Abstract: The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. The upmixing can be suspended responsive to control data.

Abstract: An apparatus for decoding an encoded audio signal, includes a spectral domain audio decoder for generating a first decoded representation of a first set of first spectral portions, the decoded representation having a first spectral resolution; a parametric decoder for generating a second decoded representation of a second set of second spectral portions having a second spectral resolution being lower than the first spectral resolution; a frequency regenerator for regenerating every constructed second spectral portion having the first spectral resolution using a first spectral portion and spectral envelope information for the second spectral portion; and a spectrum time converter for converting the first decoded representation and the reconstructed second spectral portion into a time representation.

Abstract: A method and device for resampling an audio frequency signal in an audio frequency signal coding or decoding. The method includes the following acts for each signal block to be resampled: determining, by adaptive linear prediction, a number of future signal samples, this number being defined as a function of a chosen resampling delay; constructing a resampling support vector from at least samples of the current block and determined future signal samples; applying a resampling filter to the samples of the resampling support vector.

Abstract: A method of encoding audio data includes determining an energy level of a first subband of frequency domain audio data, determining a bit allocation for a coarse quantization process and a fine quantization process, determining that the energy level of the first subband of frequency domain audio data is outside a predetermined range of energy levels for the coarse quantization process, reallocating bits assigned to the fine quantization process to an extended-range coarse quantization process, the extended-range coarse quantization process using an extended range of energy levels, wherein the extended range of energy levels is larger than the predetermined range of energy levels for the coarse quantization process, and quantizing the energy level of the first subband of frequency domain audio data using the extended-range coarse quantization process to produce a quantized extended-range coarse energy level.

Abstract: A signal processing method and device includes obtaining spectral coefficients of a current frame of an audio signal, in which N sub-bands of the current frame comprises at least one of the spectral coefficients. A total energy of M successive sub-bands of the N sub-bands, a total energy of K successive sub-bands of the N sub-bands, and an energy of a first sub-band are obtained to determine whether to modify original envelope values of the M sub-bands. When the original envelope values of the M sub-bands are modified, encoding bits are allocated to each of the N sub-bands according to the modified envelope values of the M sub-bands.

Abstract: The present invention relates to a signal processing apparatus and a signal processing method, an encoder and an encoding method, a decoder and a decoding method, and a program capable of reproducing music signal having a better sound quality by expansion of frequency band. An encoder sets an interval including 16 frames as interval section to be processed, outputs high band encoded data for obtaining the high band component of an input signal and low band encoded data obtained by encoding the low band signal of the input signal for each section to be processed. In this case, for each frame, a coefficient used in estimation of the high band component is selected and the section to be processed is divided into continuous frame segments including continuous frames from which the coefficient with the same section to be processed is selected.

Abstract: An audio/speech encoding method is provided that includes transforming a time domain input signal to a frequency spectrum, and dividing the frequency spectrum to a plural of bands. The method also includes calculating a level of energies for each band, quantizing the energies for the each band, and calculating differential indices. The method additionally includes modifying a range of the differential indices for the Nth band when N is an integer of 2 or more, and replacing the differential index with the modified differential index, and not modifying a range of the differential indices for the Nth band when N is an integer of 1. The method further includes encoding the differential indices using a Huffman table selected based on a minimum value and a maximum value of the differential indices, and transmitting the encoded differential indices and a flag signal for indicating the selected Huffman table.

Abstract: Apparatus for decoding an encoded audio signal including an encoded core signal, including: a core decoder for decoding the encoded core signal to obtain a decoded core signal; a tile generator for generating one or more spectral tiles having frequencies not included in the decoded core signal using a spectral portion of the decoded core signal; and a cross-over filter for spectrally cross-over filtering the decoded core signal and a first frequency tile having frequencies extending from a gap filling frequency to an upper border frequency or for spectrally cross-over filtering a first frequency tile and a second frequency tile.

