Abstract: A method for compressing a HOA signal being an input HOA representation with input time frames (C(k)) of HOA coefficient sequences comprises spatial HOA encoding of the input time frames and subsequent perceptual encoding and source encoding. Each input time frame is decomposed (802) into a frame of predominant sound signals (XPS(k?1)) and a frame of an ambient HOA component (CAMB (k?1)). The ambient HOA component (CAMB (k?1)) comprises, in a layered mode, first HOA coefficient sequences of the input HOA representation (cn(k?1)) in lower positions and second HOA coefficient sequences (CAMB,n(k?1)) in remaining higher positions. The second HOA coefficient sequences are part of an HOA representation of a residual between the input HOA representation and the HOA representation of the predominant sound signals.

Abstract: A method and apparatus for processing audio content is described. The method and apparatus include receiving (510) audio content, the audio content including an input audio signal, a first reference audio signal, and a second reference audio signal, determining (550) a processing function for the input audio signal, the processing function determined based on a cost function between the input audio signal, the first reference audio signal and a second reference audio signal, and processing (560) the input audio signal using the determined processing function in order to produce an output audio signal.

Abstract: In general, techniques are described for transitioning an ambient higher order ambisonic coefficient. A device comprising a memory and a processor may be configured to perform the techniques. The processor may obtain, from a frame of a bitstream of encoded audio data, a bit indicative of a reduced vector. The reduced vector may represent, at least in part, a spatial component of a sound field. The processor may also obtain, from the frame, a bit indicative of a transition of an ambient higher-order ambisonic coefficient. The ambient higher-order ambisonic coefficient may represent, at least in part, an ambient component of the sound field. The reduced vector may include a vector element associated with the ambient higher-order ambisonic coefficient in transition. The memory may be configured to store the frame of the bitstream.

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: There are two representations for Higher Order Ambisonics denoted HOA: spatial domain and coefficient domain. The invention generates from a coefficient domain representation a mixed spatial/coefficient domain representation, wherein the number of said HOA signals can be variable. A vector of coefficient domain signals is separated into a vector of coefficient domain signals having a constant number of HOA coefficients and a vector of coefficient domain signals having a variable number of HOA coefficients. The constant-number HOA coefficients vector is transformed to a corresponding spatial domain signal vector. In order to facilitate high-quality coding, without creating signal discontinuities the variable-number HOA coefficients vector of coefficient domain signals is adaptively normalized and multiplexed with the vector of spatial domain signals.

Abstract: A coding apparatus encodes a first band of an input audio signal, normalizes a first spectrum included in each sub-band of the first band using a spectrum power envelope, performs a clipping process on the normalized first spectrum, the clipping process comparing between a predetermined threshold and the absolute value of an amplitude of the spectrum and replaces the amplitude value of the spectrum with the threshold if the absolute value of the amplitude of the spectrum exceeds the threshold, calculates a correlation between a spectrum in each divided band of a second band and a spectrum in a plurality of candidate bands containing the clipped normalized first spectrum, the second spectrum being higher than a predetermined frequency, identifies the best bands of the plurality of candidate bands, and encodes the second spectrum using lag information identifying the best band for transmitting the lag information to a decoder.

Abstract: In general, techniques are directed to performing normalization with respect to ambient higher order ambisonic audio data. A device configured to decode higher order ambisonic audio data may perform the techniques. The device may include a memory and one or more processors. The memory may be configured to store an audio channel that provides a normalized ambient higher order ambisonic coefficient representative of at least a portion of an ambient component of a soundfield. The one or more processors may be configured to perform inverse normalization with respect to the audio channel.

Abstract: Systems and techniques for compression and decoding of audio data are generally disclosed. An example device for compressing higher order ambisonic (HOA) coefficients representative of a soundfield includes a memory configured to store audio data and one or more processors configured to: determine when to use ambient HOA coefficients of the HOA coefficients to augment one or more foreground audio objects obtained through decomposition of the HOA coefficients based on one or more singular values also obtained through the decomposition of the HOA coefficients, the ambient HOA coefficients representative of an ambient component of the soundfield.

Abstract: In general, techniques are described for compression and decoding of audio data are generally disclosed. An example device for compressing audio data includes one or more processors configured to apply a decorrelation transform to ambient ambisonic coefficients to obtain a decorrelated representation of the ambient ambisonic coefficients, the ambient HOA coefficients having been extracted from a plurality of higher order ambisonic coefficients and representative of a background component of a soundfield described by the plurality of higher order ambisonic coefficients, wherein at least one of the plurality of higher order ambisonic coefficients is associated with a spherical basis function having an order greater than one.

Abstract: Improved tools for authoring and rendering audio reproduction data are provided. Some such authoring tools allow audio reproduction data to be generalized for a wide variety of reproduction environments. Audio reproduction data may be authored by creating metadata for audio objects. The metadata may be created with reference to speaker zones. During the rendering process, the audio reproduction data may be reproduced according to the reproduction speaker layout of a particular reproduction environment.

Abstract: The invention relates to Higher-Order Ambisonics (HOA) Content Loudness Level Adjustment. A method for adjusting a loudness level of a HOA audio signal on a transmitting side comprises steps of measuring loudness of only the W-channel of the HOA audio signal, and leveling HOA master gain for all channels of the HOA signal according to the measured loudness of the W-channel.

