Abstract: A warped spectral estimate of an original audio signal can be used to encode a representation of a fine estimate of the original signal. The representation of the warped spectral estimate and the representation of the fine estimate can be sent to a speech recognition system. The representation of the warped spectral estimate can be passed to a speech recognition engine, where it may be used for speech recognition. The representation of the warped spectral estimate can also be used along with the representation of the fine estimate to reconstruct a representation of the original audio signal.

Abstract: Methods and devices provide improved perceived quality of an audio (or other) coded signal at a low bit-rate. An input signal may be split into an outlier portion and a stationary portion. The outlier portion of the input signal may be encoded. The stationary portion may be divided into subvectors. Each subvector may be classified as trivial or non-trivial. Each trivial subvector may be encoded using a pre-defined pattern. Each non-trivial subvector may be encoded with at least one location of at least one significant sample and a sign of the significant sample.

Abstract: For generating a parametric representation of a multi-channel signal especially suitable for low-bit rate applications, only the location of the maximum of the sound energy within a replay setup is encoded and transmitted using direction parameter information. For multi-channel reconstruction, the energy distribution of the output channels identified by the direction parameter information is controlled by the direction parameter information, while the energy distribution in the remaining ambience channels is not controlled by the direction parameter information.

Abstract: A data reproduction apparatus includes: arithmetic means for calculating difference data that indicate a difference between left-channel and right-channel data that have been compressed in a predetermined compression format; higher harmonic component generation means for generating a higher harmonic component, which was lost during compression, by performing, when the difference data's signal level exceeds a predetermined threshold, a digital limiter process that suppresses the signal level to the threshold; and adding means for adding the higher harmonic component to the left-channel and right-channel data to reproduce original data before being compressed.

Abstract: A speech decoding device of the invention smoothes, in decoding speech signal in a voice-less period, RMS and filter coefficients which is discontinuously transmitted, and provides them to a synthesis filter. Thereby, it is capable of preventing discontinuous changing of the filter coefficient caused by the intermittent transmission of the filter coefficient. As a result, a quality of decoding can be improved. Also, to remove an effect, caused by the smoothing process, from the filter coefficients or the RMS which are transmitted in the past frames, a smoothing factor is adjusted not to perform smoothing while a certain time period (or a certain number of frames) from when a transition is made from a voice period from a voice-less period, or when a decoded feature parameter satisfies a predetermined condition.

Abstract: Disclosed is an audio data packet format for transmitting an MPEG-4 HE-AAC frame via a voice channel of a mobile communication network, a method for decoding the audio data packet format, a method for correcting a codec setup error by identifying a codec used to encode sound source data inserted into a data field of voice slot data, based on the sequence number of the voice slot data, and correcting the codec setup error when a codec set up in a mobile communication terminal is different from the codec used to encode the sound source data, and a mobile communication terminal adapted to correct a codec setup error.

Abstract: In one alternative, an audio signal is analyzed using multiple psychoacoustic criteria to identify a region of the signal in which time scaling and/or pitch shifting processing would be inaudible or minimally audible, and the signal is time scaled and/or pitch shifted within that region. In another alternative, the signal is divided into auditory events, and the signal is time scaled and/or pitch shifted within an auditory event. In a further alternative, the signal is divided into auditory events, and the auditory events are analyzed using a psychoacoustic criterion to identify those auditory events in which the time scaling and/or pitch shifting processing of the signal would be inaudible or minimally audible. Further alternatives provide for multiple channels of audio.

Abstract: Playback by a decoder of a lossy compressed digital media file without quantization gaps is disclosed. The digital media file can be formed of a number of audio samples grouped into a corresponding number of audio frames. As a method, one embodiment can be carried out by identifying an encoder used to compress the digital media file; obtaining an encoder delay value for the identified encoder; obtaining a decoder delay value for the decoder; determining a audio sample count corresponding to a last valid audio sample; setting a re-synchronization after seek option marker N audio frames from the last valid audio sample; and decoding valid audio samples using the encoder delay value, the decoder delay value, and the sample count corresponding to the last valid audio sample.

Abstract: Certain aspects of a method and system for a dual mode subband acoustic echo canceller with integrated noise suppression may include splitting an input signal into a lowband component and a highband component. The subbands of each of the lowband component and the highband component may be processed in order to reduce an echo associated with the input signal and to suppress the noise associated with the input signal.

Abstract: Decoder for an audio signal coded by a coder including a long-term prediction filter wherein the decoder comprises: a block (211) for detecting transmission frame losses; a module (222) for calculating values of an error indication function representative of the cumulative adaptive excitation error during decoding following said transmission frame loss, an arbitrary value being assigned to said adaptive excitation gain for the lost frame; a module (213) for calculating an error indication parameter from said values of the error indication function; a comparator (214) for comparing said error indication parameter to at least one given threshold; and a discriminator (215) adapted to determine as a function of the results supplied by the comparator (214) a value of at least one adaptive excitation gain to be used by the decoder.

