Abstract: An audio processing method utilized to generate an audio stream. An audio frame includes N frequency subbands. An Ith frequency subband among the N frequency subbands includes M audio samples and has an Ith psychoacoustic masking value. First, an Ith offset of the Ith frequency subband is calculated. Then, the Ith psychoacoustic masking value and the Ith offset are inputted into a projection formula to generate an Ith projection value. According to the Ith projection value and a limit range, an Ith scale factor is determined. Subsequently, the M audio samples in the Ith frequency subband are adjusted according to the Ith scale factor.

Abstract: The apparatus for constructing a multi-channel output signal using an input signal and parametric side information, the input signal including the first input channel and the second input channel derived from an original multi-channel signal, and the parametric side information describing interrelations between channels of the multi-channel original signal uses base channels for synthesizing first and second output channels on one side of an assumed listener position, which are different from each other. The base channels are different from each other because of a coherence measure. Coherence between the base channels (for example the left and the left surround reconstructed channel) is reduced by calculating a base channel for one of those channels by a combination of the input channels, the combination being determined by the coherence measure. Thus, a high subjective quality of the reconstruction can be obtained because of an approximated original front/back coherence.

Abstract: A digital audio decoder for receiving an encoded audio signal and decoding the audio signal. The digital audio decoder uses both real time computations and calculations pre-stored in a look-up table to decode the encoded audio signal. The digital audio decoder uses fixed point arithmetic but a variable word format to represent the intermediate computations and pre-stored look-up table entries.

Abstract: A device in a subscriber television system receives a stream of frames of information, and each frame of information is carried in at least one network packet that is formatted according to a first protocol. Each network packet carries an application packet that is formatted according to a second protocol, which is a protocol for an application. The device includes application awareness that enables the device to selectively modify the application packets.

Abstract: An apparatus for generating a multi-channel output signal performs a center channel cancellation to obtain improved base channels for reconstructing left-side output channels or right-side output channels. In particular, the apparatus includes a cancellation channel calculator for calculating a cancellation channel using information related to the original center channel available at the decoder. The device furthermore includes a combiner for combining a transmission channel with the cancellation channel. Finally, the apparatus includes a reconstructor for generating the multi-channel output signal. Due to the center channel cancellation, the channel reconstructor not only uses a different base channel for reconstructing the center channel but also uses base channels different from the transmission channels for reconstructing left and right output channels which have a reduced or even completely cancelled influence of the original center channel.

Abstract: The transient problem may be sufficiently addressed, and for this purpose, a further delay on the side of the decoding may be reduced if a new SBR frame class is used wherein the frame boundaries are not shifted, i.e. the grid boundaries are still synchronized with the frame boundaries, but wherein a transient position indication is additionally used as a syntax element so as to be used, on the encoder and/or decoder sides, within the frames of these new frame class for determining the grid boundaries within these frames.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: An information encoding apparatus and method, an information decoding apparatus and method, a recording medium, and a program make it possible to embed information without causing an increase in the amount of data of compressed audio signals. A spectrum measurer converts an audio signal from a time axis signal into a frequency axis signal. A compressor removes frequency spectra having levels below a minimum audible threshold value. An embedability determiner determines whether information can be embedded. A spectrum switcher embeds information in an audio signal by switching or by not switching predetermined first frequency spectrum and second frequency spectrum under the control of an information adder. The present invention is applicable to a disk recording apparatus.

Abstract: A coding device capable of improving the coding efficiency and a decoding device for decoding a code sequence generated by the coding device are provided. In the coding device, for each of the possible block combinations obtained when dividing a frame, a coding unit encodes each block in the frame block by block at different bit rates, and at the same time, the coding unit decodes the resultant code sequences related to the frame. A calculation unit calculates the error powers of the decoded signals and the input signal. A determination unit selects a code sequence that makes the average bit rate in coding the frame not higher than a specified value and the corresponding error power a minimum. This selected code sequence is output.

Abstract: A method of parametrically encoding a transient audio signal, including the steps of: determining a set V of the N largest frequency components of the transient audio signal, where N is a predetermined number; determining an approximate envelope of the transient audio signal; and determining a predetermined number P of samples W of the approximate envelope for use in generating a spline approximation of the approximate envelope, whereby a parametric representation of the transient audio signal is given by parameters including V, N, P and W, such that a decoder receiving the parametric representation can reproduce a received approximation of the transient audio signal.

