Abstract: An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes operating on signal representations of a set of audio input channels. Local synthesis is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding in a further encoding process, preferably based on correlation between the residual error signals.

Abstract: One or more attributes (e.g., pan, gain, etc.) associated with one or more objects (e.g., an instrument) of a stereo or multi-channel audio signal can be modified to provide remix capability. In some implementations, a method can include obtaining a first plural-channel audio signal having one or more objects; obtaining side information, at least some of which represents a relation between the first plural-channel audio signal and the one or more objects; obtaining a set of mix parameters; and generating a second plural-channel audio signal using the side information and the set of mix parameters.

Abstract: An encoder includes, a degree-of-importance calculating unit that calculates a degree of importance of each of a first number of signals included in input signals; a signal converting unit that converts the first number of signals included in the input signals into a second number of signals; a degree-of-importance converting unit that converts a first number of degrees of importance, a number of which is equal to the first number of signals, calculated by the degree-of-importance calculating unit into a second number of degrees of importance, a number of which is equal to the second number of signals; a number-of-bits determining unit that determines a number of bits for use in quantizing each of the second number of signals obtained by the conversion performed by the signal converting; and a quantizing unit that quantizes each of the second number.

Abstract: A multi-channel audio signal transmitting apparatus of the present invention includes an encoding unit configured to encode a predetermined audio signal and a controller configured to transmit the encoded audio signal to a predetermined device based on an HDAI (high definition audio interface) protocol.

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 extracting spatial information and a downmix signal from a bitstream; and generating rendering information by using the spatial information and filter information having a surround effect, wherein the rendering information comprises first rendering information applied to one channel of the downmix signal extracted from the bitstream and then transmitted on the same channel and second rendering information applied to the channel and then transmitted on another channel.

Abstract: An audio multi-standard detection apparatus is provided to pre-record a plurality of audio broadcasting format carrier bands. The detection apparatus includes: a carrier calculation unit and a switch controller. The carrier calculation unit is used for receiving an intermediate frequency signal and calculates the first carrier energy of the intermediate frequency signal in each of the frequency bands. From this first carrier energy, the switch controller determines which frequency bands are existed in the intermediate frequency signal, then analyzes the signal via at least one threshold value to determine the frequency band's primary carrier. The switch controller obtains an effective carrier. Further, the switch controller compares the effective carrier with the pre-recorded carrier bands for selectively outputting the appropriate audio broadcasting format.

Abstract: Representations of spatial audio scenes using higher-order Ambisonics HOA technology typically require a large number of coefficients per time instant. This data rate is too high for most practical applications that require real-time transmission of audio signals. According to the invention, the compression is carried out in spatial domain instead of HOA domain. The (N+1)2 input HOA coefficients are transformed into (N+1)2 equivalent signals in spatial domain, and the resulting (N+1)2 time-domain signals are input to a bank of parallel perceptual codecs. At decoder side, the individual spatial-domain signals are decoded, and the spatial-domain coefficients are transformed back into HOA domain in order to recover the original HOA representation.

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: A system extends the high-frequency spectrum of a narrowband audio signal in the time domain. The system extends the harmonics of vowels by introducing a non linearity in a narrow band signal. Extended consonants are generated by a random-noise generator. The system differentiates the vowels from the consonants by exploiting predetermined features of a speech signal.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The preset invention includes receiving object information and a downmix signal including at least one object, the object information comprising data type information and at least one of object gain and object gain ratio; determining whether preset information to render the object is included in the extension region of the object information, based on the data type information; generating initial preset information to render the object by using at least one of the object gain and the object gain ration, if the preset information is not included in the object information; and rendering the object being included the downmix signal by using the initial preset information. Accordingly, even if preset information is not received, it is able to reconstruct an audio signal by generating initial preset information using received object information.

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

Abstract: A computer-implemented method of encoding audio includes accessing a plurality of independent audio source streams, each of which includes a sequence of source frames. Respective source frames of each sequence include respective pluralities of pulse-code modulated audio samples. Each of the plurality of independent audio source streams is separately encoded to generate a plurality of independent encoded streams, each of which corresponds to a respective independent audio source stream. The encoding includes, for respective source frames, converting respective pluralities of pulse-code modulated audio samples to respective pluralities of floating-point frequency samples that are divided into a plurality of frequency bands. An instruction to mix the plurality of independent encoded streams is received; in response, respective floating-point frequency samples of the independent encoded streams are combined. An output bitstream is generated that includes the combined respective floating-point frequency samples.

