Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes a linear-prediction-domain decoder configured to provide a first decoded audio information on the basis of an audio frame encoded in a linear prediction domain, a frequency domain decoder configured to provide a second decoded audio information on the basis of an audio frame encoded in a frequency domain, and a transition processor. The transition processor is configured to obtain a zero-input-response of a linear predictive filtering, wherein an initial state of the linear predictive filtering is defined depending on the first decoded audio information and the second decoded audio information, and modify the second decoded audio information depending on the zero-input-response, to obtain a smooth transition between the first and the modified second decoded audio information.

Abstract: The present invention provides for methods and apparatuses for processing audio data. In one embodiment, there is a provided a method for achieving bitstream scalability in a multi-channel audio encoder, said method comprising receiving audio input data; organizing said input data by a Code Excited Linear Predictor (CELP) processing module for further encoding by arranging said data according to significance of data, where more significant data is placed ahead of less significant data; and providing a scalable output bitstream. The organized CELP data comprises of a first part and a second part. The first part comprises a frame header, sub frame parameters and innovation vector quantization data from the first frame from all channels. The innovation vector quantization data from the first frames from all channels is arranged according to channel number.

Abstract: The present disclosure provides systems and methods for dynamically signaling encoder capabilities of vocoders of corresponding communication nodes. In one embodiment, during a call between a first communication node and a second communication node, a control node (e.g., base station controller or mobile switching center) for the first communication node sends capability information for a voice encoder of a vocoder of the first communication node to a control node for the second communication node. As a result, the second communication node is enabled to select and request a preferred encoder mode for the voice encoder of the vocoder of the first communication node based on the capabilities of the voice encoder of the vocoder of the first communication node.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal.

Abstract: Provided are a method, apparatus, and medium for encoding/decoding a high frequency band signal by using a low frequency band signal corresponding to an audio signal or a speech signal. Accordingly, since the high frequency band signal is encoded and decoded by using the low frequency band signal, encoding and decoding can be carried out with a small data size while avoiding deterioration of sound quality.

Abstract: Disclosed is an encoding device whereby it is possible to improve the quality of an encoded signal, even when encoding music signals. In the encoding device, a Code-Excited Linear Prediction (CELP) encoder (101) generates first encoded data by encoding an input signal, a CELP decoder (102) generates a decoded signal by decoding the first encoded data input from the CELP encoder (101), and a characteristic parameter encoder (106) calculates a parameter that expresses the degree of fluctuation in the ratio of the peak components and the floor components between the spectra of the decoded signal and the input signal.

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 audio signal decoder includes a transform domain path configured to obtain a time-domain representation of a portion of an audio content on the basis of a first set of spectral coefficients, a representation of an aliasing-cancellation stimulus signal and a plurality of linear-prediction-domain parameters. The transform domain path applies a spectrum shaping to the first set of spectral coefficients to obtain a spectrally-shaped version thereof. The transform domain path obtains a time-domain representation of the audio content on the basis of the spectrally-shaped version of the first set of spectral coefficients. The transform domain path includes an aliasing-cancellation stimulus filter to filter the aliasing-cancellation stimulus signal in dependence on at least a subset of the linear-prediction-domain parameters.

Abstract: An audio signal encoder includes a transform-domain path which obtains spectral coefficients and noise-shaping information on the basis of a portion of the audio content, and which windows a time-domain representation of the audio content and applies a time-domain-to-frequency-domain conversion. The audio signal decoder includes a CELP path to obtain a code-excitation information and a LPD parameter information. A converter applies a predetermined asymmetric analysis window in both if a current portion is followed by a subsequent portion to be encoded in the transform-domain mode or in the CELP mode. Aliasing cancellation information is selectively provided in the later case.