Abstract: A sound encoder having an improved quantization performance while suppressing an increase of the bit rate to a lowest level.

Abstract: This invention produces data packets that can vary in length and/or data compression ratio. First, an algorithm is employed to transform a data signal into fixed or variable length data packets at variable data compression ratios. If the algorithm produces fixed length data packets, the fixed length data packets are then converted to variable length data packets, which include only the valid data bytes of the fixed length data packets. Finally, the variable compression ratio, variable length data packets are provided with length codes at each end of each data packet to facilitate bi-directional searching and decompression. The transition from fixed to variable length data packets employs a buffer which stores the fixed length data packets until the fixed length data packets are converted to variable length data packets.

Abstract: Many compressed audio or video frames contain silence (if audio), or a blank image (if video); these essentially information content free (e.g. silent if audio or blank if video) frames can be both detected whilst still in compressed form and then used to carry the additional data. In an MPEG implementation, subbands associated with silent frames are rendered digitally silent and then used to carry PAD (Programme Associated Data).

Abstract: A method that allows two or more APs (application programs) to use a sound card and an AP to select an input device of the sound card regardless of a mixer device of the sound card is disclosed. A connecting module 31 is disposed among a sound card 17, a HDD 13, and an NIC 19. Switches 32 and 33 are disposed in parallel to the connecting module 31. In other words, an audio signal that is input form the sound card 17 is supplied to the NIC 19 through the switch 32. In addition, the audio signal is supplied to the HDD 13 through the switch 33. The switch 32 is equivalent to a concept of an application interface for starting and stopping a sending operation for an audio signal supplied from the sound card 17 to the network 111. The switch 33 is equivalent to a concept of an application interface for starting and stopping a saving operation for an audio signal supplied from the sound card 17 to the HDD 13.

Abstract: A mapping transform unit subjects input audio signals to a mapping transform and generates frequency region signals that take frequency as a variable; a code amount designation unit supplies a preset coding bit rate as a code amount output; a frequency region signal compression encoder, based on the code amount, subjects input frequency region signals to a compression encoding process and generates a bitstream; and a bandwidth-limiting unit executes a bandwidth-limiting processing in which a part of the frequency zone covered by frequency region signals is allotted to an attenuation frequency zone, and in which the value of the frequency region signal is multiplied by an attenuation coefficient having a value less than 1 in the attenuation frequency zone to attenuate the frequency region signal in the attenuation frequency zone, and supplies the frequency region signals that have undergone the bandwidth-limiting processing to the frequency region signal compression encoder.

Abstract: A method and an apparatus analogically output low-resolution digital signals to achieve an equal analog output result of a high-resolution digital signal under a request of signal quality. The low-resolution digital signals are compensatively output multiple times to gain the energy thereof equal to the energy output by the high-resolution digital signal. Therefore, a digital-to-analog conversion with fewer bits satisfies a higher demand for accuracy generally achieved by a digital-to-analog conversion with more bits.

Abstract: A technique is described for use in a decoder configured to decode a series of frames representing an encoded audio signal. The technique is for transitioning between a lost frame and one or more received frames following the lost frame in the series of frames. In accordance with the technique, an output audio signal associated with the lost frame is synthesized. An extrapolated signal is generated based on the synthesized output audio signal. A time lag is calculated between the extrapolated signal and a decoded audio signal associated with the received frame(s), wherein the time lag represents a phase difference between the extrapolated signal and the decoded audio signal. The decoded audio signal is time-warped based on the time lag, wherein time-warping the decoded audio signal comprises stretching or shrinking the decoded audio signal in the time domain.

Abstract: Systems and methods are described for performing packet loss concealment using an extrapolation of an excitation waveform in a sub-band predictive speech coder, such as an ITU-T Recommendation G.722 wideband speech coder. The systems and methods are useful for concealing the quality-degrading effects of packet loss in a sub-band predictive coder and address some sub-band architectural issues when applying excitation extrapolation techniques to such sub-band predictive coders.