Abstract: Provided are a method and apparatus for encoding and decoding a stereo signal or a multi-channel signal. According to the method and apparatus, a stereo signal or a multi-channel signal can be encoded and/or decoded by generating parameters based on a mono signal.

Abstract: Provided is a multi-channel acoustic signal processing device by which loads of arithmetic operations are reduced. The multi-channel acoustic signal processing device includes: a decorrelated signal generation unit, and a matrix operation unit and a third arithmetic unit. The decorrelated signal generation unit generates a decorrelated signal w? indicating a sound which includes a sound indicated by an input signal x and reverberation, by performing reverberation processing on the input signal x. The matrix operation unit and the third arithmetic unit generate audio signals of m channels, by performing arithmetic operation on the input signal x and the decorrelated signal w? generated by the decorrelated signal generation unit, using a matrix R3 which indicates distribution of a signal intensity level and distribution of reverberation.

Abstract: The purpose of the invention is to bridge the gap between parametric multi-channel audio coding and matrixed-surround multi-channel coding by gradually improving the sound of an up-mix signal while raising the bit-rate consumed by the side-information starting from 0 up to the bit-rates of the parametric methods. More specifically, it provides a method of flexibly choosing an “operating point” somewhere between matrixed-surround (no side-information, limited audio quality) and fully parametric reconstruction (full side-information rate required, good quality). This operating point can be chosen dynamically (i.e. varying over time) and in response to the permissible side-information rate, as it is dictated by the individual application.

Abstract: A headphone down mix signal can be efficiently derived from a parametric down mix of a multi-channel signal, when modified HRTFs (head related transfer functions) are derived from HRTFs of a multi-channel signal using a level parameter having information on a level relation between two channels of the multi-channel signals such that a modified HRTF is stronger influenced by the HRTF of a channel having a higher level than by the HRTF of a channel having a lower level. Modified HRTFs are derived within the decoding process taking into account the relative strength of the channels associated to the HRTFs. The HRTFs are thus modified such that a down mix signal of a parametric representation of a multi-channel signal can directly be used to synthesize the headphone down mix signal without the need of an intermediate full parametric multi-channel reconstruction of the parametric down mix.

Abstract: An apparatus and method for encoding and decoding a spatial parameter are provided. A spatial parameter encoding apparatus may encode a spatial parameter using a correlation between spatial parameters indicating a characteristic relationship between channels of a multi-channel audio signal, so that the multi-channel audio signal may be efficiently encoded.

Abstract: An audio system that provides an optimized sense of realism and an optimized three-dimensional effect to a user, the audio system including: a horizontal plane speaker set including a center channel speaker that is disposed at a reference point, a left channel speaker and a right channel speaker that are disposed at points in right and left directions of the reference point, and at least one first channel speaker and at least one second channel speaker that are disposed in right and left directions of the reference point, respectively; and a vertical plane speaker set including a left height channel speaker and a right height channel speaker that are respectively disposed in right and left directions of the reference point and in a vertical direction of the reference point, and a center height channel speaker that is disposed in the vertical direction of the reference point.

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

Abstract: 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: A wireless multi-channel audio system including an audio source with a wireless transceiver configured to communicate according to a standard wireless protocol and an audio controller, which are collectively configured to establish wireless communications with multiple audio sinks via a corresponding wireless link, to assign each audio sink a corresponding audio channel, to synchronize timing with each audio sink, and to transmit audio information for each audio channel to a corresponding audio sink via a corresponding wireless link. The audio source may inquire as to supported audio parameters, such as sample rate and audio codec, and selects a commonly supported configuration. The audio source may separate audio information into queues for each audio channel for each audio sink. The audio source transmits frames with timestamps and a common start time for synchronization, and the audio sinks use this information to synchronize timing and remain virtually synchronized with each other.

