Abstract: A sound coding device having a monaural/stereo scalable structure and capable of efficiently coding stereo sound. even when the correlation between the channel signals of a stereo signal is small. In a core layer coding block of this device, a monaural signal generating section generates a monaural signal from first and second-channel sound signal, a monaural signal coding section codes the monaural signal, and a monaural signal decoding section greatest a monaural decoded signal from monaural signal coded data and outputs it to an expansion layer coding block. In the expansion layer coding block, a first-channel prediction signal synthesizing section synthesizes a first-channel prediction signal from the monaural decoded signal and a first-channel prediction filter digitizing parameter and a second-channel prediction signal synthesizing section synthesizes a second-channel prediction signal from the monaural decoded signal and second-channel prediction filter digitizing parameter.

Abstract: An energy corrector (105) for correcting a target energy for high-frequency components and a corrective coefficient calculator (106) for calculating an energy corrective coefficient from low-frequency subband signals are newly provided. These processors perform a process for correcting a target energy that is required when a band expanding process is performed on a real number only. Thus, a real subband combining filter and a real band expander which require a smaller amount of calculations can be used instead of a complex subband combining filter and a complex band expander, while maintaining a high sound-quality level, and the required amount of calculations and the apparatus scale can be reduced.

Abstract: Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels.

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

Abstract: When encoding an audio signal, it is possible to efficiently encode the audio signal while maintaining high register signal components, and prevent deterioration of sound quality of decoded signal. A digital audio signal is divided into a plurality of frequency bands. The digital audio signal having been divided into each band is function-approximated for each divided band. Further, parameters of function having been function-approximated are encoded. When performing decoding process, parameters of the function of each band are used to perform function interpolation, synthesize the function-interpolated signal of each band interpolated, and decode the signal. Thus, when function-approximating each band, by suitably setting the function equation, it is possible to perform an encoding process while maintaining the high register components and perform a compression-coding process which enables reproduction with very good sound quality.

Abstract: The present invention relates to an encoding device for saving the number of bits of codes. In step S11, the differential value between a normalization coefficient Bi to be encoded and a normalization coefficient Bi-1 for an encoding unit Ai-1 in a band adjacent to the lower side of an encoding unit Ai corresponding to the normalization coefficient Bi is computed. In step S12, reference is made to a table in which a differential value having a high frequency of occurrence is associated with a code having a small number of bits, and a code corresponding to the computed differential value is read. In step S13, it is determined whether or not all normalization coefficients B have been encoded. If it is determined that all normalization coefficients B have been encoded, in step S14, the code read in step S12 is output. The present invention is applicable to an audio recorder.

Abstract: A method of encoding multi-channel signals having two or more channels into a first signal and a second signal, and an apparatus to perform the method, the method including generating the first signal by performing a first operation using a first channel signal in the multi-channel signals; and generating the second signal by performing a second operation using a combination of the first channel signal and a second channel signal in the multi-channel signals.

Abstract: The present invention provides a computationally efficient technique for compression encoding of an audio signal, and further provides a technique to enhance the sound quality of the encoded audio signal. This is accomplished by including more accurate attack detection and a computationally efficient quantization technique. The improved audio coder converts the input audio signal to a digital audio signal. The audio coder then divides the digital audio signal into larger frames having a long-block frame length and partitions each of the frames into multiple short-blocks. The audio coder then computes short-block audio signal characteristics for each of the partitioned short-blocks based on changes in the input audio signal.

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: A transient in a digital audio signal can be detected by generating a first set of spectral characteristics associated with a first portion of the digital audio signal and a second set of spectral characteristics associated with a second portion of the digital audio signal, wherein the first and second portions of the digital audio signal partially overlap, comparing values in the first set of spectral characteristics with corresponding values in the second set of spectral characteristics to generate a set of ratios, weighting the set of ratios, and analyzing at least a portion of the weighted set of ratios to detect a transient associated with the first portion of the digital audio signal. Further, an indicator identifying the presence of a detected transient can be output. Additionally, one or more ratios in the set of ratios can be weighted based on amplitude, frequency, or a power function.

Abstract: A method for processing audio data includes determining a first common scalefactor value for representing quantized audio data in a frame. A second common scalefactor value is determined for representing the quantized audio data in the frame. A line equation common scalefactor value is determined from the first and second common scalefactor values.