Abstract: Disclosed is a wideband audio signal coding/decoding device and method that may code a wideband audio signal while maintaining a low bit rate. The wideband audio signal coding device includes an enhancement layer that extracts a first spectrum parameter from an inputted wideband signal having a first bandwidth, quantizes the extracted first spectrum parameter, and converts the extracted first spectrum parameter into a second spectrum parameter; and a coding unit that extracts a narrowband signal from the inputted wideband signal and codes the narrowband signal based on the second spectrum parameter provided from the enhancement layer, wherein the narrowband signal has a second bandwidth smaller than the first bandwidth. The wideband audio signal coding/decoding device and method may code a wideband audio signal while maintaining a low bit rate.

Abstract: Audiovisual presentation methods, systems and apparatus for improving and enhancing the listening experience of attendees of audiovisual presentations. An exemplary audiovisual presentation system includes an audio processing and distribution unit (APDU) configured to generate and broadcast a wireless audio service containing audio of an audiovisual presentation (e.g., soundtrack and dialogue audio of a movie, in the case of a movie presentation) throughout an audiovisual presentation room or space (e.g., a movie theater, in the case of a movie presentation). The wireless audio service is received by mobile receiving devices (MRDs) having or comprising headsets, headphones or earbuds, through which MRD users listen to the audio of the audiovisual presentation provided by the wireless audio service while viewing images of the audiovisual presentation.

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a down-mix signal and down-mix identification information from an input bitstream, determining, based on the down-mix identification information, whether the down-mix signal is a 3D down-mix signal obtained by performing a three-dimensional (3D) rendering operation, and if the down-mix signal is not 3D down-mix signal, generating a 3D down-mix signal by performing a 3D rendering operation. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data include a plurality of blocks, where the blocks are classified by a block type. The first and second channel data is provided jointly if the first and second channel data are paired with each other. The method further includes obtaining frame length information indicating a length of the audio frame data, and obtaining block information indicating the block type. The block information corresponds to the first and second channel data being common when the channel data are paired. The first and second channel data are lossless decoded based on the frame length information and the block information.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. The first and second channel data is provided jointly if the first and second channel data are paired with each other. Block information indicating the block type is obtained. The block information corresponds to the first and second channel data being common when the first and second channel data are paired. The first and second channel data are lossless decoded based on the block information.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data include a plurality of blocks, where the blocks are classified by a block type. The first and second channel data are provided jointly if the first and second channel data are paired with each other. The embodiment further includes obtaining block information indicating the block type, and lossless decoding the first and second channel data based on the block information.

Abstract: A method for decoding a multi-audio-object signal having audio signals of first and second types encoded therein, the multi-audio-object signal having a downmix signal and side information having level information of the audio signals of the first and second types in a first predetermined time/frequency resolution, the method including computing a prediction coefficient matrix C based on the level information; and up-mixing the downmix signal based on the prediction coefficients to obtain a first and/or a second up-mix audio signal approximating the audio signals of the first and second types, respectively, wherein up-mixing yields the first and/or second up-mix signals S1 and S2 from the downmix signal d according to a computation representable by ( S 1 S 2 ) = D - 1 ? { ( 1 C ) ? d + H } , with “1” denoting—depending on the number of channels of d—a scalar, or an identity matrix, and D?1 being a matrix uniquely determined by a downmix prescription according

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. The embodiment further includes obtaining frame length information indicating a length of the audio frame data, and obtaining block information indicating the block type. The block information corresponds to the first and second channel data being common when the first and second channel data are paired. The first and second channel data are lossless decoded based on the frame length information and the block information.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. Block information indicating the block type is obtained. The block information corresponds to the first and second channel data being common when the first and second channel data are paired. The first and second channel data is lossless decoded based on the block information.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. The first and second channel data is provided jointly if the first and second channel data are paired with each other. The method further includes obtaining frame length information indicating a length of the audio frame data, obtaining block information indicating a block type, and lossless decoding the first and second channel data based on the frame length information and the block information.

Abstract: In a multi-channel encoder generating several different parameter sets for reconstructing a multi-channel output signal using at least one transmission channel, the data stream is written such that the two parameter sets are decodable independently of each other. Thus, a multi-channel decoder is enabled to skip a parameter set which is marked as optional and/or has a higher version number when reading the data stream and still to perform a valid multi-channel reconstruction using a data set marked as mandatory or a data set having a sufficiently low version number. This achieves a flexible encoder/decoder concept suitable for future updates characterized by backward compatibility and reliability.

