Abstract: A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.

Abstract: An audio encoder, an audio decoder or an audio processor includes a filter for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal, the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.

Abstract: An audio decoder has an arithmetic decoder for providing decoded spectral values on the basis of an arithmetically-encoded representation and a frequency-domain-to-time-domain converter for providing a time-domain audio representation. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state described by a numeric current context value which is determined in dependence on previously decoded spectral values. The arithmetic decoder obtains a plurality of context subregion values on the basis of previously decoded spectral values and derives a numeric current context value associated with one or more spectral values to be decoded in dependence on stored context subregion values. The arithmetic decoder computes the norm of a vector formed by a plurality of previously decoded spectral values in order to obtain a common context subregion value. An audio encoder uses a similar concept.

Abstract: A method and apparatus for decoding portions of a data stream, wherein each portion comprises a plurality of samples. The method comprises storing portions of the data stream, decoding portions of the data stream to form decoded portions, and storing the decoded portions. The method further comprises identifying that a portion of the data stream is degraded. Following identifying that a portion of the data stream is degraded, the method generates a decoded portion for the degraded portion of the data stream using the stored decoded portions. The method also updates a state of a decoder by: estimating a pitch period of the degraded portion; selecting a group of successive samples of the stored portions of the data stream, the group of successive samples offset from the degraded portion in the data stream by a multiple of the estimated pitch period; and decoding the selected samples at the decoder.

Abstract: Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder.

Abstract: Provided are an encoding apparatus and a decoding apparatus of a multi-channel signal. The encoding apparatus of the multi-channel signal may process a phase parameter associated with phase information between a plurality of channels constituting the multi-channel signal, based on a characteristic of the multi-channel signal. The encoding apparatus may generate an encoded bitstream with respect to the multi-channel signal using the processed phase parameter and a mono signal extracted from the multi-channel signal.

Abstract: An encoder, based on a combination of two audio channels, obtains a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid signal. The first combination signal and the prediction residual signal are encoded and written into a data stream together with the prediction information. A decoder generates decoded first and second channel signals using the prediction residual signal, the first combination signal and the prediction information. A real-to-imaginary transform may be applied for estimating the imaginary part of the spectrum of the first combination signal. For calculating the prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

Abstract: An audio decoder includes an arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically encoded representation of the spectral values, and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state described by a numeric current context value. The arithmetic decoder determines the numeric current context value in dependence on a plurality of previously decoded spectral values. The arithmetic decoder evaluates a hash table, entries of which define both significant state values and boundaries of intervals of numeric context values, in order to select the mapping rule. A mapping rule index value is individually associated to a numeric context value being a significant state value.

Abstract: In a selective signal encoder, an input signal is first encoded using a core layer encoder to produce a core layer encoded signal. The core layer encoded signal is decoded to produce a reconstructed signal and an error signal is generated as the difference between the reconstructed signal and the input signal. The reconstructed signal is compared to the input signal. One of two or more enhancement layer encoders selected dependent upon the comparison and used to encode the error signal. The core layer encoded signal, the enhancement layer encoded signal and the selection indicator are output to the channel (for transmission or storage, for example).

Abstract: Method and apparatus comprising a method of recording natural sounds with a matched microphone array, recording the signal on a high resolution recording device including creating an audio bed, and playing back the recording on a tuned playback system. The method and apparatus is used to create or duplicate an ambient sound space for ambient therapy.

Abstract: A sensor is configured to determine at least one operating condition of a device and a selector is configured to select an audio coding process for the device, based on the operating condition. The operating condition may include remaining battery life of the device and/or ambient noise level. The selected audio coding process may consume less power than another possible audio coding process during audio processing. The audio may include voice and/or audio playback, e.g., music playback.