Abstract: The present invention is based on the finding that parameters including: a first set of parameters of a representation of a first portion of an original signal and a second set of parameters of a representation of a second portion of the original signal can be efficiently encoded when the parameters are arranged in a first sequence of tuples and a second sequence of tuples. The first sequence of tuples includes tuples of parameters having two parameters from a single portion of the original signal and the second sequence of tuples includes tuples of parameters having one parameter from the first portion and one parameter from the second portion of the original signal. A bit estimator estimates the number of necessary bits to encode the first and the second sequence of tuples. Only the sequence of tuples, which results in the lower number of bits, is encoded.

Abstract: Methods and apparatuses for encoding and decoding an audio signal are provided, a method of encoding an audio signal including: receiving the audio signal including information about a moving sound source; receiving position information about the moving sound source; generating dynamic track information indicating motion of the moving sound source by using the position information; and encoding the audio signal and the dynamic track information.

Abstract: An original digital audio signal represents unimpaired audio information. The digital audio signal is compressed and encrypted without substantial impairment to produce a first audio signal. A second, unencrypted, audio signal is produced. The first and second audio signals are combined. Preferably the first audio signal is losslessy compressed. The second audio signal may be an impaired version of the original digital audio signal.

Abstract: At an audio encoder, cue codes are generated for one or more audio channels, wherein a combined cue code (e.g., a combined inter-channel correlation (ICC) code) is generated by combining two or more estimated cue codes, each estimated cue code estimated from a group of two or more channels. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels. Received cue codes include a combined cue code (e.g., a combined ICC code). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein two or more derived cue codes are derived from the combined cue code, and each derived cue code is applied to generate two or more synthesized channels.

Abstract: Provided are a method and apparatus for structuring a bitstream for an object-based audio service, and an apparatus for encoding the bitstream. A method of structuring a bitstream, may include: configuring the bitstream by separating the bitstream into a file header and frames of audio objects that are separated using a sound source separation scheme; and storing, in the file header, reproduction level information of audio objects.

Abstract: An improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal.

Abstract: A transient detector (100) analyzes (110) a given frame n of the input audio signal to determine, based on audio signal characteristics of the given frame n, a transient hangover indicator for a following frame n+1, and signals (120) the determined transient hangover indicator to an associated audio encoder (10) to enable proper encoding of the following frame n+1.

Abstract: A system for automatically trimming an audio files based upon textual content associated with the audio file is provided. The source of the textual content may be an electronic document or written language text. The textual content may include predefined hints, a text mark, or end-of-phrase punctuation mark. The system generates a trimming instruction based upon textual content corresponding to the audio file, and the audio file is trimmed based upon the trimming instruction.

Abstract: A hybrid audio encoding technique incorporates both ABR, or CBR, and VBR encoding modes. For each audio coding block, after a VBR quantization loop meets the NMR target, a second quantization loop might be called to adaptively control the final bitrate. That is, if the NMR-based quantization loop results in a bitrate that is not within a specified range, then a bitrate-based CBR or ABR quantization loop determines a final bitrate that is within the range and is adaptively determined based on the encoding difficulty of the audio data. Excessive bitrates from use of conventional VBR mode are eliminated, while still providing much more constant perceptual sound quality than use of conventional CBR mode can achieve.

Abstract: The present invention relates to improvements of predictive encoding/decoding operations performed on a signal which is transmitted over a packet switched network. The signal is encoded on a block by block basis in such way that a block A-B is predictive encoded independently of any preceding blocks. A start state (715) located somewhere between the end boundaries A and B of the block is encoded using any applicable coding method. Both block parts surrounding the start state is then predictive encoded based on the start state and in opposite directions with respect to each other, thereby resulting in a full encoded representation (745) of the block A-B. At the decoding end, corresponding decoding operations are performed.

Abstract: A method for decoding an audio signal comprises receiving a combined downmix, a combined object information, and a mix information, the combined downmix being generating using at least two downmix signals, the combined object information being made by combination of at least two sets of object information, generating a downmix processing information using the combined object information and the mix information, and processing the combined downmix using the downmix processing information. The method and an apparatus for decoding an audio signal comprising the combined downmix and the combined object information can control object gain and output in a remote conference and so on. The method and the apparatus for decoding audio signal that contains multi-object signals are fast and efficiently by reducing process time, computer resource, thereby relieving the resource requirement like the wide bandwidth by using the combined object information.

Abstract: An audio object coder for generating an encoded object signal using a plurality of audio objects includes a downmix information generator for generating downmix information indicating a distribution of the plurality of audio objects into at least two downmix channels, an audio object parameter generator for generating object parameters for the audio objects, and an output interface for generating the imported audio output signal using the downmix information and the object parameters. An audio synthesizer uses the downmix information for generating output data usable for creating a plurality of output channels of the predefined audio output configuration.