Abstract: A content system accessing an audio stream. The content system inputs segments of the audio stream into a speech classifier for classification, the speech classifier generating, for the segments of the audio stream, raw scores representing likelihoods that the respective segment of the audio stream includes an occurrence of a speech sound. The content system generates binary scores for the audio stream based on the set of raw scores, each binary score generated based on an aggregation of raw scores from consecutive series of the segments of the audio stream. The content system generates one or more timing windows for the speech sounds in the audio stream based on the binary scores, each timing window indicating an estimate of a beginning and ending timestamps of one or more speech sounds in the audio stream.

Abstract: An approach is described that obtains spectrum coefficients for a replacement frame of an audio signal. A tonal component of a spectrum of an audio signal is detected based on a peak that exists in the spectra of frames preceding a replacement frame. For the tonal component of the spectrum a spectrum coefficients for the peak and its surrounding in the spectrum of the replacement frame is predicted, and for the non-tonal component of the spectrum a non-predicted spectrum coefficient for the replacement frame or a corresponding spectrum coefficient of a frame preceding the replacement frame is used.

Abstract: The method of capturing an audio scene includes acquiring sounds having first and second directivities to obtain first and second acquisition signals, respectively, the first directivity being higher than the second directivity, the steps of acquiring being performed simultaneously, and both acquisition signals together representing the audio scene; separately storing the first and second acquisition signals or mixing individual channels in the acquisition signals to obtain first and second mixed signal, respectively, and separately storing the first and second mixed signals, or transmitting the first and second mixed signals or the first and second acquisition signals to a loudspeaker setup and rendering the first mixed signal or the first acquisition signal using a loudspeaker arrangement having a first directivity and simultaneously rendering the second mixed signal or the second acquisition signal using a loudspeaker arrangement having a second directivity, the second loudspeaker directivity being lower t

Abstract: Using various embodiments, methods and systems for assisted programming using an interconnectable block based system are described. In one embodiment, the system comprises a plurality of interconnectable blocks, and a processing unit. The processing unit can determine a programming language construct, or the portion thereof, based on a sequence in which the plurality of interconnectable blocks are connected to each other and execute the programming language construct, or the portion thereof, and transmit the output to an output device. The output device, can be coupled to the interconnectable block based system and can be at least an electroacoustic transducer device or an electroluminescent device.

Abstract: The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilizing high frequency reconstruction (HFR). It utilizes a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit.

Abstract: A method and device for resampling an audio frequency signal in an audio frequency signal coding or decoding. The method includes the following acts for each signal block to be resampled: determining, by adaptive linear prediction, a number of future signal samples, this number being defined as a function of a chosen resampling delay; constructing a resampling support vector from at least samples of the current block and determined future signal samples; applying a resampling filter to the samples of the resampling support vector.

Abstract: Audio encoder for encoding a multichannel signal is shown. The audio encoder includes a downmixer for downmixing the multichannel signal to obtain a downmix signal, a linear prediction domain core encoder for encoding the downmix signal, wherein the downmix signal has a low band and a high band, wherein the linear prediction domain core encoder is configured to apply a bandwidth extension processing for parametrically encoding the high band, a filterbank for generating a spectral representation of the multichannel signal, and a joint multichannel encoder configured to process the spectral representation including the low band and the high band of the multichannel signal to generate multichannel information.

Abstract: Several embodiments of a digital speech signal enhancer are described that use an artificial neural network that produces clean speech coding parameters based on noisy speech coding parameters as its input features. A vocoder parameter generator produces the noisy speech coding parameters from a noisy speech signal. A vocoder model generator processes the clean speech coding parameters into estimated clean speech spectral magnitudes. In one embodiment, a magnitude modifier modifies an original frequency spectrum of the noisy speech signal using the estimated clean speech spectral magnitudes, to produce an enhanced frequency spectrum, and a synthesis block converts the enhanced frequency spectrum into time domain, as an output speech sequence. Other embodiments are also described.

Abstract: At least one exemplary embodiment is directed to a method and device for voice operated control. The method can include measuring a first sound received from a first microphone, measuring a second sound received from a second microphone, detecting a spoken voice based on an analysis of measurements taken at the first and second microphone, mixing the first sound and the second sound to produce a mixed signal, and controlling the production of the mixed signal based on one or more aspects of the spoken voice.