Abstract: Methods and apparatuses for performing song detection on an audio signal are described. Clips of the audio signal are classified into classes comprising music. Class boundaries of music clips are detected as candidate boundaries of a first type. Combinations including non-overlapped sections are derived. Each section meets the following conditions: 1) including at least one music segment longer than a predetermined minimum song duration, 2) shorter than a predetermined maximum song duration, 3) both starting and ending with a music clip, and 4) a proportion of the music clips in each of the sections is greater than a predetermined minimum proportion. In this way, various possible song partitions in the audio signal can be obtained for investigation.

Abstract: An apparatus for obtaining a parameter describing a variation of a signal characteristic of a signal on the basis of actual transform-domain parameters describing the audio signal in transform-domain includes a parameter determinator. The parameter determinator is configured to determine one or more model parameters of a transform-domain variation model describing an evolution of the transform-domain parameters in dependence on one or more model parameters representing a signal characteristic, such that a model error, representing a deviation between a modeled temporal evolution of the transform-domain parameters and an evolution of the actual transform-domain parameters, is brought below a predetermined threshold value or minimized.

Abstract: A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.

Abstract: A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end.

Abstract: A method and device for searching an algebraic codebook during encoding of a sound signal, wherein the algebraic codebook comprises a set of codevectors formed of a number of pulse positions and a number of pulses distributed over the pulse positions. In the algebraic codebook searching method and device, a reference signal for use in searching the algebraic codebook is calculated. In a first stage, a position of a first pulse is determined in relation with the reference signal and among the number of pulse positions. In each of a number of stages subsequent to the first stage, (a) an algebraic codebook gain is recomputed, (b) the reference signal is updated using the recomputed algebraic codebook gain and (c) a position of another pulse is determined in relation with the updated reference signal and among the number of pulse positions.

Abstract: A packet generator for generating packets from an input signal configured to: generate at least one first signal, dependent on the input signal, the first signal comprising a first relative time value; generate at least one second signal, dependent on the input signal and associated with the at least one first signal; and generate at least one indicator associated with each of the at least one second signal, each indicator dependent on the first relative time value.

Abstract: A decoding device is capable of flexibly calculating high-band spectrum data with a high accuracy in accordance with an encoding band selected by an upper-node layer of the encoding side. In this device: a first layer decoder decodes first layer encoded information to generate a first layer decoded signal; a second layer decoder decodes second layer encoded information to generate a second layer decoded signal; a spectrum decoder performs a band extension process by using the second layer decoded signal and the first layer decoded signal up-sampled in an up-sampler so as to generate an all-band decoded signal; and a switch outputs the first layer decoded signal or the all-band decoded signal according to the control information generated in a controller.

Abstract: An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source.

Abstract: An encoder for encoding an audio signal comprising at least two channels, the encoder configured to generate an encoded signal comprising at least a first part, a second part and a third part, wherein the encoder is further configured to: generate the first part of the encoded signal dependent on at least one combination of first and second channels of the at least two channels; generate the second part of the encoded signal dependent on at least one difference between the first and second channels of the at least two channels; and generate the third part of the encoded signal dependent on at least one energy ratio of the first and second channels of the at least two channels.

Abstract: Method and apparatus for processing audio signals are provided. The method for decoding an audio signal includes receiving filter information, applying spatial information to the filter information to generate surround converting information, and outputting the surround converting information. The apparatus for decoding an audio signal includes a filter information receiving part receiving filter information; an information converting part applying spatial information to the filter information to generate surround converting information, and a surround converting information output part outputting the surround converting information.

Abstract: Provided are a method and apparatus for encoding/decoding stereo audio. In the method for encoding stereo audio, stereo audio is encoded based on at least one of the phase difference between first and second channel audios and information on an angle made by a vector on the intensity of mono-audio and a vector on the intensity of the first channel audio or a vector on the intensity of the second channel audio. Thus, the number of encoded parameters is minimized so that a compression ratio in the encoding of the stereo audio is improved.