Abstract: An L-channel signal and an R-channel signal are input into a subtractor in which the difference between the L-channel signal and the R-channel signal is determined. The resulting signal is input into first and second multipliers. In the first multiplier, a mixed output signal is generated by using the channel mixture ratio calculated in a previous frame. In the second multiplier, a mixed output signal is generated by using the channel mixture ratio calculated in a current frame. The mixed output signals from the first and second multipliers respectively undergo spectrum transform in first and second spectrum transformers, and quantized in first and second quantizers. A comparator compares a quantization error in the first quantizer with that in the second quantizer. If the quantization error in the second quantizer is greater than that in the first quantizer by a predetermined factor, the comparator controls a switch to select the output of the first quantizer.

Abstract: A method and apparatus for performing audio post processing using mode information that indicates the degree of similarity between a right channel signal and a left channel signal in MPEG-1 layer 3 audio data, are provided. If the difference between the two channel signals is small, a first mode is used in which the high frequency component of only one channel is recovered and the recovered high frequency component is used to recover the high frequency component of the other channel. If the difference between the two channel signals is large, a second mode is selected in which the high frequency component in only one of every two frames is recovered alternately in the left channel and the right channel and the high frequency component of each of the skipped frames is interpolated based on the high frequency components of the previous frame and the next frame.

Abstract: A header compression/decompression apparatus that improves the throughput of an overall multilayer protocol stack at a network node. In this apparatus, an encoding section 106 compresses multilayer header information included in a protocol data unit on a multilayer protocol stack 101. A session context ID manager 112 generates a session context ID 401 which is formed by integrating information on compression of multilayer header information by the encoding section 106 and, for example, information for identifying a scheme for compressing multilayer header information.

Abstract: An encoding device includes a transforming unit operable to extract a part of an inputted audio signal at predetermined time intervals and to transform each extracted part to produce a plurality of windows composed of short blocks, and a judging unit operable to compare the windows with one another to judge whether there is a similarity of a predetermined degree and to replace a high frequency part of a first window, which is one of the produced windows, with values “0” when there is the similarity, wherein the first window and a second window share a high frequency part of the second window, which is also one of the produced windows. The encoding device also includes a first quantizing unit operable to quantize the produced windows after replacing operation; a first encoding unit operable to encode the quantized windows to produce encoded data; and a stream output unit operable to output the produced encoded data.

Abstract: A method for transferring real time information on a record carrier, typically bitstream audio on an optical disc, which method comprises encoding consecutive segments of the real time information to compressed real time data in frames, and determining a buffer occupancy for at least one frame, which buffer occupancy is indicative of an amount of compressed real time data to be present in the playback buffer at the start of decoding said frame. A signal is transmitted carrying the compressed real time data and the buffer occupancy, which data are received, stored in a playback buffer and finally decoded. The retrieving and/or the decoding is controlled in dependence on said transferred buffer occupancy. A playback buffer can be used effectively without risk for underflow or overflow. Also a method for recording audio information on a record carrier, a recording device, a record carrier and a playback device are described.

Abstract: A method, medium, and apparatus encoding and/or decoding an audio signal to surround data. While encoding spatial information, which can up-mix an audio signal to a surround signal, to extension data, a length of a payload corresponding to the spatial information is encoded and a payload of the spatial information is decoded using the length of the payload. Accordingly, compatibility of the spatial information can be provided, and the spatial information can be transmitted by effectively embedding the spatial information.

Abstract: According to the present invention, multiple parametrically encoded audio signals can be efficiently combined using an audio signal generator, which generates an audio output signal by combining the down-mix channels and the associated parameters of the audio signals directly within the parameter domain, i.e. without reconstructing or decoding the individual input audio signals prior to the generation of the audio output signal. This is achieved by direct mixing of the associated down-mix channels of the individual input signals. It is one key feature of the present invention that the combination of the down-mix channels is achieved by simple, computationally inexpensive arithmetic operations.