Abstract: A method and a device for encoding a high frequency signal, and a method and a device for decoding a high frequency signal are provided, which relate to encoding and decoding technology. The method for encoding a high frequency signal includes: determining a signal type of a high frequency signal of a current frame; smoothing and scaling time envelopes of the high frequency signal of the current frame and obtaining time envelopes of the high frequency signal of the current frame that require to be encoded, if the high frequency signal of the current frame is a non-transient signal and a high frequency signal of the previous frame is a transient signal; and quantizing and encoding the time envelopes of the high frequency signal of the current frame that require to be encoded, and frequency information and signal type information of the high frequency signal of the current frame.

Abstract: An apparatus, method, and computer-readable storage medium that generate at least a cue point in a musical piece. The method includes generating a beat grid representing the musical piece, determining values for the beat grid, the values corresponding to an audio feature of the musical piece, and each value representing an entire duration of each beat in the beat grid of the musical piece, calculating a score for the audio feature at each of a plurality of positions in the beat grid of the musical piece, using some or all of the determined values, and generating the cue point at a particular position of the plurality of positions, based on the calculated scores.

Abstract: Certain embodiments enable improved encoding and decoding of a vector of coefficients by associating a vector element of a signed pyramid with an encoded value that includes a first portion and a second portion, where the first portion identifies a corresponding vector element of an unsigned pyramid and a second portion characterizes sign values for nonzero components of the vector element of the signed pyramid. As a result, computational constraints such as word size apply to the unsigned pyramid instead of the signed pyramid. The smaller size of the unsigned pyramid enables extending the range of signed pyramid parameters that are operable within the computational constraints.

Abstract: A data recording control device includes a recording preparator that generates the file which includes a file type box including information related to a type indicating a type of a player capable of reproducing the file and header information of a media data box storing media data according to the stream data, on the recording media; a box determiner that writes a movie box which includes information used for reproducing the media data, updates a box size included in the header information of the media data box to a size of immediately before the movie box, and appends header information of a free space box which stores predetermined information to a tail of the movie box, during a predetermined period; and a data writer that appends the media data according to the stream data to a tail of the header information of the free space box.

Abstract: The invention provides a decoder being configured for processing an encoded audio bitstream, wherein the decoder includes: a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal includes at least one decoded frame; a noise estimation device configured to produce a noise estimation signal containing an estimation of the level and/or the spectral shape of a noise in the decoded audio signal; a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to obtain an audio output signal.

Abstract: An apparatus for decoding an encoded audio signal having an encoded representation of a first set of first spectral portions and an encoded representation of parametric data indicating spectral energies for a second set of second spectral portions, has: an audio decoder for decoding the encoded representation of the first set of the first spectral portions to obtain a first set of first spectral portions and for decoding the encoded representation of the parametric data to obtain a decoded parametric data for the second set of second spectral portions indicating, for individual reconstruction bands, individual energies; a frequency regenerator for reconstructing spectral values in a reconstruction band having a second spectral portion using a first spectral portion of the first set of the first spectral portions and an individual energy for the reconstruction band, the reconstruction band having a first spectral portion and the second spectral portion.

Abstract: Methods and apparatuses are described in which an unlicensed spectrum is used for Long Term Evolution (LTE) communications. One method includes performing a clear channel assessment (CCA) for an unlicensed spectrum in a current gating interval to determine whether the unlicensed spectrum is available for a transmission in a next transmission interval, and gating OFF the transmission in the unlicensed spectrum for the next transmission interval when the determination is that the unlicensed spectrum is unavailable.

Abstract: A sound signal processing method according to an embodiment includes a step of acquiring an input sound signal, a step of acquiring a beat number per unit time period from the input sound signal, a step of normalizing the input sound signal with the beat number per unit time period, a step of calculating a beat spectrum of the normalized input sound signal, and a step of calculating a rhythm similarity between the beat spectrum of the normalized input sound signal and a normalized beat spectrum calculated from a reference sound signal.

Abstract: In accordance with an embodiment, a method is disclosed for detecting action in a video clip. An audio clip is extracted from the video clip. The audio clip is converted to an auditory spectrogram. The auditory spectrogram is used to construct a self-similarity matrix. The self-similarity matrix is then used to calculate a novelty curve. The clip is then segmented into segments according to peaks in the novelty curve. Each of segments is scored, and then classified as an action clip if the score is above or below a predetermined threshold. Related methods for folding a digitized song into a shorter version of itself and for sequencing a set of user-supplied video and photo clips to a user-supplied song are further disclosed.

