Abstract: A method includes receiving data and a plurality of values at a processor. The data can include real-valued data and/or complex data. The plurality of values includes one of a plurality of random values or a plurality of pseudo-random values. The method also includes generating an automorphism, via the processor, based on the plurality of values, and partitioning the data, via the processor, into a plurality of data blocks. The automorphism includes at least one of a linear transformation or an antilinear transformation. Each data block from the plurality of data blocks can have a predefined size. The method also includes applying the automorphism, via the processor, to each data block from plurality of data blocks, to produce a plurality of transformed data blocks, and causing transmission of a signal representing the plurality of transformed data blocks.

Abstract: A method (1100) and apparatus (100) generate a candidate code-vector to code an information signal. The method can include producing (1110) a weighted target vector from an input signal. The method can include processing (1120) the weighted target vector through an inverse weighting function to create a residual domain target vector. The method can include performing (1130) a first search process on the residual domain target vector to obtain an initial fixed codebook code-vector. The method can include performing (1140) a second search process over a subset of possible codebook code-vectors for a low weighted-domain error to produce a final fixed codebook code-vector. The subset of possible codebook code-vectors can be based on the initial fixed codebook code-vector. The method can include generating (1150) a codeword representative of the final fixed codebook code-vector. The codeword can be for use by a decoder to generate an approximation of the input signal.

Abstract: A method and apparatus provides for frame loss recovery following a loss of a frame in an audio codec. The lost frame is identified. Estimated linear predictive coefficients of a previous transform frame are generated based on a decoded audio of the previous transform frame. An estimated residual of the previous transform frame is generated based on the estimated linear predicative coefficients and the decoded audio. A pitch delay is determined from frame error recovery parameters received with the previous transform frame. An extended residual is generated based on the pitch delay and the estimated residual. A first synthesized signal is generated based on the extended residual and the linear predicative coefficients. A decoded audio output of at least the lost frame is generated based on the first synthesized signal. The frame error recovery parameters are generated by an encoder.

Abstract: A method (300) and apparatus (100) generate a candidate code-vector to code an information signal. The method can include producing (310) a target vector from a received input signal. The method can include constructing (320) a plurality of inverse weighting functions based on the target vector. The method can include evaluating (330) an error value associated with each of the plurality of inverse weighting functions to produce a fixed codebook code-vector. The method can include generating (340) a codeword representative of the fixed codebook code-vector, where the codeword can be used by a decoder to generate an approximation of the input signal.

Abstract: A method and apparatus for searching fixed codebook are provided. The method includes: obtaining a basic codebook which comprises position information of N pulses on M tracks, wherein N and M are positive integers; choosing n pulses as search pulses, wherein the n pulses are parts of the N pulses and n is a positive integer smaller than N; and replacing position information of the n search pulses respectively with other position information on the tracks to obtain a searched codebook; executing the search process for K times, wherein K is a positive integer larger than or equal to 2, at least two or more search pulses are chosen in one of the K search processes , and the chosen search pulses vary in each of the K search processes; and obtaining an optimal codebook from the basic codebook and the searched codebook according to a preset criterion.

Abstract: A method for detecting overflow on an electronic device is described. The method includes determining a linear predictive coding synthesis filter gain. The method further includes determining whether overflow is detected based on the linear predictive coding synthesis filter gain and a fixed codebook gain. The method further includes determining a scaling factor if overflow is detected.

Abstract: There is provided an apparatus and method for encoding a speech signal. The encoding comprises: receiving a plurality of current samples of the speech signals; extrapolating a plurality of look-ahead samples from the current samples; and performing linear prediction analysis using the current samples and the extrapolated look-ahead samples.

Abstract: Methods, systems, and non-transitory computer readable media for estimating speech energy of an encoded bit stream based on coding parameters extracted from the partially-decoded bit stream are disclosed. According to one aspect, a method for estimating speech energy of an encoded bit stream based on coding parameters extracted from the partially-decoded bit stream includes receiving a CELP-encoded bit stream, partially decoding the bit stream, and estimating the speech energy of the bit stream based a set of four or fewer CELP parameters extracted from the partially decoded bit stream.

Abstract: An acoustic shock protection device includes a prediction gain estimator and an audio compressor. The prediction gain estimator is configured to analyze a plurality of linear prediction coefficients of an audio signal and determine a category of the audio signal. The audio compressor is coupled to the prediction gain estimator, and the audio compressor is configured to adjust a signal level of the audio signal according to the category of the audio signal.

