Abstract: The equipment comprises two microphones, sampling means, and de-noising means. The de-noising means are non-frequency noise reduction means comprising a combiner having an adaptive filter performing an iterative search seeking to cancel the noise picked up by one of the microphones on the basis of a noise reference given by the other microphone sensor. The adaptive filter is a fractional delay filter modeling a delay that is shorter than the sampling period. The equipment also has voice activity detector means delivering a signal representative of the presence or the absence of speech from the user of the equipment. The adaptive filter receives this signal as input so as to enable it to act selectively: i) either to perform an adaptive search for the parameters of the filter in the absence of speech; ii) or else to “freeze” those parameters of the filter in the presence of speech.

Abstract: The invention relates to a digital signal processing technique that changes the length of an audio signal and, thus, effectively its play-out speed. This is used for frame rate conversion or sound effects in music production. Time scaling may further be used for fast forward or slow-motion audio play-out. According said method the waveform similarity overlap add approach is modified such that a maximized similarity is determined among similarity measures of sub-sequence pairs each comprising a sub-sequence to-be-matched from a input window and a matching sub-sequence from a search window wherein said sub-sequence pairs comprise at least two sub-sequence pairs of which a first pair comprises a first sub-sequence to-be-matched and a second pair comprises a different second sub-sequence to-be-matched. The input window allows for finding sub-sequence pairs with higher similarity than with a WSOLA approach based on a single sub-sequence to-be-matched. This results in less perceivable artefacts.

Abstract: A method is provided for attenuating pre-echoes in a digital audio signal generated from a transform encoding, comprising, upon decoding and for a current frame of said digital audio signal: defining a concatenated signal from at least the reconstructed signal of the current frame, dividing said concatenated signal into subunits of samples having a predetermined length, calculating the time envelope of the concatenated signal, detecting the transition of the time envelope towards a high-energy area, determining the low-energy sub-units preceding a subunit in which a transition has been detected, and an attenuation step in said determined subunits. The attenuation is carried out according to an attenuation factor calculated for each of the determined subunits, based on the time envelope of the concatenated signal. The invention also relates to a device for implementing said method, and to a decoder including such a device.

Abstract: Captured vocals may be automatically transformed using advanced digital signal processing techniques that provide captivating applications, and even purpose-built devices, in which mere novice user-musicians may generate, audibly render and share musical performances. In some cases, the automated transformations allow spoken vocals to be segmented, arranged, temporally aligned with a target rhythm, meter or accompanying backing tracks and pitch corrected in accord with a score or note sequence. Speech-to-song music applications are one such example. In some cases, spoken vocals may be transformed in accord with musical genres such as rap using automated segmentation and temporal alignment techniques, often without pitch correction. Such applications, which may employ different signal processing and different automated transformations, may nonetheless be understood as speech-to-rap variations on the theme.

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

Abstract: A method for training a ranking application. The method includes ranking the help postings to create an initial ranking using initial parameter values, and storing user interactions with the help postings to obtain stored interactions. Simulations are performed using the stored interactions to generate revised parameter values for the ranking application. Performing the simulations includes calculating relevance values from the stored interactions, creating a test posting, assigning, to the test posting, an initial score and a relevance value randomly selected from the relevance values to generate a test ranking, and simulating user interactions with the test ranking to generate simulated rankings. The simulated rankings are analyzed to obtain revised parameter values. The method further includes ranking, using the revised parameter values, the help postings to generate a revised ranking, and displaying the help postings in the forum according to the revised ranking.

Abstract: When a frame immediately preceding an encoding target frame to be encoded by a first encoding unit operating under a linear predictive coding scheme is encoded by a second encoding unit operating under a coding scheme different from the linear predictive coding scheme, the encoding target frame can be encoded under the linear predictive coding scheme by initializing the internal state of the first encoding unit. Therefore, encoding processing performed under a plurality of coding schemes including the linear predictive coding scheme and a coding scheme different from the linear predictive coding scheme can be realized.

Abstract: A method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals.