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: Techniques are described for combining parametric multi-channel audio coding with matrixing, reconstructing a full-quality multi-channel, independent of the decoder. A stereo signal is obtained from encoding an N-channel audio signal into spatial parameters and a stereo down-mix signal having first and second stereo signals, including adding a first signal and a third signal to obtain a first output signal, the first signal having the first stereo signal modified by a first complex function, the third signal having the second stereo signal modified by a third complex function. A second signal and fourth signal are similarly added to obtain a second output signal. Complex functions are chosen such that an energy value of the difference between first signal and the second signals (fourth signal and third signals) is larger than or equal to the energy value of the sum of the first and the second signal (fourth signal and third signal).

Abstract: A method and a device are described for processing a stereo signal obtained from an encoder, which encodes an N-channel audio signal into spatial parameters (P) and a stereo down-mix comprising first and second stereo signals (LO, RO). A first signal and a third signal are added in order to obtain a first output signal (L0w), wherein the first signal (L0wL) comprises the first stereo signal (LO) modified by a first complex function (g1), and the third signal (L0wR) comprises the second stereo signal (RO) modified by a third complex function (g3). A second signal and a fourth signal are added to obtain a second output signal (R0w). The fourth signal (R0wR) comprises the second stereo signal (RO) modified by a fourth complex function (g4), and the second signal (R0wL) comprises the first stereo signal (LO) modified by a second complex function (g2).

Abstract: A method and apparatus of effectively encoding and decoding a high-frequency signal of a multi-channel audio are provided. A multi-channel audio decoding apparatus may down-mix a multi-channel audio input signal, expand a number of channels of the down-mixed signal, select at least one of the expanded channel signal, extract a parameter indicating a characteristic relation between the selected signal and the multi-channel audio input signal, and encode the down-mixed signal and the extracted parameter.

Abstract: According to the present invention, multiple parametrically encoded audio signals can be efficiently combined using an audio signal generator, which generates an audio output signal by combining the down-mix channels and the associated parameters of the audio signals directly within the parameter domain, i.e. without reconstructing or decoding the individual input audio signals prior to the generation of the audio output signal. This is achieved by direct mixing of the associated down-mix channels of the individual input signals. It is one key feature of the present invention that the combination of the down-mix channels is achieved by simple, computationally inexpensive arithmetic operations.

Abstract: A method and apparatus for distributing stereo signal of an audio system integrated with an home automation system. The audio system is further integrated with intercom and video intercom including the controlling of the intercom communications and the home automation appliances via a network and the Internet. The stereo signals are encoded and distributed from a selector/encoder, via distributors or directly to speakers via at least one single twisted pair carrying the encoded signals multiplexed with serial signals and with microphone signals for communicating with the intercom or with e-station via the network or the Internet. The multiplexed signals are propagated via the same twisted pair that also feeds a constant current to the system devices or providing power via separated twisted pair. The speakers include wired or wireless remote control receivers and transmitters for propagating remote control signals.

Abstract: Disclosed are a encoding device, a decoding device, and encoding and decoding methods, wherein when a multi-channel signal is encoded with high efficiency, using an adaptive filter, the number of arithmetic operations to update a filter coefficient of the adaptive filter can be reduced. An update range determination unit (170) determines the range of a filter coefficient order (update order range) of a filter coefficient to be updated, among filter coefficients gk(n) of the adaptive filter, on the basis of a mutual correlation function between an input (L) signal and an input (R) signal. The adaptive filter (130) updates the filter coefficient gk(n) of the filter coefficient order (n) to be updated, using a decoding (L) signal and a decoding error (R) signal.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

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

Abstract: Provided are a signal processing method capable of inserting additional information into a bitstream without departing from the AC-3 standard or the Enhanced AC-3 standard. The signal processing method includes: extracting additional information which is for performing at least one of a channel number expansion of an audio signal in an encoding apparatus and a three-dimensional (3D) reproduction of the audio signal in the encoding apparatus; encoding the audio signal according to a predetermined standard; formatting the encoded audio signal into a bitstream including bitstream information, at least one audio block and an auxiliary data field; and inserting the additional information into at least one of additional bitstream information included in the bitstream information, a skip field included in the audio block and auxiliary data bits included in the auxiliary data field.

Abstract: Provided are a method and apparatus for encoding and decoding a stereo signal or a multi-channel signal. According to the method and apparatus, a stereo signal or a multi-channel signal can be encoded and/or decoded by generating parameters based on a mono signal.