Abstract: The present disclosure relates to a decoding method and apparatus. The method includes: receiving data frames from the coder; if any erroneous frame appears, calculating a pitch lag parameter of the erroneous frame; decoding the data frames according to the calculated pitch lag parameter of the erroneous frame, and obtaining decoded data. The process of determining the pitch lag parameter includes: determining the number of continuous erroneous frames and the pitch lag parameter of the previous frame; adjusting the pitch lag parameter of the previous frame according to the number of the continuous erroneous frames and a preset adjustment policy, and calculating and determining the pitch lag parameter of a current erroneous frame, wherein the preset adjustment policy is adjusting the determined pitch lag parameter of the current erroneous frame within a preset value range according to the number of the continuous erroneous frames.

Abstract: A method, device and system for signal encoding and decoding, the method comprising: encoding a core layer signal to obtain a core layer signal code; selecting an enhancement sample point that requires enhancement layer signal encoding according to the core layer signal code and the number of bits that can be used by an enhancement layer; obtaining an enhancement layer signal code of the enhancement sample point; and outputting a bit stream, where the bit stream includes the core layer signal code and the enhancement layer signal code. In embodiments of the present invention, according to the number of bits that can be used by the enhancement layer, the enhancement sample point that requires enhancement layer signal encoding is selected; the enhancement layer signal of the selected enhancement sample point is encoded and decoded; when no sufficient bits are available for the enhancement layer, the enhancement quality of the core layer can be improved.

Abstract: Methods and apparatuses for encoding a signal and decoding a signal and a system for encoding and decoding are provided. The method for encoding a signal includes performing a classification decision process on high frequency signals of input signals, adaptively encoding the high frequency signals according to the result of the classification decision process, and outputting a bitstream including codes of low frequency signals of the input signals, adaptive codes of the high frequency signals, and the result of the classification decision process. The classification decision process is performed on the high frequency signals, and adaptive encoding or adaptive decoding is performed according to the result of the classification decision process, so the quality of voice and audio output signals is improved.

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: Embodiments related to utilizing substantially optimal entries for a relatively low complexity dictionary for matching pursuits coding is disclosed. In various embodiments, methods are invoked for determining a substantially optimal entry from a bases dictionary comprising a plurality of entries; and utilizing the substantially optimal entry in a relatively low complexity matching pursuits data coding. In various embodiments, a system is provided comprising a bases dictionary comprising a plurality of entries each with a width of 15 or less; a signal to be coded; and a selection module configured to receive at least one of the plurality of entries from the bases dictionary, to calculate an inner product between the at least one of the plurality of entries and the signal to be coded, and to select the entry from the at least one of the plurality of entries that produces a maximum inner product for use in at least partially coding the signal to be coded.

Abstract: An encoding device and an encoding method are provided for encoding by reducing the number of samples to be processed when encoding higher-band spectrum data according to lower-band spectrum data in a wide-band signal. The device and the method can obtain a high-quality decoded signal even if a large quantization distortion is caused in the lower-band spectrum data. When encoding higher-band spectrum data in a signal to be encoded, according to lower-band spectrum data in the signal, only for a part (a head portion) of the higher-band spectrum data, the lower-band spectrum data after being quantized is subjected to approximate partial search and higher-band spectrum data is generated according to the search result.

Abstract: A multi-stage recursive sample rate converter (“SRC”) typically embodied as digital signal processor provides for an efficient structure for converting digital audio samples at one frequency, such as 48 kHz, to another frequency, such as 44.1 kHz. A parameter codebook comprising memory stores parameters used at a plurality of stages by the SRC. For each stage, a controller coordinates the SRC to use the appropriate set of parameters from the codebook, process an input audio sample stream, and store the intermediate results in a buffer. The controller then causes the intermediate results to be processed again as input to the SRC in a subsequent stage of processing using a different set of parameters. The process is repeated until all stages are completed, and the final results are the output digital audio data stream at the desired sampling rate.

Abstract: Techniques for introducing information into a data stream first obtains the spectral values of the short-term spectrum of the audio signal. Separately, information to be introduced are combined with a spread sequence obtaining a spread information signal, whereupon a spectral representation of the spread information is generated, then weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal are then summed and afterwards processed again to obtain a processed data stream including audio information and information to be introduced.

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: When performing audio communication by using different encoding/decoding methods, a code obtained by encoding audio by a certain method is converted into a code decodable by another method with a high audio quality and a small calculation amount. In a code conversion device for converting a first code string into a second code string, an audio decoding circuit acquires a first linear prediction coefficient and excitation signal information from the first code string and drives the filter having the first linear prediction coefficient by the excitation signal obtained from the excitation signal information, thereby creating a first audio signal. A fixed codebook code generation circuit uses the fixed codebook information and minimizes the distance between the second audio signal generated from the information obtained from the second code string and the first audio signal, thereby obtaining the fixed codebook information in the second code string.