Abstract: The invention comprises the retrieval of a quantized spectrum of audio-data, which had been compressed conforming to psychoacoustic principles. This spectrum can be identified with the aid of sorted (and logarithmized) differences of the magnitude of the amplitude. A problem to be solved by the invention is to find the basis of sensitive indicators for the degree of quantization of a spectrum. The result of this calculation shows for a quantized spectrum several small values at amplitudes of the same quantization level, and obvious spikes at positions, where the amplitudes have reached a higher level of quantization, while a non-quantized spectrum typically resembles noise.

Abstract: An audio encoder discards spectral components of an input signal and uses channel coupling to reduce the information capacity requirements of an encoded signal. Channel coupling represents selected spectral components of multiple channels of signals in a composite form. An audio decoder synthesizes spectral components to replace the discarded spectral components and generates spectral components for individual channel signals from the coupled-channel signal. The encoder provides scale factors in the encoded signal that improve the efficiency of the decoder to generate output signals that substantially preserve the spectral energy of the original input signals.

Abstract: An encoding device (100) includes (i) a first encoding unit (132) that encodes spectral data in the lower frequency band represented by a plularity of parameters, out of the spectral data obtained by transforming an audio signal inputted for a fixed time length, (ii) a second quantizing unit (133) that generates sub information representing characteristics of the spectral data in the higher frequency by fewer parameters than those for the lower frequency band, out of the spectral data obtained by the transformation, (iii) a second encoding unit (134) that encodes the generated sub information, and (iv) a stream output unit (140) that outputs the data encoded by the first encoding unit (132) and the data encoded by the second encoding unit (134).

Abstract: Windows of the first type and windows of the second type are identified within a frame using energy associated with each short window within the frame. The short windows of the first type and the short windows of the second type are then grouped into two preliminary groups based on the window type of each short window. Further, if the number of short windows in any of the two preliminary groups exceeds a threshold number, the short windows in this large preliminary group are further grouped into at least two more groups.

Abstract: An integrated digital BTSC encoder substantially implemented on a single CMOS integrated circuit is described. By saturating and adding offsets to the value of the feedback control signal in regions of operation where the calculation for the control signal is dominated by noise, an integrated circuit selectively adjusts the control signal for the spectral or wideband feedback loop, as required, so that the performance of a very low noise digital BTSC encoder complies with the requirements of the BTSC standard.

Abstract: In a telecommunications system, an arithmetic logic unit (ALU) that receives an input signal. The input signal includes a digital signal representative of an analog signal. The ALU selectively performs compression and decompression on the digital signal. The ALU comprises the following elements. A standard ALU component performs standard ALU operations on the input signal. An encoding unit selectively performs compression on the digital signal. A decoding unit selectively performs decompression on the digital signal. An instruction decoder receives and decodes an ALU instruction. An output selector selects a result from one of the standard ALU component, the encoding unit, and the decoding unit in accordance with the decoded instruction and provides the result as an output.

Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.

Abstract: An encoding device (200) includes: a time characteristic extracting unit (203) that specifies a band for a part of a frequency spectrum based on a characteristic of an audio input signal in a time domain; a time transforming unit (204) that transforms a signal in the specified band to a signal according to frequency-time transform; and an encoded data stream generating unit (205) that encodes the signal obtained by the time transforming unit (204) and at least a part of the frequency spectrum, and generates an output encoded data stream from the encoded signal and the encoded frequency spectrum.

Abstract: An encoding device (200) is comprised of a band dividing unit (201) that divides an input signal (207) into a low frequency signal (208) representing a signal in the lower frequency band and a high frequency signal (209) representing a signal in the higher frequency band, a lower frequency band encoding unit (202) that encodes the low frequency signal (208) and generates a low frequency code (213), a similarity judging unit (203) that judges similarity between the high frequency signal (209) and the low frequency signal (208) and generates switching information (210), “n” higher frequency band encoding units 205 that encode the high frequency signal (209) through respective encoding methods and generate a high frequency code (212), a switching unit (204) that selects one of the higher frequency band encoding units (205) and has the selected higher frequency band encoding unit (205) perform encoding, and a code multiplexing unit (206) that multiplexes the low frequency code (213), the high frequency code (212)

Abstract: An audio processing unit that provides a high-sound-quality reproduction even when a band property of an input encoded signal drops at or below a Nyquist frequency. The audio processing unit has a band-variable bandpass filter 1500; a band determining section 1700 for determining a passband of the band-variable bandpass filter with respect to a band extension component from an extended band generating section 1300 by using decoding information from a decode-processing section 1100 as a band determination information; and a bandpass filter controller 1600 for controlling the passband of the band-variable bandpass filter in accordance with an indication from the band determining section.