Abstract: A hearing device for enhancing speech intelligibility, the hearing device includes: an input transducer for providing an input signal comprising a speech signal and a noise signal; a processing unit; an acoustic output transducer coupled to the processing unit, the acoustic output transducer configured to provide an audio output signal based on an output signal form the processing unit; wherein the processing unit is configured to determine one or more parameters of the input signal based on a codebook based approach (CBA) processing; and wherein the processing unit is configured to perform a Kalman filtering of the input signal based on the determined one or more parameters so that the output signal has an enhanced speech intelligibility.

Abstract: An encoding device according to the disclosure includes a first encoding unit that generates a first encoded signal in which a low-band signal having a frequency lower than or equal to a predetermined frequency from a voice or audio input signal is encoded, and a low-band decoded signal; a second encoding unit that encodes, on the basis of the low-band decoded signal, a high-band signal having a band higher than that of the low-band signal to generate a high-band encoded signal; and a first multiplexing unit that multiplexes the first encoded signal and the high-band encoded signal to generate and output an encoded signal. The second encoding unit calculates an energy ratio between a high-band noise component, which is a noise component of the high-band signal, and a high-band non-tonal component of a high-band decoded signal generated from the low-band decoded signal and outputs the ratio as the high-band encoded signal.

Abstract: A method is described that estimates noise in an audio signal. An energy value for the audio signal is estimated and converted into the logarithmic domain. A noise level for the audio signal is estimated based on the converted energy value.

Abstract: An audio watermarking system conveys information using an audio channel by modulating an audio signal to produce a modulated signal by embedding additional information into the audio signal. Modulating the audio signal includes segmenting the audio signal into overlapping time segments using a non-rectangular analysis window function produce a windowed audio signal, processing the windowed audio signal for a time segment to produce frequency coefficients representing the windowed time segment and having phase values and magnitude values, selecting one or more of the frequency coefficients, modifying phase values of the selected frequency coefficients using the additional information to map the phase values onto a known phase constellation, and processing the frequency coefficients including the modified phase values to produce the modulated signal.

Abstract: In general, techniques are described for specifying audio rendering information in a bitstream. A device configured to generate the bitstream may perform various aspects of the techniques. The bitstream generation device may comprise one or more processors configured to specify audio rendering information that includes a signal value identifying an audio renderer used when generating the multi-channel audio content. A device configured to render multi-channel audio content from a bitstream may also perform various aspects of the techniques. The rendering device may comprise one or more processors configured to determine audio rendering information that includes a signal value identifying an audio renderer used when generating the multi-channel audio content, and render a plurality of speaker feeds based on the audio rendering information.

Abstract: A music information generating device, including: a block color name recognizing part which selects and determines, from among plural color names which are made to correspond to plural ranges of color attribute value having been set beforehand, a color name corresponding to the representative-color of the block as a block color name of the block, by recognizing to which range of the color attribute value the representative-color of the block belongs, where the plural color names have been made or are going to be made to correspond to the plural sound-source names having been stored; and a diagram music-score generating unit which selects, based on a criterion having been set beforehand, a designated block from among the blocks having been arranged in lattice shape, and thereby generates a diagram music-score.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding a stereo audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and a discrete representation of the stereo audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

Abstract: A device includes an encoder. The encoder is configured to receive two audio channels. The encoder is also configured to determine a mismatch value indicative of an amount of a temporal mismatch between the two audio channels. The encoder is further configured to determine, based on the mismatch value, at least one of a target channel or a reference channel. The target channel corresponds to a lagging audio channel of the two audio channels and the reference channel corresponds to a leading audio channel of the two audio channels. The encoder is also configured to generate a modified target channel by adjusting the target channel based on the offset value. The encoder is further configured to generate at least one encoded channel based on the reference channel and the modified target channel.