Abstract: The invention enables the inclusion of voice and remaining audio information at different parts of the audio production process. In particular, the invention embodies special techniques for VRA-capable digital mastering, accommodation of PCPV/PCA and/or SCRA signals in audio CODECs, VRA-capable encoders and decoders, and VRA in DVD and other digital audio file formats. The invention facilitates an end-listener's voice-to-remaining audio (VRA) adjustment upon the playback of digital audio media formats by focusing on new configurations of multiple parts of the entire digital audio system, thereby enabling a new technique intended to benefit audio end-users (end-listeners) who wish to control the ratio of the primary vocal/dialog content of an audio program relative to the remaining portion of the audio content in that program.

Abstract: Disclosed is an apparatus for encoding and decoding a multi-channel audio signal. The apparatus for encoding the multi-channel audio signal groups channels of a multi-channel audio signal, eliminates redundant information between channels using a mixing matrix including phase information, converts a frequency of the signal, and encodes the signal.

Abstract: The application relates to audio encoder and decoder systems. An embodiment of the encoder system comprises a downmix stage for generating a downmix signal and a residual signal based on a stereo signal. In addition, the encoder system comprises a parameter determining stage for determining parametric stereo parameters such as an inter-channel intensity difference and an inter-channel cross-correlation. Preferably, the parametric stereo parameters are time- and frequency-variant. Moreover, the encoder system comprises a transform stage. The transform stage generates a pseudo left/right stereo signal by performing a transform based on the downmix signal and the residual signal. The pseudo stereo signal is processed by a perceptual stereo encoder. For stereo encoding, left/right encoding or mid/side encoding is selectable. Preferably, the selection between left/right stereo encoding and mid/side stereo encoding is time- and frequency-variant.

Abstract: An apparatus for processing an audio signal and method thereof applied to an audio playback system are disclosed. The apparatus comprises a decoder, an error-correcting circuit and an audio correcting module. The method for processing audio signals in accordance with the present invention decodes the audio signal to generate a decoded signal by the decoder. Then, the error-correcting circuit performs an error-correcting algorithm in the decoded signal to generate an error indication signal and an output audio signal. And the audio correcting module corrects the output audio signal to generate a corrected audio signal when the error indication signal indicates that the output audio signal has error.

Abstract: A stereo encoding method, a stereo encoding device, and an encoder are provided. The stereo encoding method includes: obtaining a left channel energy relation coefficient and a right channel energy relation coefficient; obtaining a left energy sum and a right energy sum respectively; performing cross correlation between sub-bands of a first monophonic signal at a wave trough and sub-bands of the left channel signal according to the left channel energy relation coefficient, and performing cross correlation between sub-bands of the first monophonic signal at the wave trough and sub-bands of the right channel signal according to the right channel energy relation coefficient; obtaining a scaling factor by using the left energy sum, the right energy sum, and cross correlation results; and encoding the stereo left and right channel signals according to the scaling factor.

Abstract: Processing an audio signal associated with a sound recording made available to be rendered to an end user is disclosed. The audio signal is received. A high-order perceptual attribute of the audio signal as rendered is changed by modifying the audio signal. The modification may be based on real-time analysis of the audio signal.

Abstract: The invention proposes a method for embedding an ancillary data into a compressed audio signal. This is achieved by replacing Least Significant Bits (LSBs) in at least one frequency subband of the compressed audio signal by the ancillary data. When replacing LSB bits of compressed subband signals with the ancillary data, the subband signal is effectively modified, resulting in a different decoded output. The replaced LSB bits corresponding to the ancillary data are conveyed as part of the bitstream and can be easily extracted at the decoder. In such a way the decoder obtains the ancillary data that can be used for more advanced audio reproduction at the decoder. The compressed audio itself maintains a good audio quality despite the replacement of the LSB bits of the frequency subband, because the LSB bits do not contribute to the audible artefacts.

