Abstract: An encoder performs context-adaptive arithmetic encoding of transform coefficient data. For example, an encoder switches between coding of direct levels of quantized transform coefficient data and run-level coding of run lengths and levels of quantized transform coefficient data. The encoder can determine when to switch between coding modes based on a pre-determined switch point or by counting consecutive coefficients having a predominant value (e.g., zero). A decoder performs corresponding context-adaptive arithmetic decoding.

Abstract: Provided is an encoding apparatus. A threshold value calculating unit (32) calculates a threshold value from a statistical amount of conversion factors of an extended band. A representative conversion factor extracting unit (33) uses the calculated threshold value to extract conversion factors having large amplitudes. If the number of extracted conversion factors does not reach a specified number, the threshold value calculating unit (32) determines, in accordance with a lacking number of conversion factors, an amount by which the threshold value should be lowered, and modifies the threshold value accordingly. The representative conversion factor extracting unit (33) uses the threshold value, which has been modified, to extract conversion factors.

Abstract: Multiple channels of audio are combined either to a monophonic composite signal or to multiple channels of audio along with related auxiliary information from which multiple channels of audio are reconstructed, including improved downmixing of multiple audio channels to a monophonic audio signal or to multiple audio channels and improved decorrelation of multiple audio channels derived from a monophonic audio channel or from multiple audio channels. Aspects of the disclosed invention are usable in audio encoders, decoders, encode/decode systems, downmixers, upmixers, and decorrelators.

Abstract: A media signal is accessed, which has been generated with one or more first processing operations. The media signal includes one or more sets of artifacts, which respectively result from the one or more processing operations. One or more features are extracted from the accessed media signal. The extracted features each respectively correspond to the one or more artifact sets. Based on the extracted features, a conditional probability score and/or a heuristically based score is computed, which relates to the one or more first processing operations.

Abstract: Provided are an apparatus and method for coding and decoding a multi object audio signal with multi channel. The apparatus includes: a multi channel encoding means for down-mixing an audio signal including a plurality of channels, generating a spatial cue for the audio signal including the plurality of channels, and generating first rendering information including the generated spatial cue; and a multi object encoding unit for down-mixing an audio signal including a plurality of objects, which includes the down-mixed signal from the multi channel encoding unit, generating a spatial cue for the audio signal including the plurality of objects, and generating second rendering information including the generated spatial cue, wherein the multichannel encoding unit generates a spatial cue for the audio signal including the plurality of objects regardless of a Coder-DECoder (CODEC) scheme the limits the multi channel encoding unit.

Abstract: In encoding, the number of bits to be assigned to codes corresponding to noise or a pulse sequence obtained according to prediction analysis applied to time series signals included in a predetermined time interval is switched according to whether an index that indicates a level of periodicity and/or stationarity of input time series signals satisfies a condition that indicates high periodicity and/or high stationarity or a condition that indicates low periodicity and/or low stationarity, to acquire the codes corresponding to the noise and the pulse sequence. In decoding, a decoding mode for codes corresponding to noise or a pulse sequence included in codes corresponding to a predetermined time interval is switched according to the same criterion as that described above to decode the codes corresponding to the noise or the pulse sequence to acquire noise or a pulse sequence corresponding to the predetermined time interval.

Abstract: Certain aspects of the present disclosure relate to techniques for low-complexity encoding (compression) of broad class of signals, which are typically not well modeled as sparse signals in either time-domain or frequency-domain. First, the signal can be split in time-segments that may be either sparse in time domain or sparse in frequency domain, for example by using absolute second order differential operator on the input signal. Next, different encoding strategies can be applied for each of these time-segments depending in which domain the sparsity is present.

Abstract: The present invention provides a method for processing audio signals, and the method comprises the steps of: receiving input audio signals corresponding to a plurality of spectral coefficients; obtaining location information that indicates a location of a particular spectral coefficient among said spectral coefficients, on the basis of energy of said input signals: generating a shape vector by using said location information and said spectral coefficients; determining a codebook index by searching for a codebook corresponding to said shape vector; and transmitting said codebook index and said location information, wherein said shape vector is generated by using a part which is selected from said spectral coefficients, and said selected part is selected on the basis of said location information.

Abstract: A system for processing compressed audio includes a signal enhancer module configured to generate one or more signal treatments. The one or more signal treatments may be generated by the signal enhancer module based on analysis of the incoming audio signal. Alternatively, or in addition, characteristics of the incoming audio signal may be provided to the signal enhancer module for use in generating the one or more signal treatments. The one or more signal treatments may be added to the audio signals.