Abstract: Systems and methods for adaptively processing speech to improve voice intelligibility are described. These systems and methods can adaptively identify and track formant locations, thereby enabling formants to be emphasized as they change. As a result, these systems and methods can improve near-end intelligibility, even in noisy environments. The systems and methods can be implemented in Voice-over IP (VoIP) applications, telephone and/or video conference applications (including on cellular phones, smart phones, and the like), laptop and tablet communications, and the like. The systems and methods can also enhance non-voiced speech, which can include speech generated without the vocal track, such as transient speech.

Abstract: An audio communication device comprises an input connectable to a narrowband audio signal source. The input can receive a narrowband audio signal having a first bandwidth. An extraction unit is connected to the input and arranged to extract a plurality of narrowband parameters from the narrowband audio signal. An extrapolation unit is connected to receive the plurality of narrowband parameters and arranged to generate a plurality of wideband parameters from the plurality of narrowband parameters. The extrapolation unit comprises one or more adaptive neuro-fuzzy inference system modules. The device further comprises a synthesis unit connected to receive the plurality of wideband parameters and arranged to generate, using the wideband parameters, a synthesized wideband audio signal having a second bandwidth wider than the first bandwidth.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

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 quantizing method is provided that includes quantizing an input signal by selecting one of a first quantization scheme not using an inter-frame prediction and a second quantization scheme using the inter-frame prediction, in consideration of one or more of a prediction mode, a predictive error and a transmission channel state.

Abstract: In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal.

Abstract: According to the present invention, a linear prediction filter coefficient of a current frame is acquired from an input signal using linear prediction, a quantized spectrum candidate vector of the current frame, corresponding to the linear prediction filter coefficient of the current frame, is acquired on the basis of first best information, and the quantized spectrum candidate vector of the current frame and the quantized spectrum vector of the previous frame are interpolated. Accordingly, in contrast to conventional phased optimization techniques, optimum parameters which minimize quantization errors, can be obtained.

Abstract: A method of processing an audio signal is disclosed.

Abstract: An apparatus and method for concealing frame erasure and a voice decoding apparatus and method using the same. The frame erasure concealment apparatus includes: a parameter extraction unit determining whether there is an erased frame in a voice packet, and extracting an excitement signal parameter and a line spectrum pair parameter of a previous good frame; and an erasure frame concealment unit, if there is an erased frame, restoring the excitement signal and line spectrum pair parameter of the erased frame by using a regression analysis from the excitement signal and line spectrum pair parameter of the previous good frame. According to the method and apparatus, by predicting and restoring the parameter of the erased frame through the regression analysis, the quality of the restored voice signal can be enhanced and the algorithm can be simplified.

Abstract: An audio decoding apparatus comprises: a plurality of decoding units; a band replicating unit which processes a decoded signal obtained when a corresponding decoding unit decodes a coded signal, according to a scheme specified by transmitted information; and an information transmitting unit which transmits, to a signal processing unit, information identifying the corresponding decoding unit from among the plurality of decoding units.

Abstract: The present invention provides a new recursive FIR filter scheme which supports a variable order short-term predictor, and uses a pipeline stall based on the radix-2 algorithm and an autocorrelation processing time for reducing the complexity of MPEG-4 ALS hardware implementation.

Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.

Abstract: A linear prediction coefficient of a signal represented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. After the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is transformed. This reduces the occurrence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a band extension technique in the frequency domain represented by SBR.

Abstract: Packets of real-time information are sent with a source rate greater than zero kilobits per second, and a time or path or combined time/path diversity rate initially being zero kilobits per second. This results in a quality of service QoS, optionally measured at the sender or the receiver. When the QoS is on an unacceptable side of a threshold of acceptability, the sender sends diversity packets at an increased rate. Increasing the diversity rate while either reducing or maintaining the overall transmission rate is new. CELP-based multiple-description data partitioning sends the base or important information plus a subset of fixed excitation in one packet and sends the base or important information plus the complementary subset of fixed excitation in another packet. Reconstruction produces acceptable quality when only one of the two packets is received and better quality when both packets are received. Reconstruction provides for single and multiple lost packets.