Abstract: In an audio signal processing procedure, auto-regressive (AR) modeling is used to create a residual signal from an input audio signal. The residual signal is further added to the input audio in order to produce a processed output audio signal. The AR modeling can be performed frame-by-frame or sample-by-sample employing frequency warped Burg's method.

Abstract: A mobile terminal and method of eliminating a call sound noise thereof are disclosed, by which a user can have a call of a clear sound quality by removing a frame including a white noise from a voice frame received from an originator terminal. The present invention includes a communicating unit receiving a voice frame; a control unit deleting the voice frame, if a white noise is included in the voice frame; an audio processing unit decoding the voice frame under the control of the control unit.

Abstract: An apparatus for processing a signal and method thereof are disclosed. Data coding and entropy coding are performed with interconnection, and grouping is used to enhance coding efficiency. The present invention includes the steps of obtaining a group reference value corresponding to a plurality of data included in one group through grouping and a difference value corresponding to the group reference value and obtaining the data using the group reference value and the difference value.

Abstract: The invention relates to a method for supporting an encoding of an audio signal, wherein at least one section of the audio signal is to be encoded with a coding model that allows the use of different coding frame lengths. In order to enable a simple selection of the respectively best suited coding frame length, it is proposed that at least one control parameter is determined based on signal characteristics of the audio signal. The control parameter is then used for limiting the options of possible coding frame lengths for the at least one section. The invention relates equally to a module 10,11 in which this method is implemented, to a device 1 and a system comprising such a module 10,11, and to a software program product including a software code for realizing the proposed method.

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: Provided are an audio encoding device and an audio decoding device, by which optimal trade-off between code rates and sound quality can be flexibly adjusted. A variable frequency segmentation encoding unit includes: difference degree calculation units for calculating a difference degree between first and second input signals depending on a segmentation method for segmenting a frequency band into sub-bands; a selection unit for selecting one of the segmentation methods; and a difference degree and segmentation information encoding unit for encoding the selected method and the difference degree for each sub-band. A variable frequency segment decoding unit includes: a segmentation information decoding unit for decoding the segmentation information to learn the segmentation method; a switching unit for outputting a difference degree code corresponding to the segmentation method; and difference degree decoding units for decoding the difference degree code to the difference degree for each sub-band.

Abstract: An encoding method includes extracting background noise characteristic parameters within a hangover period; for a first superframe after the hangover period, performing background noise encoding based on the extracted background noise characteristic parameters; for superframes after the first superframe, performing background noise characteristic parameter extraction and DTX decision for each frame in the superframes after the first superframe; and for the superframes after the first superframe, performing background noise encoding based on extracted background noise characteristic parameters of the current superframe, background noise characteristic parameters of a plurality of superframes previous to the current superframe, and a final DTX decision. Also, a decoding method and apparatus and an encoding apparatus are disclosed.

Abstract: An audio encoder, which is capable of encoding multiple-channel signals so that only a downmixed signal is decoded and of further generating specific auxiliary information necessary for dividing the downmixed signal, is provided. An audio encoder (10), which compresses and encodes audio signals of N-channels (N>1), includes a downmixed signal encoding unit (11) which encodes the downmixed signal obtained by downmixing the audio signals, and an auxiliary information generation unit (12a) which generates auxiliary information necessary for decoding the downmixed signal encoded by the downmixed signal encoding unit (11) into N-channel audio signals.

Abstract: Blocks of audio information are arranged in groups that share encoding control parameters to reduce the amount of side information needed to convey the control parameters in an encoded signal. The configuration of groups that reduces the distortion of the encoded audio information may be determined by any of several techniques that search for an optimal or near optimal solution. The techniques include an exhaustive search, a fast optimal search and a greed merge, which allow the search technique to tradeoff the reduction in distortion against the bit rate of the encoded signal and/or the computational complexity of the search technique.

Abstract: An audio encoding scheme or a stream that encodes audio and video data is disclosed. The scheme has particular application in mezzanine-level coding in digital television broadcasting. The scheme has a mean effective audio frame length F that equals the video frame length 1/fV over an integral number M video frames, by provision of audio frames variable in length F in a defined sequence where length=F(j) at encoding. The length of the audio frames may be varied by altering the length of overlap between adjacent frames in accordance with an algorithm that repeats after a sequence of M frames. An encoder and a decoder for such a scheme are also disclosed.

