Abstract: Signals of different channels are combined into one mono signal. A set of adaptive filters, preferably one for each channel, is derived in a respective filter adaptation unit. When an adaptive filter is applied to the mono signal it reconstructs the signal of the respective channel under a perceptual constraint. The perceptual constraint is a gain and/or shape constraint. The gain constraint allows the preservation of the relative energy between the channels while the shape constraint allows more stability by avoiding unnecessary filtering of spectral nulls. The transmitted parameters are the mono signal, in encoded form, and the parameters of the adaptive filters, preferably also encoded. The receiver reconstructs the signal of the different channels by applying the adaptive filters and possibly some additional post-processing.

Abstract: An MPEG-1 layer 3 audio encoder, including a scalefactor generator for determining first scalefactors for encoding a block of audio data if a temporal masking transient is not detected in said block of audio data; and for selecting the maximum of said scalefactors for encoding said block of audio data if a temporal masking transient is detected in said block of audio data to enable greater compression of said audio data. Increases in quantization error, due to use of the maximum scalefactor are pre-masked or post-masked by the temporal masking transient. In cases where the last portion of a block includes a temporal masking transient that masks the preceding portions of the block, the maximum scalefactor is only used to encode the block if the resulting increase in quantization error is less than 30% of the quantization error for the block.

Abstract: A spectral representation of an audio signal having consecutive audio frames can be derived more efficiently, when a common time warp is estimated for any two neighboring frames, such that a following block transform can additionally use the warp information. Thus, window functions required for successful application of an overlap and add procedure during reconstruction can be derived and applied, the window functions already anticipating the re-sampling of the signal due to the time warping. Therefore, the increased efficiency of block-based transform coding of time-warped signals can be used without introducing audible discontinuities.

Abstract: Coding an audio signal of a sequence of audio values into a coded signal includes determining first and second listening thresholds for first and second blocks of audio values of the sequence of audio values; calculating versions of first second parameterizations of the parameterizable filter such that the transfer function thereof roughly corresponds to the inverse of the magnitude of the first and second listening thresholds, respectively; filtering a predetermined block of audio values of the sequence of audio values with the parameterizable filter using a predetermined parameterization which depends on the version of the second parameterization to obtain a block of filtered audio values corresponding to the predetermined block which is quantized; forming a difference between the versions of the first and second parameterizations; integrating information on, inter alias, the difference into the coded signal.

Abstract: Digital audio data are divided into a plurality of frames, each of which includes a desired number of sub-band samples, which are gradually increased in a range between “16” and “1024”, and are then compressed by way of psychoacoustics analysis and quantization, whereby compressed data are realized with a high compression ratio and small tone-generation latency. The compressed data are decoded by way of inverse quantization and sub-band synthesis, so that decoded data are sequentially written into a memory (e.g., a FIFO memory). Decoding is appropriately turned on or off in response to a presently vacant capacity of the memory.

Abstract: Input speech is coded in an encoder (11), the coded speech is decoded in a decoder (12), compensatory speech which compensates the speech of the current frame is generated in a compensatory speech generating part (20) by using past decoded speech, the quality of the compensatory speech is evaluated by using the input speech and the compensatory speech and a duplication level is generated the value of which increases incrementally with decreasing speech quality evaluation value in a speech quality evaluating part (40), and as many identical packets as the number specified by the duplication level is generated for the coded speech in a packet generating part (15), and the packets are transmitted, thereby reducing the possibility that packet loss will occur at the receiving end.

Abstract: To reduce the amount of transmitted information and further reduce the processing amount at a decoding apparatus.