Abstract: An apparatus for processing a signal and method thereof are disclosed. The present invention includes receiving coding mode information indicating a speech coding scheme or an audio coding scheme, linear prediction coding degree information indicating a linear prediction coding degree, and the signal including at least one of a speech signal and an audio signal; decoding the signal according to the speech coding scheme or the audio coding scheme based on the coding mode information; decoding linear prediction coding coefficients of the signal based on the linear prediction coding degree information; and generating an output signal by applying the decoded linear prediction coding coefficients to the decoded signal. In this case, the linear prediction coding degree information is determined based on a variation of a value of an LPC residual generated from performing the linear prediction coding on the signal.

Abstract: A tone determination device, which determines the tonality of an input signal, is capable of reducing calculation complexity. Therein a frequency conversion unit (101) converts the frequency of an input signal; a downsampling unit (102) carries out shortening processing which shortens the vector series length of the frequency-converted signal; a constancy determination unit (107) determines the constancy of the input signal; depending on the constancy of the input signal, a vector selection unit (104) selects either the vector series of the post-frequency conversion signal or the vector series after the shortening of the vector series length; a correlation analysis unit (105) uses the vector series selected by the vector selection unit (104) to obtain correlations; and a tone determination unit (106) uses the correlations to determine the tonality of the input signal.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal.

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: In a CELP coder, a combined innovation codebook coding device comprises a pre-quantizer of a first, adaptive-codebook excitation residual, and a CELP innovation-codebook search module responsive to a second excitation residual produced from the first, adaptive-codebook excitation residual. In a CELP decoder, a combined innovation codebook comprises a de-quantizer of pre-quantized coding parameters into a first excitation contribution, and a CELP innovation-codebook structure responsive to CELP innovation-codebook parameters to produce a second excitation contribution.

Abstract: A method and an apparatus for generating a lattice vector quantizer codebook are disclosed. The method includes: storing an eigenvector set that includes amplitude vectors and/or length vectors, where the amplitude vectors and/or length vectors are different from each other and correspond to a root leader of a lattice vector quantizer; storing storage addresses of the amplitude vectors and length vectors, where the amplitude vectors and length vectors correspond to the root leader and are in the eigenvector set; and generating a lattice vector quantizer codebook according to the eigenvector set and the storage addresses.

Abstract: Methods, encoders, and digital systems are provided for predictive encoding of speech parameters in which an input frame is encoded by quantizing a parameter vector of the input frame with a strongly-predictive codebook and a weakly-predictive codebook to obtain a strongly-predictive distortion and a weakly-predictive distortion, adjusting a correlation indicator based on a relative correlation of the input frame to a previous frame, wherein the correlation indicator is indicative of the strength of the correlation of previously encoded frames, and encoding the input frame with the weakly-predictive codebook unless the correlation indicator has reached a correlation threshold.

Abstract: A fixed codebook searching apparatus, includes a convolution operator, implemented by at least one processor, that convolves an impulse response of a perceptually weighted synthesis filter with an impulse response vector that has values at negative times, to generate a second impulse response vector that has values at negative times. A matrix generator, implemented by at least one processor, generates a Toeplitz-type convolution matrix using the second impulse response vector generated by the convolution operator. A searcher, implemented by at least one processor, performs a codebook search by maximizing a term using the Toeplitz-type convolution matrix.

Abstract: An audio encoding device for correcting a component having insufficient encoding capability in a core layer by an extended layer. A core layer encoder encodes an audio signal. An extended layer encoder encodes an encoding residual of the core layer encoder. A characteristic correction inverse filter arranged at a pre-stage of an LPC synthesis filter subjects the component having insufficient encoding capability in the core layer to an inverse characteristic correction process, and a characteristic correction filter arranged at a post-stage of the LPC synthesis filter performs a process for characteristic correction of the synthesis signal inputted from the LPC synthesis filter.

Abstract: Methods, and corresponding codec-containing devices are provided that have source coding schemes for encoding a component of an excitation. In some cases, the source coding scheme is an enumerative source coding scheme, while in other cases the source coding scheme is an arithmetic source coding scheme. In some cases, the source coding schemes are applied to encode a fixed codebook component of the excitation for a codec employing codebook excited linear prediction, for example an AMR-WB (Adaptive Multi-Rate-Wideband) speech codec.