Abstract: An image processing apparatus comprising an acquiring unit configured to acquire an encoding target block having a plurality of groups each including a predetermined number of pixels, a deciding unit configured to decide for each group a quantization parameter used to quantize image data of the group and an encoding scheme so that a code length of the encoding target block does not exceed a predetermined value, wherein the deciding unit selects, as the encoding scheme of a respective group, either of a first encoding scheme that outputs quantized image data and a second encoding scheme that outputs encoded data of a differential between quantized image data and prediction data, and an encoding unit configured to generate encoded data by encoding image data of the encoding target block in accordance with the quantization parameters and the encoding schemes decided for the respective groups.

Abstract: The present disclosure provide a signal processing method and apparatus. The method includes: determining a total quantity of to-be-allocated bits corresponding to a current frame; implementing primary bit allocation on to-be-processed sub-bands; performing a primary information unit quantity determining operation for each sub-band that has undergone the primary bit allocation; selecting sub-bands for secondary bit allocation from the to-be-processed sub-bands according to at least one of a sub-band characteristic of each sub-band of the to-be-processed sub-bands or the total quantity of surplus bits; implementing secondary bit allocation on the sub-bands for secondary bit allocation; and performing a secondary information unit quantity determining operation for each sub-band of the sub-bands for secondary bit allocation.

Abstract: A system configured to reconstruct audio signals. The system may identify missing audio samples due to packet loss or detect distortion caused by audio clipping and may reconstruct the audio data. The system may employ a forward-looking neural network that recursively predicts audio samples based on previous audio samples and/or a backward-looking neural network that recursively predicts audio samples based on subsequent audio samples. The system may generate audio data using only the forward-looking neural network for low latency applications or may generate audio data using both neural networks for mid to high latency applications. To reduce distortion in output audio data, the system may generate the audio data by cross-fading between outputs of the neural networks and/or may cross-fade between the generated audio data and the input audio data.

Abstract: A method, a system and a computer program product are disclosed for enhancing an audio signal in relation to a hearing impairment. An input signal is obtained comprising input sub-band signals in a frequency range comprising a source range and a target range. The input sub-band signals in the source range are selectively transposed into transposed sub-band signals in the target range according to a predefined transposing rule. A masking threshold is determined based on a predefined perceptual model and perceptually relevant sub-band signals of the transposed sub-band signals in the target range exceeding the masking threshold are detected. Input sub-band signals in the target range are selectively replaced with corresponding detected perceptually relevant sub-band signals of the transposed sub-band signals in the target range.

Abstract: Provided are a method and apparatus for encoding and decoding a high frequency signal by using a low frequency signal. The high frequency signal can be encoded by extracting a coefficient by linear predicting a high frequency signal, and encoding the coefficient, generating a signal by using the extracted coefficient and a low frequency signal, and encoding the high frequency signal by calculating a ratio between the high frequency signal and an energy value of the generated signal. Also, the high frequency signal can be decoded by decoding a coefficient, which is extracted by linear predicting a high frequency signal, and a low frequency signal, and generating a signal by using the decoded coefficient and the decoded low frequency signal, and adjusting the generated signal by decoding a ratio between the generated signal and an energy value of the high frequency signal.

Abstract: A coding efficiency of coding spectral coefficients of a spectrum of an audio signal is increased by en/decoding a currently to be en/decoded spectral coefficient by entropy en/decoding and, in doing so, performing the entropy en/decoding depending, in a context-adaptive manner, on a previously en/decoded spectral coefficient, while adjusting a relative spectral distance between the previously en/decoded spectral coefficient and the currently en/decoded spectral coefficient depending on an information concerning a shape of the spectrum.

Abstract: The present disclosure provides methods, devices and computer program products for non-uniform quantization of parameters relating to parametric spatial coding of audio signals. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system taking the non-uniformly quantized parameters into account. According to the disclosure, such an approach renders it possible to reduce bit consumption without substantially reducing the quality of the reconstructed audio object.