Abstract: The present disclosure proposes a method and an apparatus to enhance reverberated speech by applying reverberation detection in conjunction with reverberation cancellation. The reverberation detection is based on Kurtosis of cross correlation of LPC residue and outputs the result of the reverberation detection to the reverberation cancelling system. The reverberation cancellation receives the result from the reverberation detection, and the cancellation is based on dual adaptive filtering in LP residue and time domain.

Abstract: Briefly, in accordance with one or more embodiments, a wireless signal is transmitted to a remote device. The remote device determines a precoding matrix indicator (PMI) in response to the wireless signal based at least in part on a hierarchical codebook. The PMI is then received as reported back from the remote device.

Abstract: A method and apparatus for determining an auditory pattern associated with an audio segment. An average intensity at each of a first plurality of detector locations on an auditory scale based at least in part on a first plurality of frequency components that describe a signal is determined. A plurality of tonal bands in the audio segment, wherein each tonal band comprises a particular range of detector locations of the first plurality of detector locations is determined. Corresponding strongest frequency components in the tonal bands are determined. A plurality of non-tonal bands is determined, and each non-tonal band is subdivided into multiple sub-bands. Corresponding combined frequency components that are representative of a combined sum of intensities of the first plurality of frequency components that is in a corresponding sub-band are determined. An auditory based on the corresponding strongest frequency components and the corresponding combined frequency components is determined.

Abstract: In a quantization method for quantizing a received excitation signal in a communication system performing the steps of re-shuffling S301 the elements of the received excitation signal to provide a re-shuffled excitation signal; coding S302 the re-shuffled excitation signal with a variable bit-rate algorithm to provide a coded excitation signal; and reassigning S303 codewords of the coded excitation signal if a number of used bits exceeds a predetermined fixed bit rate requirement to provide a quantized excitation signal.

Abstract: An “Overcomplete Audio Coder” provides various techniques for overcomplete encoding audio signals using an MCLT-based predictive coder. Specifically, the Overcomplete Audio Coder uses unrestricted polar quantization of MCLT magnitude and phase coefficients. Further, quantized magnitude and phase coefficients are predicted based on properties of the audio signal and corresponding MCLT coefficients to reduce the bit rate overhead in encoding the audio signal. This prediction allows the Overcomplete Audio Coder to provide improved continuity of the magnitude of spectral components across encoded signal blocks, thereby reducing warbling artifacts. Coding rates achieved using these prediction techniques are comparable to that of encoding an orthogonal representation of an audio signal, such as with modulated lapped transform (MLT)-based coders.

Abstract: A method for compressing an image includes decomposing the image into one or more regions. A region of the image is selected to be evaluated. The selected region is transformed and quantized if the region does not meet a predetermined compression acceptability criteria. The predetermined compression acceptability criteria may include a specific bit rate, a specific image quality, or combinations thereof. If the region does not meet the predetermined compression acceptability criteria after the region has been transformed and quantized, then the transformation and quantization settings are adjusted and the region is transformed and quantized using the adjusted settings. The region is then encoded when the predetermined compression acceptability criteria has been reached. The encoding may include additional compression stages.

Abstract: An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.

Abstract: Methods and systems for using pitch predictors in speech/audio coders are provided. Techniques for optimal pre- and post-filtering are presented, and a general result that post-filtering is more effective than pre-filtering is derived. A practical paired-zero filter design for the low-rate regime is proposed, and this design is extended to handle frequency-dependent periodicity levels. Further, the methods described provide a general performance measure for a post-filter that only uses information available at the decoder, thereby allowing for the optimization or selection of a post-filter without increasing the rate.

Abstract: An audio signal is processed in the frequency domain to convert an input signal format to an output signal format. That is, a multichannel audio signal intended for playback over a predefined speaker layout can be formatted to achieve spatial reproduction over a different layout comprising a different number of speakers.

Abstract: Disclosed is an encoding device that improves the quality of a decoded signal in a hierarchical coding (scalable coding) method, wherein a band to be quantized is selected for every level (layer). The encoding device (101) is equipped with a second layer encoding unit (205) that selects a first band to be quantized of a first input signal from among a plurality of sub-bands, and that generates second layer encoding information containing first band information of said band; a second layer decoding unit (206) that generates a first decoded signal using the second layer encoding information; an addition unit (207) that generates a second input signal using the first input signal and the first decoded signal; and a third layer encoding unit (208) that selects a second band to be quantized of the second input signal using the first decoded signal, and that generates third layer encoding information.