Abstract: A method and apparatus multiplies a past sample a time lag ? older than a current sample by a quantized multiplier ?? on a frame by frame basis, subtracts the multiplication result from the current sample, codes the subtraction result, and codes the time lag using a fixed-length coder if the multiplier ?? is smaller than 0.2 or if information about the previous frame is unavailable, or codes the time lag using a variable-length coder if ?? is not smaller than 0.2. A multiplier ? is coded by a multiplier coder and the multiplier ?? obtained by decoding the multiplier ? is outputted. The process is performed for each frame.

Abstract: Certain aspects relate to providing an at least one audio source to at least one user. Certain aspects relate to selectively modifying an at least one first sound source to be provided to the at least one user, wherein the at least one first sound source is combined with an at least one second sound source, and wherein the selectively modifying is performed relative to the at least one audio source based at least in part on at least some specific information of the at least one first sound source. Other aspects relate to selectively modifying the at least one first sound source to be provided to the at least one user relative to the at least one second sound source based at least in part on at least some specific information of the at least one first sound source.

Abstract: An output circuit for audio codec chip includes a noise eliminating circuit electrically coupled to the audio codec chip for eliminating noise signals. The noise eliminating circuit includes a first switch and a second switch. When the audio codec chip output signals jump from low voltage level to high voltage level, the noise signals are grounded via the first and second switches respectively.

Abstract: A code excited linear prediction type speech coder, which includes a seed storage that stores seeds used as an initial state of oscillation, and an oscillator that generates different vector sequences in accordance with values of the seeds stored in the seed storage and outputs the vector sequences as excitation vectors. The speech coder also includes a linear predictive coding synthesis filter that receives, as input, the excitation vectors, which are the vector sequences generated in accordance with the values of the seeds, that synthesizes the excitation vectors, and that outputs a synthesized speech.

Abstract: A method of generating a frame of audio data for an audio signal from preceding audio data for the audio signal that precede the frame of audio data, the method comprising the steps of: predicting a predetermined number of data samples for the frame of audio data based on the preceding audio data, to form predicted data samples; identifying a section of the preceding audio data for use in generating the frame of audio data; and forming the audio data of the frame of audio data as a repetition (602) of at least part of the identified section to span the frame of audio data, wherein the beginning of the frame of audio data comprises a combination of a subset of the repetition (602) of the at least part of the identified section and the predicted data samples.

Abstract: The present invention relates to encoding technology. The encoding method includes selecting a second encoding mode for encoding an input frame signal according to an analysis on signal characteristic of the input frame signal; obtaining coding demand values for a preset first encoding mode and the second encoding mode which are used to encode the input frame signal; determining, from the above encoding modes based on the coding demand values, an encoding mode for encoding the input frame signal; and multiplexing information of the determined encoding mode and encoded data which are encoded according to the determined encoding mode. Hence, the compatibility and the prioritization in terms of the encoding modes can be achieved.

Abstract: Provided is an encoding device which improves the sound quality of a stereo signal while maintaining a low bit rate. The encoding device includes: an LP inverse filter (121) which LP-inverse-filterS a left signal L(n) by using an inverse quantization linear prediction coefficient AdM(z) of a monaural signal; a T/F conversion unit (122) which converts the left sound source signal Le(n) from a temporal region to a frequency region; an inverse quantizer (123) which inverse-quantizes encoded information Mqe; spectrum division units (124, 125) which divide a high-frequency component of the sound source signal Mde(f) and the left signal Le(f) into a plurality of bands; and scale factor calculation units (126, 127) which calculate scale factors ai and ssi by using a monaural sound source signal Mdeh,i(f), a left sound source signal Leh,i(f), Mdeh,i(f), and right sound source signal Reh,i(f) of each divided band.

Abstract: A method for processing an audio signal, comprising: receiving the audio signal; and processing the received audio signal, wherein the audio signal is processed according to a scheme comprising: comparing a size information of at least two blocks of A+1 level with a size information of a block of A level corresponding to the at least two of A+1 level; and, determining the at least two blocks of A+1 level as an optimum block if the size information of the at least two blocks of A+1 level is less than the size information of the block of A level is disclosed.

Abstract: A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.