Abstract: A system for transmitting audio signals over a telecommunications link generates the signals as two or more alternative feeds, for example at different data rates. The two feeds are encoded using coding methods having a frame structure with different frame lengths. To facilitate switching between the two, the input signal is notionally divided into temporal portions and each is coded by taking it, plus enough of the next (or preceding) portion to make up a whole number of frames, and encoding it, whereby the encoded portions overlap—at least for one of the feeds. The overlap is lost upon decoding by discarding duplicate material.

Abstract: A system and method for compressing data into a single integrated strand of data is disclosed. The system includes an encoder, sequencer, compression engine, protocol stack and a broadcaster for use over existing networks and wireless systems. Using compression and encoding techniques, multiple strands of data are taken and compressed into a single integrated continuous or block stream. The data comes from multiple sources, such video, audio, or data, such text.

Abstract: A compressed-code generating method that is used for compressing information on characters including numerical data, sound and images, and a compressed-code expanding method that is used for restoring and expanding the compressed code generated by using the compressed-code generating method to the original information. Bit strings {y}1 and {y}2 are obtained respectively from a bit string {y} of information to be compressed. A reversible loop that exists in chaos is operated to these obtained bit strings, thereby to execute a reversible compression/expansion of the information using the chaos.

Abstract: An efficient coding scheme with higher audio bandwidth and/or better audio quality at lower bitrates, wherein the scheme eliminates long-term and short-term frequency domain correlation in a signal via frequency domain predictors. The coding scheme compresses information consisting of coded low frequency components as well as a parametric representation for the high frequency components based on a non-linear model. Additionally, by working on the frequency domain representations of the signal (such as the MDCT representation which is naturally available to a PAC encoder and decoder), low pass and high pass signal components are easily obtained by windowing the appropriate ranges of frequencies in the signal. Furthermore, the power functions of the signal are replaced by corresponding convolution functions of the same order.

Abstract: A device and a method for compressing signal information by removing (thinning out) the signal component of a signal in a specific frequency band. Firstly, an input time-series signal (e.g., a PCM signal) is converted by an analyzer (11) into a spectrum signal. Next, of the bands obtained by dividing the spectrum equally into bands, the band having a predetermined or higher correlation in the spectrum distribution with the lower frequency band is specified as a harmonic band by a frequency band masking unit (12). Then, a removal band from which the spectrum is to be removed is determined from the harmonic band, and the spectrum signal of this removal band, from which the spectrum component has been removed (namely the frequency component has been thinned out), is fed to a synthesizer (13).

Abstract: A method and apparatus for audio compression receives an audio signal. Transform coding is applied to the audio signal to generate a sequence of transform frequency coefficients. The sequence of transform frequency coefficients is partitioned into a plurality of non-uniform width frequency ranges and then zero value frequency coefficients are inserted at the boundaries of the non-uniform width frequency ranges. As a result, certain of the transform frequency coefficients that represent high frequencies are dropped.

Abstract: The invention relates to a method in the decompression of a compressed audio signal. In the decompression, a predicting coding has been used, wherein samples taken of the audio signal have been formed into frames, and the samples of the frames have been compared with past samples to find out the prediction error. In the method, frames (frn, frn?1, frn?2) of the compressed audio signal are stored, and a predicting decoding is used to decompress the audio signal compressed with the predicting coding, on the basis of said stored frames (frn, frn?1, frn?2). In the method, at least one memory pointer is used to indicate the storage location of the frames (frn, frn?1, frn?2). Said memory pointers (P1, P2) are used to point to the storage location of the frame (frn?1) preceding the frame (frn) being processed at the time, and to the storage location of the frame (frn?2) preceding said past frame (frn?1).