Abstract: Embodiments are generally directed to systems and methods for bit allocation and band partitioning for gain-shape vector quantization in an audio codec. An audio codec implements a method that uses an implicit, dynamic scheme to allow an encoder and decoder to recreate a series of bit allocation decisions for gain and shape without transmitting additional side information for each decision, based on the number of bits that are left remaining and available in a given packet. For implementation in practical codecs, the band comprising the allocation of bits for the shape is recursively split into equal partitions until the number of bits allocated to each partition is less than the maximum codebook size.

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: An electronic device for coding a transient frame is described. The electronic device includes a processor and executable instructions stored in memory that is in electronic communication with the processor. The electronic device obtains a current transient frame. The electronic device also obtains a residual signal based on the current transient frame. Additionally, the electronic device determines a set of peak locations based on the residual signal. The electronic device further determines whether to use a first coding mode or a second coding mode for coding the current transient frame based on at least the set of peak locations. The electronic device also synthesizes an excitation based on the first coding mode if the first coding mode is determined. The electronic device also synthesizes an excitation based on the second coding mode if the second coding mode is determined.

Abstract: In an embodiment, a voice recognition apparatus includes: a program information storage unit; a dictionary storage unit; a calculating unit; an updating unit; a receiving unit; a recognizing unit; and an operation control unit. The program information storage unit stores metadata of a broadcast program with a user's viewing state. The dictionary storage unit stores a recognition dictionary including a recognition word and a priority of the recognition word. The calculating unit calculates a first score of a degree of the user's preference on a feature word based on the metadata and the viewing state. The updating unit updates the priority of the recognition word including the feature word according to the first score. The recognizing unit recognizes a voice using the recognition dictionary. The operation control unit controls an operation on the broadcast program based on a recognition result.

Abstract: A system and method for speech quality detection is included. The method may include receiving, at a computing device, a first speech signal associated with a particular user. The method may include extracting one or more short-term features from the first speech signal wherein extracting short-term features includes extracting a time frame of between 10-50 ms. The method may also include determining one or more statistics of each of the one or more short-term features from the first speech signal. The method may further include classifying the one or more statistics as belonging to one of a set of quality classes.

Abstract: The present invention provides a parametric speech synthesis method and a parametric speech synthesis system.

Abstract: Disclosed are an encoding device and a decoding device which suppress the occurrence of pre-echo artifacts and post-echo artifacts caused by a high layer having a low temporal resolution, and which implement high subjective quality encoding and decoding. An encoding device (100) carries out scalable coding comprising a low layer, and a high layer having a lower temporal resolution than that of the low layer. A start point detection unit (or end point detection unit) (150) determines the start point (or end point) of sections of the decoded low layer signal which have audio, and when the start point (or end point) is determined, a second layer encoding unit (160) selects a bandwidth to be excluded from encoding on the basis of the spectral energy from the decoded first layer signal, excludes the selected bandwidth, and encodes an error signal.

Abstract: A signal processing apparatus includes a speech recognition system and a voice activity detection unit. The voice activity detection unit is coupled to the speech recognition system, and arranged for detecting whether an audio signal is a voice signal and accordingly generating a voice activity detection result to the speech recognition system to control whether the speech recognition system should perform speech recognition upon the audio signal.

Abstract: A system for use in a near field communications (NFC) reader, for detecting the presence of an NFC tag, is disclosed. The NFC reader has an antenna tuned to a resonant frequency. The system comprises: a first signal generator for generating a carrier signal whose frequency varies about the resonant frequency; a second signal generator for generating a modulating signal for modulating the carrier signal; a modulator for modulating the carrier signal with the modulating signal; a driver for driving the antenna with the modulated signal; and a detection system for detecting harmonics of the modulating signal in a signal present in the antenna as a result of the modulated signal. The detector is configured to generate an output when a third or higher order harmonic is detected.

Abstract: A method is provided for hierarchical coding of a digital audio signal comprising, for a current frame of the input signal: a core coding, delivering a scalar quantization index for each sample of the current frame and at least one enhancement coding delivering indices of scalar quantization for each coded sample of an enhancement signal. The enhancement coding comprises a step of obtaining a filter for shaping the coding noise used to determine a target signal and in that the indices of scalar quantization of said enhancement signal are determined by minimizing the error between a set of possible values of scalar quantization and said target signal. The coding method can also comprise a shaping of the coding noise for the core bitrate coding. A coder implementing the coding method is also provided.