Abstract: The present invention provides a system and method to enhance speech intelligibility and improve the detection rate of automatic speech recognizer in noisy environments. The present invention reduces an acoustically coupled loudspeaker signal from a plurality of microphone signals to enhance a near end user speech signal. A decision unit checks a system configuration parameter to determine if the cleaned speech is intended for human communication and/or Automatic Speech Recognition (ASR). A formant emphasis filer and a spectrum band reconstruction unit are used to further enhance the speech quality and improve the ASR recognition rate. The present invention can also apply to devices which has a foreground microphone(s) and a background microphone(s).

Abstract: Provided is a method of encoding an audio/speech signal, the method including determining a variable length of a frame, that is, a processing unit of an input signal in accordance with a position of an attack in the input signal; transforming each frame of the input signal to a frequency domain and dividing the frame into a plurality of sub frequency bands; and, if a signal of a sub frequency band is determined to be encoded in the frequency domain, encoding the signal of the sub frequency band in the frequency domain, and if the signal of the sub frequency band is determined to be encoded in a time domain, inverse transforming the signal of the sub frequency band to the time domain and encoding the inverse transformed signal in the time domain. According to the present invention, the audio/speech signal may be efficiently encoded by controlling time resolution and frequency resolution.

Abstract: Filter banks may have different structures and different individual output signal domains. Often a translation between different filter bank domains is desirable. Usually, mapping matrices are used that, however, vary over frequency. This requires a significant amount of lookup tables. A method for transforming first data frames of a first filter bank domain to second data frames of a different second filter bank domain, comprises steps of transcoding sub-bands of the first filter bank domain into sub-bands of an intermediate domain that corresponds to said second filter bank domain but has warped phase, and transcoding the sub-bands of the intermediate domain to sub-bands of the second filter bank domain, wherein a phase correction is performed on the sub-bands of the intermediate domain.

Abstract: An method and apparatus to extract an audio signal having an important spectral component (ISC) and a low bit-rate audio signal coding/decoding method using the method and apparatus to extract the ISC. The method of extracting the ISC includes calculating perceptual importance including an SMR (signal-to-mask ratio) value of transformed spectral audio signals by using a psychoacoustic model, selecting spectral signals having a masking threshold value smaller than that of the spectral audio signals using the SMR value as first ISCs, and extracting a spectral peak from the audio signals selected as the ISCs according to a predetermined weighting factor to select second ISCs. Accordingly, the perceptual important spectral components can be efficiently coded so as to obtain high sound quality at a low bit-rate.

Abstract: A speech decoder includes a demultiplexing unit, a low frequency band decoding unit, a band splitting filter bank unit, a coded sequence analysis unit, a coded sequence decoding/dequantization unit, a high frequency band generation unit, low frequency band time envelope calculation units that acquire a plurality of low frequency band time envelopes, a time envelope calculation unit that calculates high frequency band time envelopes using time envelope information and the plurality of low frequency band time envelopes, a time envelope adjustment unit that adjusts the time envelope of high frequency band components using the time envelopes obtained by the time envelope calculation unit, and a band synthesis filter bank unit.

Abstract: The invention relates to a method for quantifying components, wherein certain components are each determined based on a plurality of audio signals and can be calculated by the application of a linear conversion on the audio signals, said method comprising: determining a quantification function to be applied to the components by testing a condition relative to an audio signal and depending on a comparison made between a psycho-acoustic masking threshold relative to the audio signal and a value determined based on the reverse linear conversion and quantification errors of the components by the function.

Abstract: An apparatus for encoding an audio signal having a stream of audio samples has: a windower for applying a prediction coding analysis window to the stream of audio samples to obtain windowed data for a prediction analysis and for applying a transform coding analysis window to the stream of audio samples to obtain windowed data for a transform analysis, wherein the transform coding analysis window is associated with audio samples within a current frame of audio samples and with audio samples of a predefined portion of a future frame of audio samples being a transform-coding look-ahead portion, wherein the prediction coding analysis window is associated with at least the portion of the audio samples of the current frame and with audio samples of a predefined portion of the future frame being a prediction coding look-ahead portion, wherein the transform coding look-ahead portion and the prediction coding look-ahead portion are identically to each other or are different from each other by less than 20%; and an enc