Abstract: A multi-channel audio encoder (10) encodes an N-channel audio signal. A first unit (110) generates a first encoded M-channel signal, e.g. a spatial stereo down-mix, for the N-channel signal (N>M). Down-mixers (115, 116, 117) generate first enhancement data for the signal relative to the N-channel audio signal. A second M-channel signal, such as an artistic stereo mix, is generated for the N-channel signal. A processor (123) then generates second enhancement data for the second M-channel signal relative to the first M-channel signal. A second unit (120) generates an output signal comprising the second M-channel signal, the first enhancement data and the second enhancement data. The generator (123) can dynamically select between generating the second enhancement data as absolute enhancement data or as relative enhancement data relative to the second encoded M-channel signal.

Abstract: An encoding device (1) and method convert a set of signals (l, r) into a dominant signal (m) containing most signal energy, a residual signal (s) containing a remainder of the signal energy, and signal parameters (IID, ICC) associated with the conversion. The dominant signal (m) and selected parts of the residual signal (s) are encoded. Selecting parts of the residual signal involves a residual signal (s?) passing perceptually relevant parts of the residual signal (s), attenuating perceptually less relevant parts of the residual signal and suppressing least relevant parts of the residual signal. An associated decoding device (2) and method decode the encoded dominant signal and the encoded residual signal so as to produce a decoded dominant signal (m?u) and a decoded residual signal (s?mod) respectively. A synthetic residual signal (s?Syn) is derived from the decoded dominant signal (m?u) and is attenuated so as to produce an attenuated synthetic residual signal (S?Syn,mod).

Abstract: The perceptual scalable audio coding/decoding technique lies in the use of a psychoacoustic mask to guide residue coding in enhancement layer coders. At the encoder, a psychoacoustic mask is calculated for the enhancement layer coders or is simply extracted from the coded base layer bitstream. One can also decode the coded base layer bitstream into the audio waveform, and calculate the psychoacoustic mask from the decoded base layer waveform. Furthermore, a predictive technology can be used to refine the psychoacoustic mask derived from the base layer bitstream to form a more accurate psychoacoustic mask of the enhancement layer. In addition, one can calculate the enhancement layer psychoacoustic mask from the original audio, and send the difference between the enhancement layer psychoacoustic mask and the base layer psychoacoustic mask as side information to the decoder. This psychoacoustic mask may then be used for the perceptual coding and decoding of the residue.

Abstract: In one embodiment, at least one audio data frame having at least one channel is generated. Each channel is divided into a plurality of blocks. A sub-block partitioning scheme is selected, and a number of sub-blocks into which the block is to be partitioned is selected. The selected number of sub-blocks is chosen from numbers of sub-blocks available for the selected sub-block partitioning scheme. The block of audio data is partitioned into sub-blocks according to the selected number of sub-blocks, and the partitioned sub-blocks are coded according to a selected entropy coding scheme.

Abstract: A parametric model is used for error concealment. The model filter allows for recovering signal components of original audio channel signals that now are lost or erroneous from signal components of at least one other audio channel. During error-free reception of valid frames, the parameters of that model will be derived and stored. In case of frame loss or frame error affecting the multi-channel information, a conjecture of the missing information is recovered by applying the model, using the stored parameters. In case of several subsequent lost or erroneous frames, it is possible either to use the parameters derived during the last valid frame or to use parameters derived from the recovered multi-channel information of the respective previous invalid frame. Furthermore, if there are long sequences of lost frames, it can be beneficial to apply some gradual muting of the model parameters, which essentially results in a gradual attenuation of the recovered multi-channel information.

Abstract: Scalable stereo audio coding and decoding method and apparatus are provided. The scalable stereo audio coding method includes transforming a first channel and a second channel audio samples; quantizing the transformed first channel and a second channel audio samples; and coding the quantized first channel audio samples up to a predetermined transition layer and then interleavingly coding the quantized first and second channel audio samples with increasing a layer index from a layer succeeding the transition layer, until coding for a predetermined plurality of layers is finished.

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: An audio encoder receives multi-channel audio data comprising a group of plural source channels and performs channel extension coding, which comprises encoding a combined channel for the group and determining plural parameters for representing individual source channels of the group as modified versions of the encoded combined channel. The encoder also performs frequency extension coding. The frequency extension coding can comprise, for example, partitioning frequency bands in the multi-channel audio data into a baseband group and an extended band group, and coding audio coefficients in the extended band group based on audio coefficients in the baseband group. The encoder also can perform other kinds of transforms. An audio decoder performs corresponding decoding and/or additional processing tasks, such as a forward complex transform.