Abstract: Disclosed is a stereo encoding device which can improve critical channel encoding accuracy without increasing the encoding information amount. The device includes: a monaural signal synthesis unit (101) which combines a left channel signal L(n) and a right channel signal R(n) so as to generate a monaural signal M(n); a correlation coefficient calculation unit (102) which calculates a correlation coefficient CML between M(n) and L(n) and a correlation coefficient CMR between M(n) and R(n); a critical channel judging unit (103) which decides one of the L(n) and R(n) having a smaller correlation with M(n) as the critical channel if the ratio of CML against CMR is not within a predetermined range from 90% to 111%, for example; and an ICP encoding unit (104) which performs ICP encoding by adjusting the degree of the ICP parameter of the critical channel to be higher than the degree of the ICP parameter of the non-critical channel.

Abstract: An audio decoding apparatus for decoding and reproducing a plurality of compressed audio streams simultaneously without sound interruption, even when the number of samples per frame is different. The audio decoding apparatus includes: a first and second audio decoder which decode two inputted compressed audio streams, and output audio data; a first and second intermediate buffer which temporarily hold the outputted audio data; a first and second audio output unit which convert the audio data into audio signals and output such audio signals; an output control unit which reads the audio data from the first and second intermediate buffer, and transmits the audio data to the first and second audio output unit. The output control unit repeats the reading and transmission of either the same number of samples of audio data or the number of samples of audio data for the same amount of transmission time.

Abstract: High audibility output is realized when audio output is provided in special playback. In special playback with audio output, skip/repeat control is done so that decoding and outputting of the audio data is periodically repeated/skipped during part of one frame. The output level may be corrected so as to emphasize appropriate frequency components. This realizes good audio output. In addition, the skip/repeat control and output level correcting methods are changed according to characteristics of the audio data to be reproduced. Also, this realizes good audio output.

Abstract: A decoding method for MP3 bit streams, which replaces a buffer required in the decoding process by manipulating the order of data decoding. The decoding method includes reading the head and side information of the current frame, and calculating a main data's start address of the current frame. While decoding the main data, the head and side information of subsequent frames are skipped if the reading of the main data is not yet completed. The start address of the next frame is calculated and directly accessed after finished reading the main data of the current frame. An optimum method for accessing frequency lines utilizes the characteristics of the MP3 frequency line, instead of inserting a plurality of zeros in the rzero zone containing successive zeros, the initial boundary address of the rzero zone is memorized.

Abstract: An acoustic signal encoding device for down-mixing at different ratios to encode a multichannel signal with a small number of channels, and an acoustic signal decoding device for decoding the signal encoded by the acoustic signal encoding device. In these devices, weighting means (103) in the acoustic signal encoding device (100) weights input signals of two channels individually according to a down-mixing coefficient thereby to calculate the level difference of the signals of two channels weighted by a level difference calculation unit (104). A separating unit (202) in the acoustic signal decoding device (200) separates the down-mixed signals into signals of two channels with the level difference information weighted.

Abstract: An audio decoder (100) comprising: effect means, decoding means, and rendering means. The effect means (500) generate modified down-mix audio signals from received down-mix audio signals. Said received down-mix audio signals comprise a down-mix of a plurality of audio objects. Said modified down-mix audio signals are obtained by applying effects to estimated audio signals corresponding to audio objects comprised in said received down-mix audio signals. Said estimated audio signals are derived from the received down-mix audio signals based on received parametric data. Said received parametric data comprise a plurality of object parameters for each of the plurality of audio objects. Said modified down-mix audio signals based on a type of the applied effect are decoded by decoding means or rendered by rendering means or combined with the output of rendering means.

Abstract: A method for providing a motion signal with a sound signal using an existing sound signal encoding format. The method comprises providing the motion signal, providing the sound signal, inserting the motion signal in an available data field provided in the existing encoding algorithm, encoding the sound signal with the inserted motion signal according to the existing encoding algorithm to generate an encoded bitstream sound signal and providing the encoded bitstream sound signal comprising the motion signal and the sound signal.

