Abstract: An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.

Abstract: Provided are a method and an apparatus for encoding and decoding an audio signal. A method for encoding an audio signal includes receiving a transformed audio signal, dividing the transformed audio signal into a plurality of subbands, performing a first sinusoidal pulse coding operation on the subbands, determining a performance region of a second sinusoidal pulse coding operation among the subbands on the basis of coding information of the first sinusoidal pulse coding operation, and performing the second sinusoidal pulse coding operation on the determined performance region, wherein the first sinusoidal pulse coding operation is performed variably according to the coding information. Accordingly, it is possible to further improve the quality of a synthesized signal by considering the sinusoidal pulse coding of a lower layer when encoding or decoding an audio signal in an upper layer by a layered sinusoidal pulse coding scheme.

Abstract: Methods and apparatus for audio and speech processing including generating a plurality of frames, each of the frames comprising a plurality of transform coefficients, and allocating bits to the transform coefficients in each of the frames such that at least two of the transform coefficients in the same frame have different bit allocations and the total number of the bits allocated to the transform coefficients in at least two of the frames is equal.

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: 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: This invention relates to indexing an input vector contained in a set of vectors of a plurality of sets of vectors. The indexing comprises performing, in case that the input vector is contained in a set of vectors of a pre-defined group of one or more sets of vectors of the plurality of sets of vectors, a specific processing that is adapted to a characteristic of the sets of vectors in the pre-defined group of sets of vectors and is only applicable in case of input vectors contained in sets of vectors with the characteristic. The indexing further comprises performing, in case that the input vector is not contained in a set of vectors of the pre-defined group of sets of vectors, a general processing. The invention further relates to an according determining of a target vector contained in a set of vectors of a plurality of sets of vectors based on an index associated with said target vector.

Abstract: The present invention provides a computationally efficient technique for compression encoding of an audio signal, and further provides a technique to enhance the sound quality of the encoded audio signal. This is accomplished by including more accurate attack detection and a computationally efficient quantization technique. The improved audio coder converts the input audio signal to a digital audio signal. The audio coder then divides the digital audio signal into larger frames having a long-block frame length and partitions each of the frames into multiple short-blocks. The audio coder then computes short-block audio signal characteristics for each of the partitioned short-blocks based on changes in the input audio signal.

Abstract: In syllable or vowel or phone boundary detection during speech, an auditory spectrum may be determined for an input window of sound and one or more multi-scale features may be extracted from the auditory spectrum. Each multi-scale feature can be extracted using a separate two-dimensional spectro-temporal receptive filter. One or more feature maps corresponding to the one or more multi-scale features can be generated and an auditory gist vector can be extracted from each of the one or more feature maps. A cumulative gist vector may be obtained through augmentation of each auditory gist vector extracted from the one or more feature maps. One or more syllable or vowel or phone boundaries in the input window of sound can be detected by mapping the cumulative gist vector to one or more syllable or vowel or phone boundary characteristics using a machine learning algorithm.

Abstract: The restoration of images by vector quantization utilizing visual patterns is disclosed. One disclosed embodiment comprises restoring detail in a transition region of an unrestored image, by first identifying the transition region and forming blurred visual pattern blocks. These blurred visual pattern blocks are compared to a pre-trained codebook, and a corresponding high-quality visual pattern blocks is obtained. The high-quality visual pattern block is then blended with the unrestored image to form a restored image.

Abstract: A method and apparatus are for performing one of encoding and decoding a code word that is used to communicate a portion of a signal. For encoding, at least a portion of a code word is encoded from a signal based value using an approximation of a combinatorial function, wherein the signal based value represents one or more aspects of a signal. For decoding, at least a portion of a code word is decoded to a signal based value using an approximation of a combinatorial function, wherein the signal based value represents one or more aspects of a signal. The approximation of the combinatorial function is based on a linear combination of a set of basis functions.

Abstract: A coding method, a decoding method, a coding-decoding (codec) method, a codec system and relevant apparatuses are disclosed. The coding method includes: obtaining an amplitude vector and a length vector corresponding to a vector to be coded; sorting elements of the amplitude vector and elements of the length vector; and obtaining a position index value according to the sorted amplitude vector and the sorted length vector. A decoding method, a codec system, and relevant apparatuses are also provided.