Abstract: A processed representation of an audio signal having a sequence of frames is generated by sampling the audio signal within first and second frames of the sequence of frames, the second frame following the first frame, the sampling using information on a pitch contour of the first and second frames to derive a first sampled representation. The audio signal is sampled within the second and third frames, the third frame following the second frame in the sequence of frames. The sampling uses the information on the pitch contour of the second frame and information on a pitch contour of the third frame to derive a second sampled representation. A first scaling window is derived for the first sampled representation, and a second scaling window is derived for the second sampled representation, the scaling windows depending on the samplings applied to derive the first sampled representations or the second sampled representation.

Abstract: A method, system and program for encoding and decoding speech according to a source-filter model whereby speech is modelled to comprise a source signal filtered by a time-varying filter. The method comprises: receiving a speech signal comprising successive frames, for each of a plurality of frames of the speech signal, deriving a first line spectral frequency vector for a first portion of the frame, and a second line spectral frequency vector for a second portion of the frame, and determining a transmit line spectral frequency vector and an interpolation factor based on the first and second line spectral frequency vectors, and on the transmit line spectral frequency vector for a preceding one of the frames.

Abstract: A method of encoding speech, the method comprising: receiving a signal representative of speech to be encoded; at each of a plurality of intervals during the encoding, determining a pitch lag between portions of the signal having a degree of repetition; selecting for a set of said intervals a pitch lag vector from a pitch lag codebook of such vectors, each pitch lag vector comprising a set of offsets corresponding to the offset between the pitch lag determined for each said interval and an average pitch lag for said set of intervals, and transmitting an indication of the selected vector and said average over a transmission medium as part of the encoded signal representative of said speech.

Abstract: An address generator for searching an algebraic codebook is disclosed. The address generator includes: a multiplier multiplying the dimension and a width value of a correlation matrix; a first adder adding a length value and an offset address of the correlation matrix; and a second adder adding the results of the multiplier and the first adder to generate an address for algebraic codebook searching. The amount of calculation required for an address calculation to search an algebraic codebook can be reduced.

Abstract: An information coding apparatus includes a predictive signal generator that generates a predictive signal; a predictive residual signal generator that generates a predictive residual signal; a quantizer that quantizes a quantization input signal generated based on the predictive residual signal; a quantization error signal generator that generates a quantization error signal; a feedback signal generator that generates a feedback signal for controlling the frequency characteristic of the quantization noise after decoding based on the quantization error signal; and a quantization input signal generator that generates the quantization input signal.

Abstract: Packet loss concealment systems and methods are described that may be used in conjunction with a Bluetooth® Low-Complexity Sub-band Coding (LC-SBC) codec or other sub-band codecs, including but not limited to an MPEG-1 Audio Layer 3 (MP3) codec, an Advanced Audio Coding (AAC) codec, and a Dolby AC-3 codec.

Abstract: Provided is a stereo audio encoding device which can improve the ICP (Inter-channel Prediction) performance of a stereo audio signal while suppressing the bit rate. The device (100) includes: a QMF analysis unit (101) which divides two channel signals constituting a stereo audio signal into a plurality of frequency band signals; a monaural signal generation unit (104) which generates a monaural signal by averaging the two channel signals of the divided frequency bands; parameter band constituting units (102, 105) each of which collects one or more of the continuous frequency bands to constitute a parameter band in such a manner that less bands are contained in a lower frequency for the two channel signals and monaural signals of the divided frequency bands; and an ICP analysis unit (106) which performs inter-channel prediction by using the channel signal and the monaural signal of the divided frequency bands.

Abstract: An apparatus and method for encoding/decoding a speech signal which determines a variable bit rate based on reserved bits obtained from a target bit rate, is provided. The variable bit rate is determined based on a source feature of the speech signal and the reserved bits is obtained based on the target bit rate. The apparatus for encoding the speech signal may include a linear predictive (LP) analysis unit/quantization unit to determine an immittance spectral frequencies (ISF) index, a closed loop pitch search unit, a fixed codebook search unit, a gain vector quantization (VQ) unit to determine a gain vector quantization (VQ) index, and a bit rate control unit to control at least two indexes of the ISF index, the pitch index, the code index, and the gain VQ index to be encoded to be variable bit rates based on a source feature of a speech signal and the reserved bits.