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: Embodiments of the present invention provide a signal classification method and device, and encoding and decoding methods and devices. The encoding method includes: dividing a current frame into a low-frequency band signal and a high-frequency band signal; attenuating the high-frequency band signal or a to-be-encoded characteristic parameter of the high-frequency band signal according to an energy attenuation value of the low-frequency band signal, where the energy attenuation value indicates energy attenuation of the low-frequency band signal caused by encoding of the low-frequency band signal; and encoding the attenuated high-frequency band signal or the attenuated to-be-encoded characteristic parameter of the high-frequency band signal. The technical solutions according to the embodiments of the present invention can improve the effect of combining the low-frequency band signal and the high-frequency band signal at the decoder.

Abstract: A method for encoding of an audio signal comprises performing (214) of a transform of the audio signal. An energy offset is selected (216) for each of the first subbands. An energy measure of a first reference band within a low band of an encoding of a synthesis signal is obtained (212). The first high band is encoded (220) by providing quantization indices representing a respective scalar quantization of a spectrum envelope in the first subbands of the first high band relative to the energy measure of the first reference band by use of the selected energy offset. An encoder apparatus comprises means for carrying out the steps of the method. Corresponding decoder methods and apparatuses are also described.

Abstract: Method and device of extending a signal band of a voice or audio signal are provided. The bandwidth extension method includes the steps of: performing a modified discrete cosine transform (MDCT) process on an input signal to generate a first transform signal; generating a second transform signal and a third transform signal on the basis of the first transform signal; generating normalized components and energy components of the first transform signal, the second transform signal, and the third transform signal therefrom; generating an extended normalized component from the normalized components and generating an extended energy component from the energy components; generating an extended transform signal on the basis of the extended normalized component and the extended energy component; and performing an inverse MDCT (IMDCT) process on the extended transform signal.

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

Abstract: A method and associated device are provided for spatial synthesis of a sum signal to obtain at least two output signals, the sum signal as well as the spatialization parameters being output from a parametric coding by matrixing of an original multi-channel signal. The method comprises: decorrelation of the sum signal to obtain a decorrelated signal; applying a synthesis matrix, whose coefficients depend on the spatialization parameters, to the decorrelated signal and to the sum signal to obtain said output signals, wherein for at least one range of value of at least one spatialization parameter, the coefficients of the synthesis matrix are determined according to a criterion of minimizing a quantitative function, relating to the quantity of decorrelated signal in each of the output signals obtained by applying the synthesis matrix.

Abstract: Methods and apparatus for signal processing are disclosed. Source separation can be performed to extract source signals from mixtures of source signals by way of independent component analysis. Source direction information is utilized in the separation process, and independent component analysis techniques described herein use multivariate probability density functions to preserve the alignment of frequency bins in the source separation process. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims.

Abstract: An encoding apparatus comprises a frame processor (105) which receives a multi channel audio signal comprising at least a first audio signal from a first microphone (101) and a second audio signal from a second microphone (103). An ITD processor 107 then determines an inter time difference between the first audio signal and the second audio signal and a set of delays (109, 111) generates a compensated multi channel audio signal from the multi channel audio signal by delaying at least one of the first and second audio signals in response to the inter time difference signal. A combiner (113) then generates a mono signal by combining channels of the compensated multi channel audio signal and a mono signal encoder (115) encodes the mono signal. The inter time difference may specifically be determined by an algorithm based on determining cross correlations between the first and second audio signals.

Abstract: A scalable decoder device (50) for signals representing audio comprises a primary decoder (21) connected to an input (40). The primary decoder (21) is arranged to provide a primary decoded signal (23) based on received parameters (4). A primary postfilter (31) is connected to the primary decoder (23) to provide a primary postfiltered signal (32). A secondary enhancement decoder (45) is connected to the input (40) and arranged to provide a secondary decoded enhancement signal (44). The device further comprises a combiner arrangement (55), arranged for combining the primary postfiltered signal (32) and a signal (53) based on the secondary decoded enhancement signal (44) into an output signal (6) to be provided at an output (6). The combining is made with an adaptable strength relation between contributions from the two signals. A method for decoding coded signals representing audio operates in analogy with the scalable decoder device (50).