Abstract: Apparatus (119) for encoding at least one parameter associated with a signal source for transmission over k frames to a decoder comprises a processor (119) which is configured in operation to assign a predetermined bit pattern to n bits associated with the at least one parameter of a first frame of k frames and set the n bits associated with the at least one parameter of each of k?1 subsequent frames to values, such that the values of the n bits of the k?1 subsequent frames represent the at least one parameter. The predetermined bit pattern indicates a start of the at least one parameter.

Abstract: A scalable encoding apparatus, a scalable decoding apparatus and the like are disclosed which can achieve a band scalable LSP encoding that exhibits both a high quantization efficiency and a high performance. In these apparatuses, a narrow band-to-wide band converter receives and converts a quantized narrow band LSP to a wide band, and then outputs the quantized narrow band LSP as converted (i.e., a converted wide band LSP parameter) to an LSP-to-LPC converter. The LSP-to-LPC converter converts the quantized narrow band LSP as converted to a linear prediction coefficient and then outputs it to a pre-emphasizer. The pre-emphasizer calculates and outputs the pre-emphasized linear prediction coefficient to an LPC-to-LSP converter. The LPC-to-LSP converter converts the pre-emphasized linear prediction coefficient to a pre-emphasized quantized narrow band LSP as wide band converted, and then outputs it to a prediction quantizer.

Abstract: A plurality of samples are vector-quantized to obtain a vector quantization index and quantized values; bits are assigned in a predetermined order of priority based on auditory perceptual characteristics to one or more sets of sample positions among a plurality of sets of sample positions, each set having a plurality of sample positions and being given an order of priority based on the auditory perceptual characteristics, the number of bits not being larger than the number of bits obtained by subtracting the number of bits used for a code corresponding to the vector quantization index from the number of bits assigned for the code corresponding to the vector quantization index; and index information indicating a group of coefficients that minimizes the sum of the error between the value of each sample included in each of the sets of sample positions to which the bits are assigned and the value obtained by multiplying the quantized value of each sample included in the set of sample positions by a coefficient co

Abstract: A method of multi-path trellis coded quantization (TCQ) usable in a speech coding system, and a quantizer using the method. Specifically the method includes calculating accumulated distortions corresponding to 2N survivor paths, wherein N indicates an integer greater than two, each of the 2N survivor paths is going towards one of nodes at an i th stage of a trellis, and i indicates an integer greater than zero, comparing the accumulated distortions respectively corresponding to the 2N survivor paths to select N paths among the 2N survivor paths, wherein the accumulated distortions corresponding to selected N paths are smaller than the accumulated distortions corresponding to unselected N paths establishing the selected N paths as survivor paths going toward an i+1 th stage, and selecting an optimal path among the 2N survivor paths corresponding to each node of a last stage.

Abstract: In one aspect, a method of processing a voice signal to extract information to facilitate training a speech synthesis model is provided. The method comprises acts of detecting a plurality of candidate features in the voice signal, performing at least one comparison between one or more combinations of the plurality of candidate features and the voice signal, and selecting a set of features from the plurality of candidate features based, at least in part, on the at least one comparison. In another aspect, the method is performed by executing a program encoded on a computer readable medium. In another aspect, a speech synthesis model is provided by, at least in part, performing the method.

Abstract: The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.

Abstract: To improve the encoding compressibility of prediction residuals. An encoder performs prediction analysis of input time-series signals to generate prediction residuals expressed by integers, and sets an integer separation parameter that depends on the amplitude of the prediction residuals for each certain time segment. The encoder selects a side information code table corresponding to an index representing the prediction effectiveness of the time-series signals from a set of side information code tables including a side information code table used for variable length coding of side information corresponding to the separation parameter.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for receiving (i) audio data that encodes a spoken natural language query, and (ii) environmental audio data, obtaining a transcription of the spoken natural language query, determining a particular content type associated with one or more keywords in the transcription, providing at least a portion of the environmental audio data to a content recognition engine, and identifying a content item that has been output by the content recognition engine, and that matches the particular content type.

Abstract: The invention relates to a method, computer, computer program and computer program product for speech quality estimation. The method comprises the steps of: determining a coding distortion parameter (QCOD), a bandwidth related distortion parameter (BW) and a presentation level distortion parameter (PL) of a speech signal; extracting a first coefficient (?l) and a second coefficient (?2), the first coefficient and the second coefficient being dependent on the coding distortion parameter; and calculating a signal quality measure (Q), where the signal quality measure is QCOD+?1BW+?2PL using the signal quality measure in a quality estimation of the speech signal.