Abstract: The system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, a transmitter analysis filter for receiving the digital speech signal and dividing it into a baseband signal and an enhancement residual band signal, a standard baseband encoder for accepting the baseband signal and coding it using an ITU-T encoder, an additional baseband encoder for reducing standard coding distortion in the baseband signal, an enhancement residual band encoder for coding a signal obtained by removing the coded baseband signal from the original digital speech signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

Abstract: Disclosed is an adaptive sound source vector quantization device capable of improving quantization accuracy of adaptive sound source vector quantization while suppressing increase of the calculation amount in CELP sound encoding which performs encoding in sub-frame unit.

Abstract: Aspects of a method and system for dual voice path processing in an audio CODEC may enable selecting two or more signals received via one or more audio input devices, and filtering and down-sampling each of the selected signals via two or more signal processing branches. Furthermore, an output sample rate of each of the signal processing branches may be configured independently. The signal processing branches may each comprise one or more IIR filters with configurable coefficients and one or more cascaded-integrate-comb (CIC) decimation filters having a configurable decimation ratio. A first of the selected signals, may be filtered and/or down-sampled to generate a signal having a first, lower, sample rate, and a second of the signals may be filtered and/or down-sampled to generate a signal having a second, higher, sample rate. One or more post-processing algorithms such as audio beamforming may also be performed on the selected signals.

Abstract: An apparatus for noise suppression having a linear prediction analysis circuit having an LP error filter (LFF), which takes a first, noisy voice signal y(n)=x(n)+?(n) as a basis for producing an LP-error-filter output signal e(n), having a coefficient calculation unit (KBE), which updates the coefficients of the LP error filter on the basis of the internal signals (including the input and out signals y(n) and e(n)) in the LP error filter, and having a subtraction unit, which subtracts the LP error filter output signal e(n) from the first voice signal y(n) in a subtractor and, following the subtraction, outputs the remainder as a second voice signal x(n)=y(n)?e(n) in which the noise is suppressed.

Abstract: Provided is an audio encoding device which performs a closed loop search of a gain and a sound source vector without significantly increasing the calculation amount as compared to an open loop search. In the audio encoding device, firstly, a first parameter decision unit (121) performs a sound source search by an adaptive sound source codebook and then a second parameter decision unit (122) simultaneously performs by a closed loop, the sound source and the gain search by using a fixed sound source codebook. More specifically, for a combination of a fixed sound source vector and gain, the sum of a value obtained by multiplying a candidate fixed sound source vector by a candidate gain and a value obtained by multiplying an adaptive sound source vector by a candidate gain is subjected to a combination filter formed by a filter coefficient based on a quantization linear prediction coefficient so as to generate a combined signal.

Abstract: A method, device and system to implement hiding the loss packet are provided. The provided method, device and system recover the lost frame according to the data before and after the lost frame and enhances the correlation of the recovered lost frame data and the data after the lost frame. A method and device for estimating pitch period are also provided which select a pitch period from the initial pitch period and the pitch periods corresponding to the frequencies which are one or more times higher than the frequencies corresponding to the initial pitch period as the final estimated pitch period, may improve frequency multiplication when estimating the pitch period; in addition, by tuning of the pitch period by matching the waves, the error of estimating pitch period may be reduced and the quality of the audio data may be improved.

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, device and system to implement hiding the loss packet are provided. The provided method, device and system recover the lost frame according to the data before and after the lost frame and enhances the correlation of the recovered lost frame data and the data after the lost frame. A method and device for estimating pitch period are also provided which select a pitch period from the initial pitch period and the pitch periods corresponding to the frequencies which are one or more times higher than the frequencies corresponding to the initial pitch period as the final estimated pitch period, may improve frequency multiplication when estimating the pitch period; in addition, by tuning of the pitch period by matching the waves, the error of estimating pitch period may be reduced and the quality of the audio data may be improved.

Abstract: A device and a method for quantizing a LPC filter in the form of an input vector in a quantization domain, comprises a calculator of a first-stage approximation of the input vector, a subtractor of the first-stage approximation from the input vector to produce a residual vector, a calculator of a weighting function from the first-stage approximation, a warper of the residual vector with the weighting function, and a quantizer of the weighted residual vector to supply a quantized weighted residual vector.

Abstract: It is an object to disclose a voice coding device, etc. in which the deterioration of a voice quality of a decoded signal can be reduced in the case that low frequency domain components of a spectrum are used for coding high frequency domain components and that no low frequency domain components exist.