Abstract: Disclosed is a method of processing a signal, which includes receiving at least one of a first signal and a second signal, receiving mode information, and decoding the at least one of the first signal and the second signal using at least one of a first coding scheme and a second coding scheme according to the mode information. The mode information is information for indicating that a prescribed mode corresponds to one of at least three modes. The method includes detecting when a restricted mode change occurs and changing at least one mode when detecting a restricted mode change.

Abstract: The present disclosure relates to a signal analyzer for processing an overlapped input signal frame comprising 2N subsequent input signal values. The signal analyzer comprises: a windower adapted to window the overlapped input signal frame to obtain a windowed signal, wherein the windower is adapted to zero M+N/2 subsequent input signal values of the overlapped input signal frame, wherein M is equal or greater than 1 and smaller than N/2; and a transformer adapted to transform the remaining 3N/2?M subsequent windowed signal values of the windowed signal using N?M sets of transform parameters to obtain a transformed-domain signal comprising N?M transformed-domain signal values.

Abstract: Systems and methods utilize individually selected modulation spectral features for speech and speaker characterization. The method involves construction of a sparse feature space and a method of finding the approximately best feature subset for attributing a specific characteristic of speech or speaker. The current selection method is based on the Kolmogorov-Smirnov statistical test applied to individual features. The characterization task can be defined empirically and no a-priori theory is necessary to explain characteristic attribution processes. Experimental results indicate that employment of selected modulation spectral features works better than the current state-of-the-art at least in some instances of speech characterization task, e.g. prediction of speaker personality traits, as it is evident from the official results of Interspeech'2012 Speaker Personality Recognition Challenge.

Abstract: A more efficient encoder/decoder is provided in which an N-point MDCT transform is mapped into smaller sized N/2-point DCT-IV and/or DCT-II transforms with isolated pre-multiplications which can be moved to a prior or subsequent windowing stage. That is, the windowing operations may be merged with first/last stage multiplications in the core MDCT/IMDCT functions, respectively, thus reducing the total number of multiplications. Additionally, the MDCT may be systematically decimated by factor of 2 by utilizing a uniformly scaled 5-point DCT-II core function as opposed to the DCT-IV or FFT cores used in many existing MDCT designs in audio codecs. The modified windowing stage merges factors from a transform stage and windowing stage to obtain piece-wise symmetric windowing factors, which can be represented by a sub-set of the piece-wise symmetric windowing factors to save storage space. Such features offer appreciable reduction in complexity and less memory usage than the prior art.

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: A speech signal encoding method and a speech signal decoding method are provided. The speech signal encoding method includes the steps of specifying an analysis frame in an input signal; generating a modified input based on the analysis frame; applying a window to the modified input; generating a transform coefficient by performing an MDCT (Modified Discrete Cosine Transform) on the modified input to which the window has been applied; and encoding the transform coefficient. The modified input includes the analysis frame and a self replication of all or a part of the analysis frame.

Abstract: In one embodiment, a method of transceiving an audio signal is disclosed. The method includes providing low band spectral information having a plurality of spectrum coefficients and predicting a high band extended spectral fine structure from the low band spectral information for at least one subband, where the high band extended spectral fine structure are made of a plurality of spectrum coefficients. The predicting includes preparing the spectrum coefficients of the low band spectral information, defining prediction parameters for the high band extended spectral fine structure and index ranges of the prediction parameters, and determining possible best indices of the prediction parameters, where determining includes minimizing a prediction error between a reference subband in high band and a predicted subband that is selected and composed from an available low band. The possible best indices of the prediction parameters are transmitted.

Abstract: A method of receiving an audio signal includes measuring a periodicity of the audio signal to determine a checked periodicity. At least one best available subband is determined. At least one extended subband is composed, wherein composing includes reducing a ratio of composed harmonic components to composed noise components if the checked periodicity is lower than a threshold, and scaling a magnitude of the at least one extended subband based on a spectral envelope on the audio signal.