Abstract: The present invention is based on the finding that a reconstructed output channel, reconstructed with a multi-channel reconstructor using at least one downmix channel derived by downmixing a plurality of original channels and using a parameter representation including additional information on a temporal fine structure of an original channel can be reconstructed efficiently with high quality, when a generator for generating a direct signal component and a diffuse signal component based on the downmix channel is used. The quality can be essentially enhanced, if only the direct signal component is modified such that the temporal fine structure of the reconstructed output channel is fitting a desired temporal fine structure, indicated by the additional information on the temporal fine structure transmitted.

Abstract: Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent.

Abstract: A prediction performance between individual channels of a stereo signal is improved to improve a sound quality of a decoded signal. A first low pass filter LPF interrupts a high-range component of a first channel signal S1, and outputs a first low-range component S1?. A second low pass filter LPF interrupts a high-range component of a second channel signal S2, and outputs a second low-range component S2?. A predictor predicts the S2? from the S1?, and outputs a prediction parameter composed of a delay time difference t and an amplitude ratio g. first channel encoder encodes the S1. A prediction parameter encoder encodes the prediction parameter. The encoded parameters of the encoded parameter of the S1 and the prediction parameter are then outputted.

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: Methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR) are introduced. The problem of insufficient noise contents is addressed in a reconstructed highband, by using Adaptive Noise-floor Addition. New methods are also introduced for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The methods and apparatus used are applicable to both speech coding and natural audio coding systems.

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: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: Disclosed is a method capable of adaptively aligning windows to extract features according to the types and characteristics of voice signals. To this end, window lengths based on the window update points in a corresponding order are determined by employing the concept of a higher order peak, and windows are aligned according to window lengths. When the windows are aligned according to such a manner, the start and end points of each window is known, so that it becomes possible to easily extract and analyze peak feature information.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: A subband coding apparatus carries out subband coding which prevents deterioration in coding performance and improves audio quality of decoded signals. The subband coding apparatus includes a low-band coding section (103) to code a low-band spectrum (S13). A low-band decoding section (106) decodes a low-band coded data (S14) and outputs a decoded low-band spectrum (S18) to a high-band coding section (107). A spectrum rearranging section (105) rearranges to make each frequency component of a high-band spectrum (S16) in reverse order on the frequency axis and outputs a modified high-band spectrum (S17) after rearranging to a high-band coding section (107). The high-band coding section (107) uses the decoded low-band spectrum (S18) output from the low-band decoding section (106) to code the modified high-band spectrum (S17) output from the spectrum rearranging section (105).

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

Abstract: An audio system for processing two channels of audio input to provide more than two output channels. The input may be conventional stereo material or compressed audio signal data. The audio processing includes separating the input signals into frequency bands and processing the frequency bands according to processes which may differ from band to band. The audio processing includes no processing of L?R signals.

Abstract: Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent.

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

Abstract: A scalable coding apparatus is provided to suppress deterioration of a quality of a coded signal in a normal frame next to a frame compensated for the occurrence of a data loss. The scalable coding apparatus is provided with a core-layer coding section (11) to carry out core-layer coding for the n-th frame input audio signal, an ordinary coding section (121) to generate expanding-layer ordinary-coding layer L2(n) by carrying out ordinary-coding of an expanding layer for the input audio signal, a deterioration-compensation coding section (123) to generate an expanding-layer-deterioration coding data L2?(n) by carrying out compensation for quality deterioration of coded audio in a current frame due to a past frame loss, a judging section (125) to determine whether either the expanding-layer ordinary-coding data L2(n) or the expanding-layer deterioration-coding data L2?(n) should be output from the expanding-layer coding section (12) as expanding-layer coding data of the current frame.

Abstract: A method and apparatus for transforming an audio signal, a method and apparatus for adaptively encoding an audio signal, a method and apparatus for inversely transforming an audio signal, and a method and apparatus for adaptively decoding an audio signal. The method of transforming an audio signal includes determining a transform unit into which the audio signal in a time domain is to be transformed into an audio signal in a frequency domain, and transforming the audio signal into an audio signal in the frequency domain according to the determined transform units using a window coefficient other than 0. Accordingly, it is possible to minimize distortion of the audio signal when encoding the audio signal even at a high bit rate while increasing efficiency of compression.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: Methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR) are introduced. The problem of insufficient noise contents is addressed in a reconstructed highband, by using Adaptive Noise-floor Addition. New methods are also introduced for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The methods and apparatus used are applicable to both speech coding and natural audio coding systems.