Abstract: A method and apparatus for prediction in a speech-coding system extends a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, to a multi-tap LTP filter. From another perspective, a conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. Such a multi-tap LTP filter offers a number of advantages over the prior-art. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients (?i's) of the multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component. Consequently their main function is to maximize the prediction gain of the LTP filter via modeling the degree of periodicity that is present and by imposing spectral shaping.

Abstract: A method of processing a signal is disclosed. The present invention includes receiving extension information and at least one downmix signal of a first downmix signal decoded by a audio coding scheme and a second downmix signal decoded by a speech coding scheme; determining an extension base signal corresponding to a partial region of the downmix signal based on the extension information; and generating an extended downmix signal having a bandwidth extended by reconstructing a high frequency region signal using the extension base signal and the extension information. According to a signal processing method and apparatus of the present invention, signal corresponding to a partial frequency region of the downmix signal is used as the extension base signal. Therefore, the high frequency region of the downmix signal is reconstructed by using the extension base signal having variable bandwidth.

Abstract: Information useful for modifying the dynamics of an audio signal is derived from one or more devices or processes operating at one or more respective nodes of each of a plurality of hierarchy levels, each hierarchical level having one or more nodes, in which the one or more devices or processes operating at each hierarchical level takes a measure of one or more characteristics of the audio signal such that the one or more devices or processes operating at each successively lower hierarchical level takes a measure of one or more characteristics of progressively smaller subdivisions of the audio signal.

Abstract: Method, apparatus, and system for encoding and decoding signals are disclosed. The encoding method includes: converting a first-domain signal into a second-domain signal; performing Linear Prediction (LP) processing and Long-Term Prediction (LTP) processing for the second-domain signal; obtaining a long-term flag according to a decision criterion; obtaining a second-domain predictive signal according to the LP processing result and the LTP processing result when the long-term flag is a first flag; or obtaining a second-domain predictive signal according to the LP processing result when the long-term flag is a second flag; converting the second-domain predictive signal into a first-domain predictive signal, calculating a first-domain predictive residual signal; and outputting a bit stream that includes the first-domain predictive residual signal.

Abstract: The present disclosure includes processing a signal to generate a first sub-set of data, transmitting the first sub-set of data for generation of a reconstructed audio signal, the reconstructed audio signal having a fidelity relative to the signal, processing the signal to generate a second sub-set of data and a third sub-set of data, the second sub-set of data defining a second portion of the signal and comprising data that is different than data of the first sub-set of data, and the third sub-set of data defining a third portion of the signal and comprising data that is different than data of the first and second sub-sets of data, comparing a priority of the second sub-set of data to a priority of the third sub-set of data, and transmitting one of the second sub-set of data and the third sub-set of data over the network for improving the fidelity.

Abstract: An audio encoding method and apparatus and an audio decoding method and apparatus are provided. The audio signal decoding method includes extracting a downmix signal and object-based side information from an audio signal; generating a modified downmix signal based on the downmix signal and extracted information which is extracted from the object-based side information; generating channel-based side information based on the object-based side information and control data for rendering the downmix signal; and generating a multi-channel audio signal based on the modified downmix signal and the channel-based side information.

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: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises a linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; a quantization unit for quantizing a transform domain signal; a long term prediction unit for determining an estimation of the frame of the filtered input signal based on a reconstruction of a previous segment of the filtered input signal; and a transform domain signal combination unit for combining, in the transform domain, the long term prediction estimation and the transformed input signal to generate the transform domain signal.

Abstract: The embodiments of the present invention relate to a compression coding and decoding method, a coder, a decoder and a coding device. The compression coding method includes: extracting sign information of an input signal to obtain an absolute value signal of the input signal; obtaining a residual signal of the absolute value signal by using a prediction coefficient, where the prediction coefficient is obtained by prediction and analysis that are performed according to a signal characteristic of the absolute value signal of the input signal; and multiplexing the residual signal, the sign information and a coding parameter to output a coding code stream, after the residual signal, the sign information and the coding parameter are respectively coded, so as to improve compression efficiency of a voice and audio signal.