Abstract: In one embodiment, at least first and second channels in a frame of the audio signal are independently subdivided into blocks if the first and second channels are not correlated with each other. The first and second channels are corresponding subdivided into blocks such that the lengths of the blocks into which the second channel is subdivided correspond to the lengths of the blocks into which the first channel is subdivided if the first and second channels are correlated with each other. First information may be generated to indicate whether the first and second channels are independently subdivided or correspondingly subdivided.

Abstract: An apparatus for decoding audio data that is capable of reducing the amount of calculations that are performed during the arithmetic decoding of an audio signal coded by bit sliced arithmetic coding (BSAC) to improve the performance of a decoder and a method thereof are provided.

Abstract: Presented herein are system(s) and method(s) for handling audio jitters. In one embodiment; there is presented a method for decoding an audio signal. The method comprises receiving a portion of the audio signal, the portions of the audio signal associated with a time stamp; comparing the time stamp associated with the portion of the audio signals to a reference time; generating another portion of the audio signal, if the time stamp is later than the time reference by over a certain margin or error; and dewindowing the another portion with a previously played portion of the audio signal, thereby resulting in a an another dewindowed portion.

Abstract: A scalable audio data arithmetic decoding method, medium, and apparatus, and a method, medium, and apparatus truncating an audio data bitstream. The arithmetic decoding method of decoding a scalable arithmetic coded symbol may include arithmetic decoding of a symbol by using the symbol and a probability value for the symbol desired to be decoded, and determining whether or not to continue decoding by checking an ambiguity indicating whether or not decoding of the symbol to be decoded is completed. According to a method, medium, and apparatus of the present invention, data to which scalability is applied when arithmetic coding is performed in MPEG-4 scalable lossless audio coding can be efficiently decoded. Even when a bitstream is truncated, a decoding termination point can be known such that additional decoding of the truncated part can be performed.

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: The invention provides an efficient technique for encoding a multi-channel audio signal. The invention relies on the principle of encoding (S1) a signal representation of one or more of the multiple channels in a first encoding process, and encoding another signal representation of one or more channels in a second, filter-based encoding process. A basic idea according to the invention is to select (S2), for the second encoding process, a combination of i) frame division configuration of an overall encoding frame into a set of sub-frames, and ii) filter length for each sub-frame, according to a predetermined criterion. The second signal representation is then encoded (S3) in each sub-frame of the overall encoding frame according to the selected combination. The possibility to select frame division configuration and at the same time adjust the filter length for each sub-frame provides added degrees of freedom, and generally results in improved performance.

Abstract: An apparatus and method of audio encoding and decoding based on a Variable Bit Rate (VBR) is provided. The audio encoding and decoding apparatus and method may determine an optimum bit rate per superframe and per frame, determine an optimum encoding mode by applying an open-loop mode/closed-loop mode based on a characteristic of an audio signal, and perform indexing based on the optimum encoding mode.

Abstract: An encoding device (1) for converting a first number (M) of input audio channels into a second, smaller number (N) of output audio channels comprises at least one conversion unit (12) for converting a first signal (Lf; Rf; Co) and a second signal (Lr; Rr; Le) into a third signal (L; R; C) and a fourth signal (Ls; Rs; Cs). The third, dominant signal contains most of the signal energy of the first and second signals, while the fourth, residual signal contains the remainder of said signal energy. The encoding device is arranged for using the third signal (L; R; C) to produce an output signal and for outputting the fourth signal (Ls; Rs; Cs). A decoding device (2) for converting a first number (N) of input audio channels into a second, larger number (M) or output audio channels comprises at least one conversion unit (24) for converting a first signal (L; R; C) and a second signal (Ld; Rd; Ld) into a third signal (Lf, Rf; Co) and a fourth signal (Lr; Rr; Le).

Abstract: Polyphonic signals are used to create a main signal, typically a mono signal, and a side signal. A number of encoding schemes for the side signal are provided. Each encoding scheme is characterized by a set of sub-frames of different lengths. The total length of the sub-frames corresponds to the length of the encoding frame of the encoding scheme. The encoding scheme to be used on the side signal is selected dependent on the present signal content of the polyphonic signals. In a preferred embodiment, a side residual signal is created as the difference between the side signal and the main signal scaled with a balance factor. The balance factor is selected to minimize the side residual signal. The optimized side residual signal and the balance factor are encoded and provided as encoding parameters representing the side signal.