Abstract: Methods for processing a multichannel audio signal that includes transient noise signals are provided. The method includes buffering the multichannel audio signal in a subband domain, and estimating the subband frames for transient noise likelihood. A probability of transient noise for the buffered subband frames is determined and a multichannel spatial filter is applied to decompose the subband frames to transient attenuated target source and noise estimation cancelled of the target source signal. A spectral filter is applied to the target source frame to enhance the target source frame and the subband frames that are determined to have a probability of the transient noise greater than a first threshold and a probability of target source less than a second threshold are muted.

Abstract: A method for generating a psychoacoustic model from an audio signal transforms a block of samples of an audio signal into a frequency spectrum comprising frequency components. From this frequency spectrum, it derives group masking energies. These group masking energies each correspond to a group of neighboring frequency components in the frequency spectrum. For a group of frequency components, the method allocates the group masking energy to the frequency components in the group in proportion to energy of the frequency components within the group to provide adapted mask energies for the frequency components within the group, the adapted mask energies providing masking thresholds for the psychoacoustic model of the audio signal.

Abstract: Multiple audio files may be synchronized using energy vectors produced from energy portions of individual frequency energy representations. Individual energy samples and time values of individual energy vectors may be compared using a multi-resolution framework to correlate energy samples and time values of multiple audio tracks to one another.

Abstract: A content server accessing an audio stream, and inputs portions of the audio stream into one or more non-speech classifiers for classification, the non-speech classifiers generating, for portions of the audio stream, a set of raw scores representing likelihoods that the respective portion of the audio stream includes an occurrence of a particular class of non-speech sounds associated with each of the non-speech classifiers. The content server generates binary scores for the sets of raw scores, the binary scores generated based on a smoothing of a respective set of raw scores. The content server applies a set of non-speech captions to portions of the audio stream in time, each of the sets of non-speech captions based on a different one of the set binary scores of the corresponding portion of the audio stream.

Abstract: An image processing apparatus comprising an acquiring unit configured to acquire an encoding target block having a plurality of groups each including a predetermined number of pixels, a deciding unit configured to decide, for each group, a quantization parameter used to quantize image data of the group and an encoding scheme so that a code length of the encoding target block does not exceed a predetermined value corresponding to a target code amount of the encoding target block, and an encoding unit configured to generate encoded data by encoding image data of the encoding target block in accordance with the quantization parameters and the encoding schemes decided for the respective groups by the deciding unit.

Abstract: The present application discloses a pulse encoding method for use by a voice encoder. An initial number of coding bits is determined according to number of pulses in an input signal. An index value is calculated according to information of the pulses in the input signal. The index value is compared with an adjustment threshold value. If the index value is smaller than the adjustment threshold value, the index value is encoded using a reduced number of encoding bits. If the index value is not smaller than the adjustment threshold value, the index value plus an offset value is encoded using the initial number of encoding bits. The reduced number of encoding bits is smaller than the initial number of encoding bits, and the offset value is greater than or equal to the adjustment threshold value.

Abstract: Presented herein are excitation modeling and matching sound coding techniques that are configured to account for a recipient's estimated or predetermined neural response so as to improve a recipient's ability to perceive sound. More specifically, the techniques presented herein correlate a modeled excitation pattern with a target excitation pattern to generate a pattern of stimulation current configured to evoke an excitation pattern within the recipient's auditory system that approximates the target excitation pattern.

Abstract: A Unified Speech and Audio Codec (USAC) that may process a window sequence based on mode switching is provided. The USAC may perform encoding or decoding by overlapping between frames based on a folding point when mode switching occurs. The USAC may process different window sequences for each situation to perform encoding or decoding, and thereby may improve a coding efficiency.

Abstract: Provided is a method and apparatus for encoding or decoding a signal corresponding to a high frequency band in an audio signal. The method and apparatus for encoding a high frequency band detects and encodes frequency component(s) according to a pre-set criterion from a signal corresponding to a frequency band higher than a pre-set frequency and encodes energy value(s) of a signal to reconstruct band(s) in which the detected frequency component(s) are included. The method and apparatus for decoding a high frequency band decodes the signal by adjusting a signal to reconstruct a band in which important frequency component(s) are included by considering an energy value of the important frequency component(s). Accordingly, even though encoding or decoding is performed using a small number of bits, there is no degradation in sound quality of a signal corresponding to a high frequency band, and thus coding efficiency can be maximized.