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 method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals.

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

Abstract: Methods, systems, and apparatus are presented for decoding an audio signal that includes bandwidth extension data. An audio signal that includes core audio data and bandwidth extension data can be received in a decoder. The core audio data can be associated with a core portion of an audio signal, such as the frequency range below a cutoff frequency, and the bandwidth extension data can be associated with an extended portion of the audio signal, such as a frequency range above the cutoff frequency. The core audio data can be decoded to generate a decoded core audio signal in a time domain representation. Further, an extended portion of the audio signal can be reconstructed in accordance with extension data and decoded core audio signal. Additionally, the decoded core audio signal can be lowpass filtered and the extended portion can be highpass filtered before being combined to generate a decoded output signal.

Abstract: An integer-reversible MDCT transformation is split into consecutive lifting steps, each introducing considerable rounding errors to the signal. Without noise shaping the rounding error noise will impact all frequency bins of the transformed signal equally. This is a particular problem for low signal level frequency bins. The invention limits the impact of rounding error noise coming with each lifting step in the integer-reversible transformation on the data rate of a lossless audio codec. The filter coefficients of an adaptive noise shaping filter for transform coefficients are adapted in individual lifting steps according to the current time domain signal characteristics. As an alternative, an auto-regressive pre-filter can be added in front of the lossless transformation, for raising the level of frequency regions with low power to decrease the dominance of rounding errors in these areas. Both processes can be combined to further improve lossless codec compression ratio.

Abstract: Provided is a signal distortion elimination apparatus comprising: an inverse filter application means that outputs the signal obtained by applying an inverse filter to an observed signal as a restored signal when a predetermined iteration termination condition is met and outputs the signal obtained by applying the inverse filter to the observed signal as an ad-hoc signal when the predetermined iteration termination condition is not met; a prediction error filter calculation means that segments the ad-hoc signal into frames and outputs a prediction error filter of each frame obtained by performing linear prediction analysis of the ad-hoc signal of each frame; an inverse filter calculation means that calculates an inverse filter such that a concatenation of innovation estimates of the respective frames becomes mutually independent among their samples, where the innovation estimate of a single frame (an innovation estimate) is the signal obtained by applying the prediction error filter of the corresponding frame

Abstract: A method for detecting a transient in an audio signal that has been broken up into frames includes obtaining a time domain feature of the frames and comparing the domain feature with a predetermined value. If the time domain feature is greater than the predetermined value, the frames are taken as transient and if the time domain feature is less than the predetermined value, the frames are taken as non-transient. The method has a low computational intensity and is thus very suitable for devices with limited processing resources.

Abstract: A system and method for redundant transmission is provided. In one embodiment, an input signal S is encoded as a list of fragments. Each fragment includes an index value and a projection value. The index points to an entry in a dictionary of signal elements. A repetition factor is assigned to each fragment based on its importance. After a fragment is added, a reconstructed signal is generated by decoding the list of fragments. Encoding terminates once the reconstructed signal is sufficiently close to the original signal S.

Abstract: Disclosed are a quantizer, encoder, and the methods thereof, wherein the computational load is reduced when the values related to the transform coefficients of the principal component analysis transform are quantized when a principal component analysis transform is applied to code stereo.

Abstract: A signal processing platform (300) presents (101) a signal to be processed and identifies (102) signal portions with specific characteristics that are used (103) to automatically determine at least one bounding frequency that can be used to facilitate bandwidth extension for the signal. Identifying these signal portions can comprise identifying signal portions that exhibit at least a predetermined level of energy. The step of determining the bounding frequency can comprise computing a magnitude spectrum for each of the identified signal portions that can be used to determine a corresponding measure of flatness within a pass band as pertains to a corresponding normalized signal portion to thereby provide corresponding vetted signal portions. Determining the bounding frequency can then comprise accumulating the magnitude spectrum for these vetted signal portions and using the resultant accumulation to estimate a corresponding signal envelope.