Abstract: A voice speed control apparatus comprising: an utterance/non-utterance judging unit judging whether a processing target segment in inputted a voice signal is an utterance segment or a non-utterance segment; a non-utterance continuation length acquiring unit acquiring a non-utterance continuation length representing a length of the voice signal judged continuously to be the non-utterance; a determining unit determining a reproducing speed of the processing target segment in the voice signal in accordance with the non-utterance continuation length so that the reproducing speed gets higher as the non-utterance continuation length gets larger and so that an increase in reproducing speed is restrained to a greater degree as the non-utterance continuation length becomes smaller; and a changing unit changing the reproducing speed of the voice signal, corresponding to the reproducing speed.

Abstract: A multi-channel synthesizer for generating at least three output channels using an input signal having at least one base channel, the base channel being derived from the original multi-channel signal, the input signal further including at least two different up-mixing parameters, and an up-mixer mode indication indicating, in a first state that a first up-mixing rule is to be performed, and, indicating, in a second state, that a different second up-mixing rule is to be performed, uses an up-mixer for up-mixing the at least one base channel using the at least two different up-mixing parameters based on the first or the second up-mixing rule in response to the up-mixer mode indication so that the at least three output channels are obtained.

Abstract: A method and system for operating an electronic device (102) that is operably coupled to an audio output device (104) is provided. The method includes receiving an encoded audio file at the electronic device. Further, the method includes selectively decoding (202) the encoded audio file, in correspondence with a spectral response of the audio output device, to provide decoded audio data. Furthermore, the method includes playing (204) the decoded audio data over the audio output device.

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 system and method for seamless switching and concatenation of compressed audio streams in Internet, Digital Radio, Digital Television, DVD, storage, and other applications. The technology allows switching between streams at pre-determined points without the introduction of audible artifacts. It can be used for the personalization messages such as advertisements, news systems and other.

Abstract: Digital audio samples are represented as a product of scale factors codes and corresponding quantity codes, sometimes referred to as exponent/mantissa format. To compress audio data, scale factors are organized by sample time and frequency either by filtering or frequency transformation, into a two-dimensional frame. The frame may be decomposed into “tiles” by partition. One or more such scale factor tiles are compressed by transformation by a two-dimensional, orthogonal transformation such as a two dimensional discrete cosine transform. Optional further encoding is applied to reduce redundancy. A decoding method and an encoded machine readable medium complement the method of encoding.

Abstract: There is provided a communication device for effectively encoding an audio/music signal while maintaining a predetermined quality by controlling the transmission bit rate of the transmission side considering the use environment of the reception side. In this device, a transmission mode decision unit (101) detects an environment noise contained in the background of the audio/music signal in the input signal and decides the transmission mode controlling the transmission bit rate of the signal transmitted from a communication terminal device (150), which is a communication terminal of the partner side, according to the environment noise level. A signal decoding unit (103) decodes encoded information transmitted from the communication terminal device (150) via a transmission path (110) and outputs the obtained signal as an output signal.

Abstract: A method of decoding symbols in which a first codeword has been spread by a second codeword to recover first information and second information is provided. The decoding occurs jointly, with an overall output determining both the first and second information. A first parallel code multiplying operation for each codeword of the second code is followed by a second parallel code multiplying operation for the first code. An overall maximum output of the second parallel code multiplying operations determines the output information.

Abstract: A method and apparatus include a voice activity detection module configured to detect silent frames, and a codec mode selection module configured to determine a codec mode. The voice activity detection module includes a receiver configured to receive a frame, a first determiner configured to determine a first set of parameters from the frame, and a providing unit configured to provide the first set of parameters to the codec mode selection module. The codec mode selection module includes a second determiner configured to determine a second set of parameters in dependence on the first set of parameters, and a selector configured to select a codec mode in dependence on the second set of parameters.

Abstract: A method and related apparatus comprising: buffering an encoded audio input signal comprising at least one combined signal of a plurality of audio channels and one or more corresponding sets of side information parameters describing a multi-channel sound image; changing the length of at least one audio frame of said combined signal by adding or removing a segment of said combined signal; modifying said one or more sets of side information parameters with a change corresponding to the change in the length of said at least one audio frame of said combined signal; and transferring said at least one audio frame of said combined signal with a changed length and said modified one or more sets of side information parameters to a further processing unit.