Abstract: Techniques disclosed herein include systems and methods for voice-enabled searching. Techniques include a co-occurrence based approach to improve accuracy of the 1-best hypothesis for non-phrase voice queries, as well as for phrased voice queries. A co-occurrence model is used in addition to a statistical natural language model and acoustic model to recognize spoken queries, such as spoken queries for searching a search engine. Given an utterance and an associated list of automated speech recognition n-best hypotheses, the system rescores the different hypotheses using co-occurrence information. For each hypothesis, the system estimates a frequency of co-occurrence within web documents. Combined scores from a speech recognizer and a co-occurrence engine can be combined to select a best hypothesis with a lower word error rate.

Abstract: System and method to improve intelligibility of coded speech, the method including: receiving an encoded speech signal from a network; extracting an encoded media data stream and one or more control data packets from the encoded speech signal; decoding the encoded media data stream to produce a decoded speech signal; boosting an upper spectral portion of the decoded speech signal to produce a boosted speech signal; and outputting the boosted speech signal. In another embodiment, the method may include: receiving an uncoded speech signal; processing the uncoded speech signal, wherein the processing comprises generating an unencoded data stream from the uncoded speech signal; boosting an upper spectral portion of the unencoded data stream to produce a boosted speech signal; encoding the boosted speech signal to produce an encoded speech signal; and outputting the boosted speech signal.

Abstract: A method and apparatus to convert a linear predictive coding (LPC) coefficient into a coefficient having order characteristics, such as a line spectrum frequency (LSF), and to vector quantize the coefficient having the order characteristics when a speech signal is encoded. The method and apparatus split the vector of the coefficient having the order characteristics into a plurality of subvectors, select a codebook in which an available bit is variably allocated to each subvector according to distribution of elements of each subvector, and quantize each subvector according to the selected codebook. The method and apparatus use normalized codebooks.

Abstract: The invention relates to a method of identifying and correcting errors in a noisy binary mask. An object of the present invention is to provide a scheme for improving a binary mask representing speech. The problem is solved in that the method comprises a) providing a noisy binary mask comprising a binary representation of the power density of an acoustic signal comprising a target signal and a noise signal at a predefined number of discrete frequencies and a number of discrete time instances; b) providing a statistical model of a clean binary mask representing the power density of the target signal; and c) using the statistical model to detect and correct errors in the noisy binary mask. This has the advantage of providing an alternative and relatively simple way of improving an estimate of a binary mask representing a speech signal. The invention may e.g. be used for speech processing, e.g. in a hearing instrument.

Abstract: A system and method may be configured to analyze audio information derived from an audio signal. The system and method may track sound pitch across the audio signal. The tracking of pitch across the audio signal may take into account change in pitch by determining at individual time sample windows in the signal duration an estimated pitch and a representation of harmonic envelope at the estimated pitch. The estimated pitch and the representation of harmonic envelope may then be implemented to determine an estimated pitch for another time sample window in the signal duration with an enhanced accuracy and/or precision.

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: A method and an apparatus to encode and decode a speech signal using a code excited linear prediction (CELP) algorithm. In order to reduce a bit rate without degrading performance in an enhancement layer based on CELP, each of a fixed codebook of a core layer and a fixed codebook of the enhancement layer is divided into a plurality of spaces. The spaces of the fixed codebook of the enhancement layer excludes a space corresponding to a least distorted space determined from among the spaces of the fixed codebook of the core layer are searched.

Abstract: A method for processing audio data includes determining a first common scalefactor value for representing quantized audio data in a frame. A second common scalefactor value is determined for representing the quantized audio data in the frame. A line equation common scalefactor value is determined from the first and second common scalefactor values.

Abstract: Systems and methods are described for performing packet loss concealment (PLC) to mitigate the effect of one or more lost frames within a series of frames that represent a speech signal. In accordance with the exemplary systems and methods, PLC is performed by searching a codebook of speech-related parameter profiles to identify content that is being spoken and by selecting a profile associated with the identified content for use in predicting or estimating speech-related parameter information associated with one or more lost frames of a speech signal. The predicted/estimated speech-related parameter information is then used to synthesize one or more frames to replace the lost frame(s) of the speech signal.

