transcoding input audio data in a first encoding format to generate audio data in a second encoding format, and filterbanks for use in such systems. Some such systems include a combined synthesis and analysis filterbank (configured to generate transformed frequency-band coefficients indicative of at least one sample of the input audio data by transforming frequency-band coefficients in a manner equivalent to upsampling the frequency-band coefficients and filtering the resulting up-sampled values to generate the transformed frequency-band coefficients, where the frequency-band coefficients are partially decoded versions of input audio data that are indicative of the at least one sample) and a processing subsystem configured to generate transcoded audio data in the second encoding format in response to the transformed frequency-band coefficients.

Abstract: Audio signal bandwidth expansion is performed on a narrow bandwidth signal received from a far end source. The far end source may transmit the signal over the audio communication network. The narrow band signal bandwidth is expanded such that the bandwidth exceeds that of the audio communication network. The signal may be expanded by performing frequency folding on the signal. One or more features are determined for the narrow bandwidth signal, and the expanded signal is modified based on a feature. The feature may be signal band energy slope, narrow band signal energy, or some other feature. The modification may be performed by a shelf filter selected based on the feature. The modified signals are provided for additional processing. In some embodiments, a noise component is added to the narrow band signal prior to folding to create an excitation that reduces the appearance of a fully harmonic signal characteristic.

Abstract: A speech processing system includes a plurality of signal analyzers that extract salient signal attributes of an input voice signal. A difference module computes the differences in the salient signal attributes. One or more control modules control a plurality of speech generators using an output signal from the difference module in a speech-locked loop (SLL), the speech generators use the output signal to generate a voice signal.

Abstract: Disclosed is a system and method for implementing compression coding of audio signals, such as speech signals, using two long-term prediction (LTP) models. The method determines the parameters of a second long-term prediction model on the basis of the parameters of at least one first LTP model. The present invention is aimed at switching from an LTP model with a single coefficient (monotap) to an LTP model with several coefficients, (multitap) and vice versa, as well as at switching between two multitap LTP models. The complexity of the method may be adjusted, especially as a function of a desired compromise between a target complexity and a desired quality. A device for implementing the method according to the invention is, moreover, very useful for multiple codings in cascade (transcodings) or in parallel (multi-codings and multi-mode codings).

Abstract: Provided are an encoding apparatus and a decoding apparatus of a multi-channel signal. The encoding apparatus of the multi-channel signal may process a phase parameter associated with phase information between a plurality of channels constituting the multi-channel signal, based on a characteristic of the multi-channel signal. The encoding apparatus may generate an encoded bitstream with respect to the multi-channel signal using the processed phase parameter and a mono signal extracted from the multi-channel signal.