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: The present invention provides for processing voice data. The vocalic of at least one word associated with the electronic voice signal is elongated. The magnitude of at least one consonant spike of the at least one word associated with the electronic voice signal is increased. Through the emphasis of the consonants, intelligibility of speech is increased.

Abstract: [Problems] To provide a high-quality audio signal encoding technique by controlling the number of time/frequency groups in a frame. [Means for Solving Problems] An audio encoding device includes: a time group boundary candidate position extraction unit (101) for analyzing a sub-band signal (2001) obtained by frequency-changing an input signal and calculating a candidate position of the time group boundary; a time group quantity generation unit (103) for outputting a maximum value of the time group quantity; a time group selection unit (102) for generating a time group quantity not greater than the maximum time group quantity by using the candidate position; and a frequency group generation unit (104) for generating a frequency group by using the generated time group information. The device generates a time/frequency group accurately reflecting a change of the input signal and performs operations while controlling the number of time/frequency groups in the frame.

Abstract: An audio decoding device of the present invention includes: a decoding unit decoding a stream to a spectrum coefficient, and outputting stream information when a frame included in the stream cannot be decoded; an orthogonal transformation unit transforming the spectrum coefficient to a time signal; a correction unit generating a correction time signal based on an output waveform within a reference section that is in a section that overlaps between an error frame section to which the stream information is outputted and an adjacent frame section and that is a section in the middle of the adjacent frame section, when the decoding unit outputs the stream information: and an output unit generating the output waveform by synthesizing the correction time signal and the time signal.

Abstract: An embodiment of an apparatus for generating audio subband values in audio subband channels has an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function having a sequence of window coefficients to obtain windowed samples. The analysis window function has a first group of window coefficients and a second group of window coefficients. The first group of window coefficients is used for windowing later time-domain samples and the second group of window coefficients is used for windowing an earlier time-domain samples. The apparatus further has a calculator for calculating the audio subband values using the windowed samples.

Abstract: A low bit rate digital audio coding system includes an encoder which assigns codebooks to groups of quantization indexes based on their local properties resulting in codebook application ranges that are independent of block quantization boundaries. The invention also incorporates a resolution filter bank, or a tri-mode resolution filter bank, which is selectively switchable between high and low frequency resolution modes or high, low and intermediate modes such as when detecting transient in a frame. The result is a multichannel audio signal having a significantly lower bit rate for efficient transmission or storage. The decoder is essentially an inverse of the structure and methods of the encoder, and results in a reproduced audio signal that cannot be audibly distinguished from the original signal.

Abstract: An audio disk has a first audio zone and a second audio zone. In the first audio zone, both of a plurality of tracks each including a single piece of music and a first management information including first control information for allowing each track to be reproduced are recorded. The music is composed of audio data produced from either a linear PCM signal or a signal produced by applying lossless compression to the linear PCM signal. In the second audio zone, music composed of audio signal produced under lossy compression, a plurality of files each storing a signal of a single piece of music subjected to the lossy compression, and second management information including second control information for allowing each file to be reproduced are recorded. Each piece of music stored in each file corresponds to the music included in any one of the tracks in the first audio zone.

Abstract: Systems are disclosed for operating a communications network. The system includes a module to buffer frames of a signal, and a module to determine an access delay. The system also includes a module to compress a portion of the signal based on the access delay by removing a first portion of a frame of the signal and generating an overlap-added segment from a first segment and a second segment of the frame. In another embodiment, the system includes a module to buffer frames of a signal, a module to establish a communication channel with a handset, and a module to determine an access delay. The system also includes a module to compress a portion of the signal based on the access delay by removing a first portion of a frame of the signal and generating an overlap-added segment from a first segment and a second segment of the frame.