Abstract: Provided is an audio decoder which can reduce an amount of arithmetic operations while suppressing occurrence of aliasing noise. The audio decoder includes: a decoder (102) and an analysis filter bank (110) which generate, from a coded down-mixed signal, the first frequency band signal (x) corresponding to a down-mixed signal (M); a channel expansion unit (130) which converts the first frequency band signal (x) generated by the analysis filter bank (110) into output signals (y) corresponding to respective audio signals of N channels, using BC information; an synthesis filter bank (140) which performs band synthesis for the output signals (y) generate by the channel expansion unit (130) and thereby converts the output signals (y) into the respective audio signals of the N channels on a time axis; and an aliasing noise detection unit (120) which detects occurrence of aliasing noise in the first frequency band signal (x).

Abstract: A method for generating a parametrically encoded audio signal, the method comprising: inputting a multi-channel audio signal comprising a plurality of audio channels; generating at least one combined signal of the plurality of audio channels; and generating one or more corresponding sets of side information including channel configuration information for controlling audio source locations in a synthesis of a binaural audio signal.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: A method and apparatus to encode audio with spatial information in a manner that does not depend on the exhibition setup, and to decode and play out optimally for any given exhibition setup, maximizing the sweet-spot area, and including setups with loudspeakers at different heights, and headphones. The part of the audio that requires very precise localization is encoded into a set of mono tracks with associated directional parameters, whereas the remaining audio is encoded into a set of Ambisonics tracks of a chosen order and mixture. Upon specification of a given exhibition system, the exhibition-independent format is decoded adapting to the specified system, by using different decoding methods for each assigned group.

Abstract: A portable player or a multi-channel home player includes: a mixed signal decoding unit that extracts, from a first inputted coded stream, a second coded stream representing a downmix signal into which multi-channel audio signals are mixed and supplementary information for reverting the downmix signal back to the multi-channel audio signals before being downmixed, and that decodes the second coded stream representing the downmix signal; a signal separation processing unit that separates the downmix signal obtained by decoding based on the extracted supplementary information and that generates audio signals which are acoustically approximate to the multi-channel audio signals before being downmixed; and headphones or speakers that reproduce the decoded downmix signal or speakers that reproduce the multi-channel audio signals separated from the downmix signal.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: A system and method are disclosed for extending the bandwidth of a narrowband signal such as a speech signal. The method applies a parametric approach to bandwidth extension but does not require training. The parametric representation relates to a discrete acoustic tube model (DATM). The method comprises computing narrowband linear predictive coefficients (LPCs) from a received narrowband speech signal, computing narrowband partial correlation coefficients (parcors) using recursion, computing Mnb area coefficients from the partial correlation coefficient, and extracting Mwb area coefficients using interpolation. Wideband parcors are computed from the Mwb area coefficients and wideband LPCs are computed from the wideband parcors.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: Parameters being a measure for a characteristic of a channel or of a pair of channels, wherein the parameter is a measure for a characteristic of the channel or of the pair of channels with respect to another channel of a multi-channel signal can be quantized more efficiently using a quantization rule that is generated based on a relation of an energy measure of the channel or the pair of channels and an energy measure of the multi-channel signal. With generation of the quantization rule taking into account a psycho acoustic approach, the size of an encoded representation of the multi-channel signal can be decreased by coarser quantization without significantly disturbing the perceptual quality of the multi-channel signal when reconstructed from the encoded representation.

Abstract: Provided is an encoding device (1) including: a pitch contour analysis unit (101) which detects information, a dynamic time-warping unit (102) which generates, based on the information, pitch change ratios (Tw_ratio in FIG. 18) within a range (86) including a range (86a) of the pitch change ratios corresponding to absolute pitch differences of 42 cents or larger; a first lossless coding unit (103) which codes the generated pitch parameters (102x); a time-warping unit (104) which shifts a pitch of a signal according to the information; and a second encoding unit which codes a signal (104x) obtained by the shifting.

Abstract: A method, system, apparatus and computer program product are provided for improved stereo coding. In particular, the method, system, apparatus and computer program product provide a technique for performing Mid-Side (M/S) stereo coding, in which an additional step is added to the coding process, whereby a parameter that is used in determining when the mid and side signals will be used instead of the left and right input signals is modified prior to making the selection between the signal pairs. In particular, the masking threshold associated with either the left or the right input signal may be modified based on a relationship between the energies of the two input signals. In addition, once the selection between the signal pairs has been made, the masking thresholds of the selected signals may be further modified, again based on a relationship between the energies of the left and right input signals.