Abstract: The invention pertains to a method and apparatus of efficient encoding and decoding of vector quantized data. The method and system explores and implements sub-division of a quantization vector space comprising class-leader vectors and representation of the class-leader vectors by a set of class-leader root-vectors facilitating faster encoding and decoding, and reduced storage requirements.

Abstract: Provided is an apparatus for integrally encoding and decoding a speech signal and an audio signal. An encoding apparatus for integrally encoding a speech signal and an audio signal, may include: a module selection unit to analyze a characteristic of an input signal and to select a first encoding module for encoding a first frame of the input signal; a speech encoding unit to encode the input signal according to a selection of the module selection unit and to generate a speech bitstream; an audio encoding unit to encode the input signal according to the selection of the module selection unit and to generate an audio bitstream; and a bitstream generation unit to generate an output bitstream from the speech encoding unit or the audio encoding unit according to the selection of the module selection unit.

Abstract: Speech is coded using an orthogonal search by calculating a search reference value. An adaptive codevector representing a pitch component is generated. A random codevector representing a random component is also generated. The orthogonal search further includes generating a synthetic speech signal by a synthesis filter being excited by the adaptive codevector and the random codevector. A distortion between the input speech signal and the synthetic speech signal is calculated. One random codevector that minimizes the distortion is selected.

Abstract: For a bandwidth extension of an audio signal, in a signal spreader the audio signal is temporally spread by a spread factor greater than 1. The temporally spread audio signal is then supplied to a demicator to decimate the temporally spread version by a decimation factor matched to the spread factor. The band generated by this decimation operation is extracted and distorted, and finally combined with the audio signal to obtain a bandwidth extended audio signal. A phase vocoder in the filterbank implementation or transformation implementation may be used for signal spreading.

Abstract: A CELP speech decoder includes an adaptive codebook that generates an adaptive code vector and a random codebook that generates a random code vector. The random codebook includes an input vector provider that provides an input vector including at least one pulse, each pulse having a position and a polarity, a fixed waveform storage that stores at least one fixed waveform, and a selector that selects at least one of a first process and a second process based on a value of an adaptive codebook gain. The random codebook further includes a convolution section that generates the random code vector by convoluting the at least one fixed waveform with the input vector when the first process is selected. A synthesis filter outputs synthesized speech by performing linear prediction coefficient synthesis on a signal based on the adaptive code vector and the random code vector.

Abstract: Provided is an audio decoding device which can adjust the high-range emphasis degree in accordance with a background noise level.

Abstract: A communication system (100) includes devices (102, 104, 200) for transmitting and receiving digital audio. The devices use audio encoders (210, 804) and decoders (222, 916) such as ACELP or DCT/IDCT to compress and decompress audio and use arithmetic encoders (212) and decoders (220) to encode and decode the compressed audio on-the-fly (without a codebook of pre-stored codes).

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: A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.

Abstract: There is provided an audio encoding device for correcting the component having insufficient encoding capability in the core layer by an extended layer. In this device, a core layer encoding unit (101) encodes an audio signal, an extended layer encoding unit (150) encodes an encoding residual of the core layer encoding unit (101), a characteristic correction inverse filter (102 arranged at the pre-stage of an LPC synthesis filter (104) subjects the component having insufficient encoding capability in the core layer to the inverse characteristic correction process, and a characteristic correction filter (105) arranged at the post-stage of the LPC synthesis filter (104) performs a process for characteristic correction of the synthesis signal inputted from the LPC synthesis filter (104).

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

Abstract: Speech is coded using an orthogonal search by calculating a search reference value. An adaptive codevector representing a pitch component is generated. A random codevector representing a random component is also generated. The orthogonal search further includes generating a synthetic speech signal by a synthesis filter being excited by the adaptive codevector and the random codevector. A distortion between the input speech signal and the synthetic speech signal is calculated. One random codevector that minimizes the distortion is selected.