Abstract: Vector Quantizer and method therein for efficient vector quantization, e.g. in a transform audio codec. The method comprises comparing an input target vector s with a plurality of centroids, each centroid representing a respective class of codevectors in a codebook. Further, a starting point for a search related to the input target vector in the codebook is determined, based on the result of the comparison. The codevectors in the codebook are sorted according to a distortion measure reflecting the distance between each codevector and the centroids of the classes. The Vector Quantizer and method enables that the class of codevectors comprising the most probable candidate codevectors in regard of the input vector. s may be searched first.

Abstract: Disclosed is an encoding device, wherein the energy information of a given layer is efficiently encoded using a scalable encoding method in which the band to be encoded is selected in each layer, and the quality of decoded signals can be enhanced. An encoding device (101) is equipped with: a second layer encoding unit (205) which generates a second layer encoded information included in which is the first band information of said band; a second layer decoding unit (206) which generates a first decoding signal by using the second layer encoded information; an adding unit (207) which generates a second input signal by using the first decoding signal; and a third layer encoding unit (208) which generates a third layer encoded information included in which is a second band information obtained by selecting a second band to be quantized in the second input signal, and a corrected gain (energy information).

Abstract: This invention introduces audio/speech encoding apparatus audio/speech decoding apparatus, audio/speech encoding method and audio/speech decoding method to efficiently encode the quantization parameters of split multi-rate lattice vector quantization. In this invention, the position of the sub-vector whose codebook indication consumes the most bits is firstly located, and then the value of the codebook is estimated based on the total number of bits available and the bits usage information for other sub-vectors. The difference value is calculated between the actual value and estimated value. Finally, instead of transmitting the codebook indication which consumes the most bits, the position of the sub-vector whose codebook indication consumes the most bits and the difference value between the actual value and the estimated value are transmitted. By applying of the invented method, bits can be saved by the codebook indications.

Abstract: Disclosed is an encoding device which can accurately specify a band having a large error among all the bands by using a small calculation amount. A first position identifier uses a first layer error conversion coefficient indicating an error of a decoding signal for an input signal so as to search for a band having a large error in a relatively wide bandwidth in all the bands of the input signal and generates first position information indicating the identified band. A second position identifier searches for a target frequency band having a large error in a relatively narrow bandwidth in the band identified by the first position identifier and generates second position information indicating the identified target frequency band. An encoder encodes a first layer decoding error conversion coefficient contained in the target frequency band.

Abstract: Disclosed is an encoding device which can accurately specify a band having a large error among all the bands by using a small calculation amount. A first position identifier uses a first layer error conversion coefficient indicating an error of a decoding signal for an input signal so as to search for a band having a large error in a relatively wide bandwidth in all the bands of the input signal and generates first position information indicating the identified band. A second position identifier searches for a target frequency band having a large error in a relatively narrow bandwidth in the band identified by the first position identifier and generates second position information indicating the identified target frequency band. An encoder encodes a first layer decoding error conversion coefficient contained in the target frequency band.

Abstract: An encoding device and encoding method improve a quality of a decoded signal under very low bit rate conditions using a small amount of computation. A spectrum corrector performs correction processing on a subspectrum in each subband in such a manner that samples equal to or greater than a subspectrum average value are left unchanged and samples smaller than the subspectrum average value are replaced by zero. As a result of this, it is possible to significantly reduce the number of bits required to quantize the subspectrums without a substantial reduction in quality in a local searcher and in a multi-rate indexer.

Abstract: A method and apparatus for transcoding audio data. The method includes determining if AAC joint stereo exists, running a reference AC-3 rematrixing when the AAC joint stereo does not exist, when AAC joint stereo does exist, enabling rematrixing when the number of corresponding AAC bands is greater than half the size of the band, otherwise, running reference AC-3 rematrixing.

Abstract: An apparatus configured to model an encoded signal to estimate at least one distribution of the signal, rotate the signal with respect to a lattice, for lattice quantization of the signal, dependent on the at least one distribution of the signal, and quantize the signal rotated with respect to the lattice.

Abstract: Method for speaker identification includes detecting a target speaker's utterance locally; extracting features from the detected utterance locally, analyzing the extracted features in the local device to obtain information on the speaker identification and/or encoding the extracted features locally, transmitting the encoded extracted features to a remote server, decoding and analyzing the received extracted features by the server to obtain information on the speaker identification, and transmitting the information on the speaker identification from the server to the location where the speaker's utterance was detected. The method further includes detecting speech activity locally. Extracting features, encoding the extracted features, and/or transmitting the encoded extracted features to the server, are only performed if speech activity above some predetermined threshold is detected.