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: Digital signal samples X in a floating-point format, each of which is composed of 1 bit of sign, 8 bits of exponent E and 23 bits of mantissa M, are converted through rounding by an integer formatting part 12 into digital signal samples Y in an integer format, the sequence of the digital signal samples Y is losslessly compression-coded by a compressing part 13 into a code sequence Ca, and the code sequence Ca is output. The digital signal samples Y are converted by a floating point formatting part 15 into digital signal samples X? in the floating-point format, a difference signal ?X indicating the difference between the digital signal sample X? and the digital signal sample X is determined by a subtraction part 16, the difference signal ?X is losslessly coded, and the resulting code sequence Cb is output.

Abstract: A data processing apparatus for data processing an audio signal includes an input terminal (1) for receiving the audio signal, a 1-bit A/D converter (4) for A/D converting the audio signal to for a bitstream signal, a prediction unit (10) for carrying out a prediction step on the bitstream signal to form a predicted bitstream signal, a signal combination unit (42) for combining the bitstream signal and the predicted bitstream signal to form a residue bitstream signal, and an output terminal (14) for supplying the residual bitstream signal.

Abstract: In an audio signal encoding apparatus, a first audio signal and a second audio signal are added into an addition-result signal. The first audio signal is subtracted from the second audio signal to generate a subtraction-result signal. A first difference signal is generated which represents a difference in the addition-result signal. A second difference signal is generated which represents a difference in the subtraction-result signal. A plurality of first predictors have different prediction characteristics respectively, and are responsive to the first difference signal for generating first different prediction signals for the first difference signal, respectively. A plurality of first subtracters operate for generating first prediction-error signals representing differences between the first difference signal and the first different prediction signals, respectively. A first minimum prediction-error signal representative of a smallest difference is selected from among the first prediction-error signals.

Abstract: Sound signal is received which contains sound characteristics to be represented in musical notation. The characteristics, such as a volume level of the sound signal, are extracted out of the received sound signal, and various parameters for use in subsequent analysis of the sound signal are set in accordance with the extracted characteristics. Also, a desired scale determining condition is set by a user. Pitch of the sound signal is determined using the thus-set parameters. The determined pitch is rounded to any one of scale notes, corresponding to the user-set scale determining condition. Also, a given unit note length is set as a predetermined criterion or reference for determining a note length, and a length of the scale note determined from the received sound signal is determined using the thus-set unit note length as a minimum determination unit, i.e., with an accuracy of the unit note length.

Abstract: In an audio signal encoding apparatus, a first audio signal and a second audio signal are added into an addition-result signal. The first audio signal is subtracted from the second audio signal to generate a subtraction-result signal. A first difference signal is generated which represents a difference in the addition-result signal. A second difference signal is generated which represents a difference in the subtraction-result signal. A plurality of first predictors have different prediction characteristics respectively, and are responsive to the first difference signal for generating first different prediction signals for the first difference signal, respectively. A plurality of first subtracters operate for generating first prediction-error signals representing differences between the first difference signal and the first different prediction signals, respectively. A first minimum prediction-error signal representative of a smallest difference is selected from among the first prediction-error signals.

Abstract: A sound separation apparatus for separating a target signal from a mixed input signal, wherein the mixed input signal includes the target signal and one or more sound signals emitted from different sound sources. The sound separation apparatus according comprises a frequency analyzer for performing a frequency analysis on the mixed input signal and calculating spectrum and frequency component candidate points at each time. The apparatus further comprises feature extraction means for extracting feature parameters which are estimated to correspond with the target signal, comprising a local layer for analyzing local feature parameters using the spectrum and the frequency component candidate points and one or more global layers for analyzing global feature parameters using the feature parameters extracted by the local layer. The apparatus further comprises a signal regenerator for regenerating a waveform of the target signal using the feature parameters extracted by the feature extraction means.

Abstract: An audio decoding device is provided for decoding NA (where NA>1) channels of audio signals by a sub-band synthesis operation using sub-band synthesis filter data and sub-band signal data. The decoding device includes a first memory section for storing MA (where MA<NA) channels of the sub-band synthesis filter data and the sub-band signal data, a second memory section for storing at least some of NA channels, an operation section for receiving encoded audio data and decoding the encoded audio data into sub-band signal data, and a data transfer section for, switching, by MA channels, the sub-band synthesis filter data and the sub-band signal data in the first memory section and the second memory section.