Abstract: A method and apparatus of providing an audio output to a user in a communications system in which the audio to be output to a user, preferably an audio frame, is assessed before it is broadcast to the user, and then selectively changed on the basis of the assessment. The assessment may be carried out in the audio encoding process, in the audio decoding process and/or after the audio decoding process. The selective changing of the audio output may comprise selectively replacing the audio output and/or re-encoding of the audio output.

Abstract: A sound reproducing device has a loudspeaker arranged to produce sound from an audio signal provided by an audio signal source. A microphone is positioned to pick up ambient noise and generate a microphone signal which comprises the noise. An ambient noise cancellation (ANC) system receives the microphone signal from the microphone and generates anti-noise corresponding to the ambient noise in the microphone signal. An automatic polarity adaptation (AAP) system monitors the ANC system and, when a decision criterion is fulfilled, causes a switch in polarity for the generated anti-noise.

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: A decoding device is capable of flexibly calculating high-band spectrum data with a high accuracy in accordance with an encoding band selected by an upper-node layer of the encoding side. In this device: a first layer decoder decodes first layer encoded information to generate a first layer decoded signal; a second layer decoder decodes second layer encoded information to generate a second layer decoded signal; a spectrum decoder performs a band extension process by using the second layer decoded signal and the first layer decoded signal up-sampled in an up-sampler so as to generate an all-band decoded signal; and a switch outputs the first layer decoded signal or the all-band decoded signal according to the control information generated in a controller.

Abstract: A voice encoding device accurately encodes a spectrum shape of a signal having a strong tonality such as a vowel. The device includes: a sub-band divider which divides a first layer error conversion coefficient to be encoded into M sub-bands so as to generate M sub-band conversion coefficients; a shape vector encoder which performs encoding on each of the M sub-band conversion coefficients so as to obtain M shape encoded information and calculates a target gain of each of the M sub-band conversion coefficients; a gain vector former which forms one gain vector by using M target gains; a gain vector encoder which encodes the gain vector so as to obtain gain encoded information; and a multiplexer which multiplexes the shape encoded information with the gain encoded information.

Abstract: There is provided an audio coding device which appropriately sets the quantization bit number by a small calculation amount in each stage when coding an input audio signal by performing multi-stage normalization/quantization. A quantization information calculation section determines total quantization information idwl0, based on normalization information idsf, and allocates the total quantization information idwl0 for quantization information idwl1 and quantization information idwl2. At this time, the quantization information calculation section limits the quantization information idwl1 by a limiter lim1, and allocates the total quantization information idwl0 for quantization information idwl1. If the quantization information idwl1 exceeds the limiter lim1, the excess is allocated for the quantization information idwl2. A first normalization section and a first quantization section normalizes and quantizes a frequency spectrum mdspec1 in the first stage.

Abstract: An adaptive sound source vector quantization device includes a first pitch cycle instructor, a search range calculator, and a second pitch cycle instructor. The first pitch cycle instructor successively instructs pitch cycle search candidates in a predetermined search range having a search resolution which transits over a predetermined pitch cycle candidate for the first sub-frame. The search range calculator calculates a predetermined range before and after the pitch cycle of the first sub-frame as the pitch cycle search range for the second sub-frame, if the predetermined range includes the predetermined pitch cycle search candidate. In the predetermined range, the search resolution transits over a boundary defined by the predetermined pitch cycle. The second pitch cycle instructor successively instructs the pitch cycle search candidates in the search range for the second sub-frame.

Abstract: A method, system, and computer program for enhancing a signal are presented. The signal is received, and energy estimates of the signal may be determined. At least one characteristic of the signal may be inferred based on the energy estimates. A mask may be generated based, in part, on the at least one characteristic. In turn, the mask may be applied to the signal to produce an enhanced signal, which may be outputted.

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: Codebook indices for a scalable speech and audio codec may be efficiently encoded based on anticipated probability distributions for such codebook indices. A residual signal from a Code Excited Linear Prediction (CELP)-based encoding layer may be obtained, where the residual signal is a difference between an original audio signal and a reconstructed version of the original audio signal. The residual signal may be transformed at a Discrete Cosine Transform (DCT)-type transform layer to obtain a corresponding transform spectrum. The transform spectrum is divided into a plurality of spectral bands, where each spectral band having a plurality of spectral lines. A plurality of different codebooks are then selected for encoding the spectral bands, where each codebook is associated with a codebook index. A plurality of codebook indices associated with the selected codebooks are then encoded together to obtain a descriptor code that more compactly represents the codebook indices.