Abstract: A signal processing device is provided, including a prediction error calculating unit that calculates an error signal between a left signal l(n) and a prediction signal of the left signal l (n) predicted from a right signal r(n), a gain adjusting unit that makes a gain adjustment and outputs an error signal, a first adder that adds the left signal l(n) and the error signal and outputs, and a second adder that adds the right signal r(n) and the error signal in opposite phase and outputs.

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

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

Abstract: An 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 invention relates to a system and method for gathering data for use in a spoken dialog system. An aspect of the invention is generally referred to as an automated hidden human that performs data collection automatically at the beginning of a conversation with a user in a spoken dialog system. The method comprises presenting an initial prompt to a user, recognizing a received user utterance using an automatic speech recognition engine and classifying the recognized user utterance using a spoken language understanding module. If the recognized user utterance is not understood or classifiable to a predetermined acceptance threshold, then the method re-prompts the user. If the recognized user utterance is not classifiable to a predetermined rejection threshold, then the method transfers the user to a human as this may imply a task-specific utterance. The received and classified user utterance is then used for training the spoken dialog system.

Abstract: Disclosed is an encoding apparatus that can efficiently encode a signal that is a broad or extra-broad band signal or the like, thereby improving the quality of a decoded signal. This encoding apparatus includes a band establishing unit (301) that generate, based on the characteristic of the input signal, band establishment information to be used for dividing the band of the input signal to establish a first band part of lower frequency side and a second band part of higher frequency side; a lower frequency encoding unit (302) for encoding, based on the band establishment information, the input signal of the first band part to generate encoded lower frequency part information; and a higher frequency encoding unit (303) for encoding, based on the band establishment information, the input signal of the second band part to generate encoded higher frequency part information.

Abstract: An audio decoder includes an arithmetic decoder for providing decoded spectral values on the basis of an arithmetically-encoded representation of the spectral values and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state described by a numeric current context value, and determines the numeric current context value in dependence on a plurality of previously-decoded spectral values. The arithmetic decoder modifies a number representation of a numeric previous context value, describing a context state associated with one or more previously decoded spectral values, in dependence on a context subregion value, to acquire a number representation of a numeric current context value describing a context state associated with one or more spectral values to be decoded. An audio encoder uses a similar concept.

Abstract: An encoding device, a decoding device, and related methods are provided that eliminate the loss of synchronization of the adaptive filters of a terminal at the encoding end and a terminal at the decoding end caused by transmission errors. Deterioration of the sound quality is suppressed when a multiple channel signal is encoded with high efficiency using an adaptive filter. In the terminal at the encoding end, a buffer stores updated filter coefficients. When packet loss detection information indicating whether there is any packet loss in the terminal at the decoding end indicates that there is packet loss, a switch outputs the past filter coefficients of the previous (NX+1) frames from the buffer to an adaptive filter. The adaptive filter uses the past filter coefficients of the previous (NX+1) frames to conduct filtering.

Abstract: A difference signal calculating unit of a noise detecting device calculates a difference between the amplitudes of a residual signal at each sample timing and a residual signal at the preceding sample timing. A difference signal comparing unit determines whether or not an impulsive noise is present on the basis of the difference signal at the current sample timing, and the difference signal at each sample timing within a predetermined duration from the current sample timing.

Abstract: Adaptive Gain Control (AGC) is performed directly in a coded domain. A Coded Domain Adaptive Gain Control (CD-AGC) system modifies at least one parameter of a first encoded signal, resulting in corresponding modified parameter(s). The CD-VQE system replaces the parameter(s) of the first encoded signal with the modified parameter(s), resulting in a second encoded signal. In a decoded state, the second encoded signal approximates a target signal that is a function of two signals, including the first encoded signal and a third encoded signal, in at least a partially decoded states. Thus, the first encoded signal does not have to go through intermediate decode/re-encode processes, which can degrade overall speech quality. Computational resources required for a complete re-encoding are not needed. Overall delay of the system is minimized. The CD-AGC system can be used in any network in which signals are communicated in a coded domain, such as a Third Generation (3G) wireless network.

Abstract: An apparatus and method of low bit rate encoding and reproducing. The method includes transforming input audio signals in a time domain into spectral signals in a frequency domain, extracting important-spectrum components from the spectral signals in the frequency domain, and quantizing the important-spectrum components, extracting residual-spectrum components other than the important-spectrum components from the spectral signals in the frequency domain, and calculating and quantizing a noise level of the residual-spectrum components, and encoding the quantized important-spectrum components and the quantized noise level losslessly, and outputting encoded bitstreams.