Abstract: A method and an apparatus for generating a lattice vector quantizer codebook are disclosed. The method includes: storing an eigenvector set that includes amplitude vectors and/or length vectors, where the amplitude vectors and/or length vectors are different from each other and correspond to a root leader of a lattice vector quantizer; storing storage addresses of the amplitude vectors and length vectors, where the amplitude vectors and length vectors correspond to the root leader and are in the eigenvector set; and generating a lattice vector quantizer codebook according to the eigenvector set and the storage addresses.

Abstract: Provided is an encoding/decoding apparatus and method of multi-channel signals. The encoding apparatus and method of multi-channel signals may encode phase information of the multi-channel signals using a quantization scheme and a lossless encoding scheme, and the decoding apparatus and method of multi-channel signals may decode the phase information using an inverse-quantization scheme and a lossless decoding scheme.

Abstract: Provided are parametric audio encoding and decoding apparatuses and methods thereof. In the parametric audio encoding method, an audio signal is segmented into a plurality of segments. At least one sine wave is extracted from each of the segments, and the extracted sine waves are connected. It is determined whether an extracted sine wave is a birth sine wave. If the extracted sine wave is a birth sine wave, a bit stream is generated by encoding the phase of the birth sine wave on the basis of the frequency of the birth sine wave, wherein the number of bits allocated to encode the phase of the birth sine wave is adjusted according to the frequency of the birth sine wave.

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: 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: There is provided an audio encoding device capable of effectively encoding stereo audio in audio encoding having monaural-stereo scalable configuration. In this device, a correlation degree comparison unit (304) calculates correlation in a first channel (correlation degree between the past signal and the current signal in the first channel) from the first channel audio signal and calculates correlation in a second channel (correlation degree between the past signal and the current signal in the second channel) from the second channel audio signal. The correlation in the first channel is compared to the correlation in the second channel. A channel having the greater correlation is selected.

Abstract: A technique of spectral noise shaping in an audio coding system is disclosed. Frequency decomposition of an input audio signal is performed to obtain multiple frequency sub-bands that closely follow critical bands of human auditory system decomposition. The tonality of each sub-band is determined. If a sub-band is tonal, time domain linear prediction (TDLP) processing is applied to the sub-band, yielding a residual signal and linear predictive coding (LPC) coefficients of an all-pole model representing the sub-band signal. The residual signal is further processed using a frequency domain linear prediction (FDLP) method. The FDLP parameters and LPC coefficients are transferred to a decoder. At the decoder, an inverse-FDLP process is applied to the encoded residual signal followed by an inverse TDLP process, which shapes the quantization noise according to the power spectral density of the original sub-band signal. Non-tonal sub-band signals bypass the TDLP process.

Abstract: There is provided an audio encoding device capable of effectively encoding a stereo audio even when a correlation between channels of the stereo audio is small. In the device, a monaural signal generation unit (110) generates a monaural signal by using a first channel signal and a second channel signal contained in the stereo signal. An encoding channel selection unit (120) selects one of the first channel signal and the second channel signal. An encoding unit including a monaural signal encoding unit (112), a first channel encoding unit (122), a second channel encoding unit (124), and a switching unit (126) encodes the generated monaural signal to obtain core-layer encoded data and encodes the selected channel signal to obtain extended layer encoded data corresponding to the core-layer encoded data.

Abstract: An encoder capable of reducing the degradation of the quality of the decoded signal in the case of band expansion in which the high band of the spectrum of an input signal is estimated from the low band. In this encoder, a first layer encoder encodes an input signal and generates first encoded information, a first layer decoder decodes the first encoded information and generates a first decoded signal, a characteristic judger analyzes the intensity of the harmonic structure of the input signal and generates harmonic characteristic information representing the analysis result, and a second layer encoder changes, on the basis of the harmonic characteristic information, the numbers of bits allocated to parameters included in second encoded information created by encoding the difference between the input signal and the first decoded signal before creating the second information.