Abstract: A method for concealing lost frame includes: using history signals before the lost frame that corresponds to a lost MDCT coefficient to generate a first synthesized signal when it is detected that the MDCT coefficient is lost; performing fast IMDCT for the first synthesized signal to obtain an IMDCT coefficient corresponding to a lost MDCT coefficient; and using the IMDCT coefficient corresponding to the lost MDCT coefficient and an IMDCT coefficient adjacent to the IMDCT coefficient corresponding to the lost MDCT coefficient to perform TDAC and obtain signals corresponding to the lost frame. An apparatus for concealing lost frame is also disclosed herein. The method and the apparatus for concealing lost frames in the embodiments of the present invention make full use of the received partial signals to recover high-quality voice signals and improve the QoS.

Abstract: It is possible to improve quality of a decoding signal in a band spread for estimating a high band from a low band of a decoding signal. A first layer encoder encodes a lower band portion below a predetermined frequency of an input signal so as to generate first layer encoded information. A first layer decoder decodes the first layer encoded information so as to generate a first layer demodulated signal. A second layer encoder divides a high band portion higher, than a predetermined frequency, of an input signal into a plurality of sub-bands and estimates each of the sub-bands from the input signal or the first layer decoded signal by using the estimation result of the sub-band adjacent to the lower band side so as to generate second encoded information including the estimation results of the sub-bands.

Abstract: Provided is an encoder which can decode a high-quality stereo signal while keeping the amount of information in the bit allocation information to a minimum when a scalable coding technique is used for a stereo signal. In the encoder, a principal component analysis (PCA) converter converts the left signal and the right signal of the stereo signal and generates the main signal of the first layer and the sub-signal of the first layer. In the first layer to the M-th layer (where M is a natural number, 2 or greater), an adaptive residual encoder compares the importance of the main signal of the m-th layer, where m is a natural number from 1 to M, and the importance of the sub-signal of the m-th layer, selects the signal having the higher importance, encodes the selected signal, and generates the encoded data of the m-th layer.

Abstract: In an encoding process, a CPU transforms an audio signal from the real-time domain to the frequency domain, and transforms the signal into spectra consisting of MDCT coefficients. The CPU separates the audio signal into several frequency bands, and performs bit shifting in each band such that the MDCT coefficients can be expressed with pre-configured numbers of bits. The CPU re-quantizes the MDCT coefficients at a precision differing for each band, and transmits the values acquired thereby and shift bit numbers as encoded data. Meanwhile, in a decoding process, a CPU receives encoded data and inverse re-quantizes and inverse bit shifts the data, thereby restoring the MDCT coefficients. Furthermore, the CPU transforms the data from frequency domain to the real-time domain by using the inverse MDCT, and restores and outputs the audio signal.

Abstract: An audio encoder/decoder uses a combination of an overlap windowing transform and block transform that have reversible implementations to provide a reversible, integer-integer form of a lapped transform. The reversible lapped transform permits both lossy and lossless transform domain coding of an audio signal having variable subframe sizes.

Abstract: An apparatus and method for adaptive sub-band allocation of spectral coefficients are disclosed. The sizes of sub-bands are determined according to the distribution of spectral coefficients transformed from an input speech/audio signal to perform more elaborate quantization in units of sub-bands. Thus, quantization noise of the spectral coefficients is reduced, and sound quality in a frequency region is enhanced, thereby improving the quality of the signal.

Abstract: A method (100) includes receiving (101) an input digital audio signal comprising a narrow-band signal. The input digital audio signal is processed (102) to generate a processed digital audio signal. A high-band energy level corresponding to the input digital audio signal is estimated (103) based on a transition-band of the processed digital audio signal within a predetermined upper frequency range of a narrow-band bandwidth. A high-band digital audio signal is generated (104) based on the high-band energy level and an estimated high-band spectrum corresponding to the high-band energy level.

Abstract: A method for decoding audio frames includes producing a first frame of coded audio samples, producing at least a portion of a second frame of coded audio samples, generating audio gap filler samples based on parameters representative of a weighted segment of the first frame of coded audio samples or a weighted segment of the portion of the second frame of coded audio samples, and forming a sequence including the audio gap filler samples and the portion of the second frame of coded audio samples.