Abstract: For an enhanced sequential compression of data vectors in a respective compression pass, a current data vector is mapped to at least one current code vector of at least one codebook in at least one quantization stage. The at least one codebook is reordered taking account of at least one intermediate result from the current compression pass and at least one intermediate result from a preceding compression pass. At least one codebook index that is associated in the at least one reordered codebook to the at least one current code vector is then provided for further use. For a decompression of compressed data vectors represented by such codebook indices, at least one codebook index is mapped to at least one code vector of at least one equally reordered codebook.

Abstract: Information that includes first information identifying integer quotients obtained by divisions using prediction residuals or integers not smaller than 0 that increase monotonically with increases in the amplitude of the prediction residuals, as dividends, and a separation parameter decided for a time segment corresponding to the prediction residuals or a mapped integer value of the separation parameter, as a modulus, and second information identifying the remainders obtained when the dividends are divided by the modulus is generated as a code corresponding to the prediction residuals, and each piece of side information that includes the separation parameter is subjected to variable length coding.

Abstract: A method of transmitting speech data to a remote device in a distributed speech recognition system, includes the steps of: dividing an input speech signal into frames; calculating, for each frame, a voice activity value representative of the presence of speech activity in the frame; grouping the frames into multiframes, each multiframe including a predetermined number of frames; calculating, for each multiframe, a voice activity marker representative of the number of frames in the multiframe representing speech activity; and selectively transmitting, on the basis of the voice activity marker associated with each multiframe, the multiframes to the remote device.

Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises a linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; a quantization unit for quantizing a transform domain signal; a long term prediction unit for determining an estimation of the frame of the filtered input signal based on a reconstruction of a previous segment of the filtered input signal; and a transform domain signal combination unit for combining, in the transform domain, the long term prediction estimation and the transformed input signal to generate the transform domain signal.

Abstract: The embodiments of the present invention relate to a compression coding and decoding method, a coder, a decoder and a coding device. The compression coding method includes: extracting sign information of an input signal to obtain an absolute value signal of the input signal; obtaining a residual signal of the absolute value signal by using a prediction coefficient, where the prediction coefficient is obtained by prediction and analysis that are performed according to a signal characteristic of the absolute value signal of the input signal; and multiplexing the residual signal, the sign information and a coding parameter to output a coding code stream, after the residual signal, the sign information and the coding parameter are respectively coded, so as to improve compression efficiency of a voice and audio signal.

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: A method and system for adaptive auto-encoders is disclosed. An input audio training signal may be transformed into a sequence of feature vectors, each bearing quantitative measures of acoustic properties of the input audio training signal. An auto-encoder may process the feature vectors to generate an encoded form of the quantitative measures, and a recovered form of the quantitative measures based on an inverse operation by the auto-encoder on the encoded form of the quantitative measures. A duplicate copy of the sequence of feature vectors may be normalized to form a normalized signal in which supra-phonetic acoustic properties are reduced in comparison with phonetic acoustic properties of the input audio training signal. The auto-encoder may then be trained to compensate for supra-phonetic features by reducing the magnitude of an error signal corresponding to a difference between the normalized signal and the recovered form of the quantitative measures.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for receiving (i) audio data that encodes a spoken natural language query, and (ii) environmental audio data, obtaining a transcription of the spoken natural language query, determining a particular content type associated with one or more keywords in the transcription, providing at least a portion of the environmental audio data to a content recognition engine, and identifying a content item that has been output by the content recognition engine, and that matches the particular content type.

Abstract: The present disclosure provides an audio quantization coding and decoding device and a method thereof. In the method, before a quantization coding process is performed on a digital signal, the signal is pre-processed, the digital signal is split into multiple frames based on positive and negative half periods of the signal, and all audio data between two adjacent zero-crossing points belongs to the same positive and negative half periods, so as to have the same sign-bit. A pre-processing module groups the numeric data belonging to the same positive and negative half periods into the same frame. When coding, an audio quantization coding module only needs to record a sign-bit of the frame at a head of the frame, so the sign-bit of each batch of voice data in the frame may be omitted to reduce a data amount or improve a resolution of each batch of voice data.

Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.