Abstract: An apparatus for processing a media signal and method thereof are disclosed, by which the media signal can be converted to a surround signal by using spatial information of the media signal. The present invention provides a method of processing a signal, the method comprising of generating rendering information by using spatial information indicating features between multi-sources and filter information having a surround effect; and generating a surround signal by applying the rendering information to a downmix signal generated by downmixing the multi-sources, wherein a tab number of a filter used in applying the rendering information to generate the surround signal is variable.

Abstract: Disclosed are a context-based arithmetic encoding apparatus and method and a context-based arithmetic decoding apparatus and method. The context-based arithmetic decoding apparatus may determine a context of a current N-tuple to be decoded, determine a Most Significant Bit (MSB) context corresponding to an MSB symbol of the current N-tuple, and determine a probability model using the context of the N-tuple and the MSB context. Subsequently, the context-based arithmetic decoding apparatus may perform a decoding on an MSB based on the determined probability model, and perform a decoding on a Least Significant Bit (LSB) based on a bit depth of the LSB derived from a process of decoding on an escape code.

Abstract: This invention relates to a method, a computer program product, apparatuses and a system for extracting coded parameter set from an encoded audio/speech stream, said audio/speech stream being distributed to a sequence of packets, and generating a time scaled encoded audio/speech stream in the parameter coded domain using said extracted coded parameter set.

Abstract: The signal processing is based on the concept of using a time-domain aliased (12, TDA) frame as a basis for time segmentation (14) and spectral analysis (16), performing segmentation in time based on the time-domain aliased frame and performing spectral analysis based on the resulting time segments. The time resolution of the overall ?segmented? time-to-frequency transform can thus be changed by simply adapting the time segmentation to obtain a suitable number of time segments based on which spectral analysis is applied. The overall set of spectral coefficients, obtained for all the segments, provides a selectable time-frequency tiling of the original signal frame.

Abstract: An encoding system includes a sampling unit, a computing unit, a comparing unit, a quantifying unit, and an encoding unit. The sampling unit obtains first sample data of a current sampling point and second sample data of a previous sampling point. The computing unit computes a data difference between the first sample data and the second sample data. The data difference includes a numeral and a sign. The comparing unit determines whether the data difference is more than or equal to 0 and outputs a determining result. The quantifying unit quantifies the numeral of the data difference. The encoding unit encodes the numeral of the data difference with or without the sign according to the determining result.

Abstract: A frame is received that has the wideband audio signal. The low band audio signal is encoded to generate an encoded low band signal. The high band signal is analyzed to determine whether the high band signal is perceptually relevant to the low band signal. If the high band signal is not perceptually relevant to the low band signal, the low band signal is encoded and provided in a frame to the decoder without including parameters corresponding to characteristics of the high band signal. If the high band signal is perceptually relevant, the high band signal is encoded to generate an encoded high band signal. The resultant frame that is sent to the decoder will include a combination of the encoded low band signal and the encoded high band signal.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: Provided are a method and apparatus for encoding the frequency of a continuation sinusoidal signal and a method and apparatus for decoding the same. In the encoding method, a continuation sinusoidal signal successive to a sinusoidal signal in a previous section is extracted from a current section; a frequency of the continuation sinusoidal signal at the boundary between the current and previous sections is changed to a first frequency, based on representative frequencies of the continuation sinusoidal signal and at least one sinusoidal signal that belongs to a section adjacent to the current section and is successive to the continuation sinusoidal signal; and the first frequency is encoded.

Abstract: The present invention provides a frequency-domain spatial audio coding framework based on the perceived spatial audio scene rather than on the channel content. In one embodiment, time-frequency spatial direction vectors are used as cues to describe the input audio scene.

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.