Abstract: A sound signal encoder for high efficiency encoding of sound signals from a plurality of channels is provided which includes a to-be-correlated object setter (52), to-be-correlated object selector (56) and a variable-length encoder (58). The to-be-correlated object setter (52) sets, on the basis of left-channel frequency information held in a left-channel frequency information holder (50) and right-channel frequency information held in a right-channel frequency information holder (51), index [i] indicating which ones of sine waves on the left channel are to be correlated with, namely, are to be subtracted from, sine waves on the right channel. The to-be-correlated object selector (56) selects a default value read from a storage unit (55) or index [i]-th amplitude information read from a left-channel amplitude information holder (53) as an object to be subtracted from the i-th amplitude information on the right channel according to the index [i].

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 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: To encode multi-channel digital data by adjusting the number of bits allocated to each channel to perform entropy coding of the multi-channel data, there is provided a multi-channel encoder including n encoders for audio data from n channels and an inter-channel bit allocator that allocates the number of bits usable for each channel on the basis of the provisional number of in-use bits from each of the encoders. Each of the encoders performs entropy coding on the basis of the provisional number of quantizing steps, outputs the provisional number of in-use bits resulting from summing of a code length of each unit of coding, and adjusts the number of in-use bits by updating the quantizing steps correspondingly to the number of bits supplied based on the provisional number of in-use bits.

Abstract: A data de-shuffler includes a buffer having a set of addressable locations for storing data and control circuitry for de-shuffling a sequence of shuffled data samples. The control circuitry stores a first data sample of the sequence of shuffled data samples at a first location in the buffer with a first address generated from an entry in a look up table and stores a second data sample at a second location in the buffer with a second address generated by incrementing from the first address by a selected incrementation value. The first and second locations in the buffer place the first and second samples in corresponding positions in an un-shuffled sequence of samples.

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 procession of the signal would be inaudible or minimally audible. Further alternatives provide for multiple channels of audio.

Abstract: Systems and processes for transmission of multi-channel audio from a sender to one or more recipients. Multi-channel audio is encoded with a plurality of different dialog channels. The encoded multi-channel audio and dialog channels can be compressed to facilitate transmission with limited bandwidth. A recipient can select a desired dialog channel from the plurality of dialog channels transmitted. A receiver side decoder can reconstruct a multi-channel audio with the selected dialog for playback. The plurality of dialog options can include different languages, different dialects, different accents, and/or different viewpoints/biases.

Abstract: Techniques and tools for selectively using multiple entropy models in adaptive coding and decoding are described herein. For example, for multiple symbols, an audio encoder selects an entropy model from a first model set that includes multiple entropy models. Each of the multiple entropy models includes a model switch point for switching to a second model set that includes one or more entropy models. The encoder processes the multiple symbols using the selected entropy model and outputs results. Techniques and tools for generating entropy models are also described.

Abstract: A scalable audio codec encodes an input audio signal as a base layer at a high compression ratio and one or more residual signals as an enhancement layer of a compressed bitstream, which permits a lossless or near lossless reconstruction of the input audio signal at decoding. The scalable audio codec uses perceptual transform coding to encode the base layer. The residual is calculated in a transform domain, which includes a frequency and possibly also multi-channel transform of the input audio. For lossless reconstruction, the frequency and multi-channel transforms are reversible.

Abstract: An encoder/decoder for multi-channel audio data, and in particular for audio reproduction through wave field synthesis. The encoder comprises a two-dimensional filter-bank to the multi-channel signal, in which the channel index is treated as an independent variable as well as time, and and the resulting spectral coefficient are quantized according to a two-dimensional psychoacoustic model, including masking effect in the spatial frequency as well as in the temporal frequency. The coded spectral data are organized in a bitstream together with side information containing scale factors and Huffman codebook identifiers.