Abstract: A method and apparatus for processing speech in a wireless communication system uses CELP speech encoded signals. A speech input receives samples of a speech signal and a codebook analysis block for selects an index of a code from at least one of a plurality of codebooks. A weighted synthesis filter is used in the generation of a prediction error between a predicted current sample and a current sample of the speech samples. The index is transmitted to the receiver to enable reconstruction of the speech signal at the receiver.

Abstract: Provided is an audio encoding device for modeling a spectrum waveform and accurately restoring the spectrum waveform. The audio encoding device includes: an FFT unit (104) for subjecting a spectrum amplitude of a drive sound source signal to an FFT process to obtain an FFT transform coefficient; a second spectrum amplitude calculation unit (105) for calculating a second spectrum amplitude of the FFT transform coefficient; a peak point position identification unit (106) for identifying the positions of the most significant N peaks of the second spectrum amplitude; a coefficient selection unit (107) for selecting FFT transform coefficients corresponding to the identified positions; and a quantization unit (108) for quantizing the selected FFT transform coefficients.

Abstract: According to the invention, an excitation signal is generated as a result of sampled excitation values in order to excite an audio synthesis filter, the generated sampled excitation values being continuously stored in an adaptive codebook. A noise generator is provided which continuously generates random sampled values. A sequence of the stored sampled excitation values is selected from the adaptive codebook based on a fed audio fundamental frequency parameter by means of which a time gap between the sequence that is to be selected and the actual time reference is predefined. The excitation signal is generated by mixing the selected sequence with a random sequence encompassing actual random sampled valued of the noise generator.

Abstract: A scalable encoding device for realizing scalable encoding by CELP encoding of a stereo sound signal and improving the encoding efficiency. In this device, an adder and a multiplier obtain an average of a first channel signal CH1 and a second channel signal CH2 as a monaural signal M. A CELP encoder for a monaural signal subjects the monaural signal M to CELP encoding, outputs the obtained encoded parameter to outside, and outputs a synthesized monaural signal M? synthesized by using the encoded parameter to a first channel signal encoder. By using the synthesized monaural signal M? and the second channel signal CH2, the first channel signal encoder subjects the first channel signal CH1 to CELP encoding to minimize the sum of the encoding distortion of the first channel signal CH1 and the encoding distortion of the second channel signal CH2.

Abstract: The invention proposed a Dual-Pulse Excitation Model; wherein two pulses of each pair of pulses are always adjacent each other. Only one position index for each pair of pulses needs to be sent to the decoder, which saves bits to code all pulse positions. The magnitudes of each pair of pulses have limited number of patterns. Because the two pulses are adjacent each other, each pair of pulses with different magnitudes can produce different high-pass and/or low-pass effect. Since the magnitudes have enough variation, it is possible to assign the candidate positions of each pair of pulses within a small range in order to save the searching complexity.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal.

Abstract: The present invention provides a frame erasure concealment device and method that is based on reestimating gain parameters for a code excited linear prediction (CELP) coder. During operation, when a frame in a stream of received data is detected as being erased, the coding parameters, especially an adaptive codebook gain gp and a fixed codebook gain gc, of the erased and subsequent frames can be reestimated by a gain matching procedure. By using this technique with the IS-641 speech coder, it has been found that the present invention improves the speech quality under various channel conditions, compared with a conventional extrapolation-based concealment algorithm.

Abstract: There is disclosed a speech processing device in which prediction taps for finding prediction values of the speech of high sound quality are extracted from the synthesized sound obtained on affording linear prediction coefficients and residual signals, generated from a preset code, to a speech synthesis filter, speech of high sound quality being higher in sound quality than the synthesized sound, and in which the prediction taps are used along with preset tap coefficients to perform preset predictive calculations to find the prediction values of the speech of high sound quality. The speech of high sound quality is higher in sound quality than the synthesized sound.