Abstract: An “adaptive audio playback controller” operates by decoding and reading received packets of an audio signal into a signal buffer. Samples of the decoded audio signal are then played out of the signal buffer according to the needs of a player device. Jitter control and packet loss concealment are accomplished by continuously analyzing buffer content in real-time, and determining whether to provide unmodified playback from the buffer contents, whether to compress buffer content, stretch buffer content, or whether to provide for packet loss concealment for overly delayed or lost packets as a function of buffer content. Further, the adaptive audio playback controller also determines where to stretch or compress particular frames or signal segments in the signal buffer, and how much to stretch or compress such segments in order to optimize perceived playback quality.

Abstract: The present invention provides method and device for transcoding between audio coding formats with different time-frequency analysis domains, as used for example by MPEG-AAC and mp3, particularly for facilitated and faster transcoding between such audio signals. A method for transcoding a framed audio signal from a first parameter domain into a second parameter domain comprises linearly transforming two or more parameters of the first parameter domain to at least one parameter of the second parameter domain, wherein the two or more parameters of the first parameter domain come from different frames of the audio signal in the first parameter domain. The linear transformation can be described as a matrix and implemented as a look-up table.

Abstract: Techniques and tools related to delayed or lost coded audio information are described. For example, a concealment technique for one or more missing frames is selected based on one or more factors that include a classification of each of one or more available frames near the one or more missing frames. As another example, information from a concealment signal is used to produce substitute information that is relied on in decoding a subsequent frame. As yet another example, a data structure having nodes corresponding to received packet delays is used to determine a desired decoder packet delay value.

Abstract: This invention relates to a method, a device and a software application product for N-level quantization of vectors, wherein N is selectable prior to said quantization from a set of at least two pre-defined values that are smaller than or equal to a pre-defined maximum number of levels M. A reproduction vector for each vector is selected from an N-level codebook of N reproduction vectors that are, for each N in said set of at least two pre-defined values, represented by the first N reproduction vectors of the same joint codebook of M reproduction vectors. The invention further relates to a method, a device and a software application product for retrieving reproduction vectors for vectors that have been N-level quantized, to a system for transferring representations of vectors, to a method, a device and a software application product for determining a joint codebook, and to such a joint codebook itself.

Abstract: A system, method and computer-readable medium are disclosed for operating a communications network. The method aspect comprises receiving a signal and removing a first portion of a frame of the signal, and generating an overlap-added segment from (1) a first segment of the frame, the first segment being located before the first portion; and (2) a second segment of the frame, the second segment comprising an endmost portion of a terminal section of the frame. The method preferably operates in a discontinuous transmission packet telephony network having a channel access delay.

Abstract: Encoding an audio signal is provided wherein the audio signal includes a first audio channel and a second audio channel, the encoding comprising subband filtering each of the first audio channel and the second audio channel in a complex modulated filterbank to provide a first plurality of subband signals for the first audio channel and a second plurality of subband signals for the second audio channel, downsampling each of the subband signals to provide a first plurality of downsampled subband signals and a second plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, deriving spatial parameters from the sub-subband signals and from those downsampled subband signals that are not further subband filtered, and deriving a single channel audio signal comprising derived subband signals derived from the first plurality of downsampled subband signals and the second plurality of

Abstract: Compander architecture and method for ensuring that the components of an output signal are synchronized includes calculating a gain calculation for an input signal, detecting a predetermined condition such as a zero crossing or absence of a zero crossing within a specified period, and ensuring that the result of the gain calculation is providing to the system synchronously with the input signal. The input signal may be divided into one or more signals for processing, and the gain calculation may include one or more power estimations and one or more gain cells associated with the multiple signals, and may further include variable attack and release.

Abstract: A device, method, and system are disclosed. In one embodiment the device includes a first virtual machine to directly access a physical audio codec. The device also includes a virtual audio codec that is managed by the first virtual machine. The virtual audio codec can provide a custom interface to the physical audio codec for one or more additional virtual machines apart from the first virtual machine.