Abstract: The present disclosure provides methods, devices and computer program products which provide less complex and more flexible control of the introduced decorrelation in an audio coding system. According to the disclosure, this is achieved by calculating and using two weighting factors, one for an approximated audio object and one for a decorrelated audio object, for introduction of decorrelation of audio objects in the audio coding system.

Abstract: An audio decoder for decoding an encoded audio signal, the encoded audio signal including a first channel element and a second channel element in a payload section of a data stream and first decoder configuration data for the first channel element and second decoder configuration data for the second channel element in a configuration section of the data stream, includes: a data stream reader for reading the configuration data for each channel element in the configuration section and for reading the payload data for each channel element in the payload section; a configurable decoder for decoding the plurality of channel elements; and a configuration controller for configuring the configurable decoder so that the configurable decoder is configured in accordance with the first decoder configuration data when decoding the first channel element and in accordance with the second decoder configuration data when decoding the second channel element.

Abstract: Encoding and decoding devices for encoding the channels of an audio system having at least four channels are disclosed. The decoding device has a first stereo decoding component which subjects a first pair of input channels to a first stereo decoding, and a second stereo decoding component which subjects a second pair of input channels to a second stereo decoding. The results of the first and second stereo decoding components are crosswise coupled to a third and a fourth stereo decoding component which each performs stereo decoding on one channel resulting from the first stereo decoding component, and one channel resulting from the second stereo decoding component.

Abstract: A coding apparatus normalizes a low-frequency spectrum included in each of sub-bands obtained from dividing a low band part, using a largest amplitude value among the low-frequency spectrum included in each sub-band, obtains a normalized low-frequency spectrum by decoding the first encoded data, and calculates a correlation between each divided band of a high-frequency spectrum and a plurality of candidate bands of the normalized low-frequency spectrum. The best bands of a plurality of candidate bands are identified, each candidate band having a starting frequency position with non-zero amplitude in the normalized low-frequency spectrum, the high-frequency spectrum being in a high band part of the input audio signal that is higher than the predetermined frequency, and the high-frequency spectrum is encoded using lag information identifying the best band for transmitting the lag information to a decoder.

Abstract: Audio signals are recorded with microphones receiving acoustic information from one or more directions. The corresponding audio signals can be pre-listened to in production studios. However, Higher Order Ambisonics (HOA) audio signals are matrixed in such a way that the matrixing prevents listening to the matrixed sound signals without dematrixing the matrixed sound signals. For enabling a sound engineer to listen to such a matrixed signal without full HOA decoding, an informative audio signal is added together with related side information data at encoding side to a selected part of the matrixed signal. This informative audio signal is removed before the inverse matrixing process at decoding side.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding of a vector of parameters in an audio coding system. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system. According to the disclosure, a modulo differential approach for coding and encoding a vector of a non-periodic quantity may improve the coding efficiency and provide encoders and decoders with less memory requirements. Moreover, an efficient method for encoding and decoding a sparse matrix is provided.

Abstract: An audio processing apparatus including an audio phase detecting device and an adjusting device is provided. After detecting a phase relationship between a first channel signal and a second channel signal, the audio phase detecting device generates a phase control signal. The adjusting device is coupled to the audio phase detecting device and used for selectively adjusting the first channel signal according to the phase control signal.

Abstract: A delivery station and a method for operating the delivery station including a station server, at least one of storage device for controlling storage facility and a delivery device for delivery including confirmation, located in one of a building and distant and connected to at least one e-provider server via a network for gathering and delivering without error at least one of merchandise and service item included in an order received by at least one e-provider from at least one dweller of the building, each of the devices comprising a CPU, a memory, indicators and touch display screen identifying particulars pertaining to a stored item, the order received from a dweller, a barcode reader, an RFID reader, a communication transceiver for verifying and communicating the gathering and the delivery, including updating of the stored item quantity via the station server upon execution or cancellation of the delivery.

Abstract: There are two representations for Higher Order Ambisonics denoted HOA: spatial domain and coefficient domain. The invention generates from a coefficient domain representation a mixed spatial/coefficient domain representation, wherein the number of said HOA signals can be variable. A vector of coefficient domain signals is separated into a vector of coefficient domain signals having a constant number of HOA coefficients and a vector of coefficient domain signals having a variable number of HOA coefficients. The constant-number HOA coefficients vector is transformed to a corresponding spatial domain signal vector. In order to facilitate high-quality coding, without creating signal discontinuities the variable-number HOA coefficients vector of coefficient domain signals is adaptively normalized and multiplexed with the vector of spatial domain signals.

Abstract: A signal processing apparatus includes: an audio image localization processing unit performing audio image localization processing on a sound signal of each frequency band for each channel of the sound signal based on information used to determine an audio image localization position of each frequency band; and a mixing unit mixing the sound signals of the respective channels subjected to the audio image localization processing by the audio image localization processing unit.

Abstract: A multi-stage analog-to-digital conversion method and system use window functions and translation to match high gain frames of data to target frames of data. The technique selects window data packets for the output stream based the stage of data having the highest gain satisfying selection criteria, such as requiring a frame of data for the respective stage to satisfy a predetermined accuracy of fit value compared to a target frame of data for a zero gain stage.

Abstract: An apparatus for downmixing three or more audio input channels to obtain two or more audio output channels is provided. The apparatus includes a receiving interface for receiving the three or more audio input channels and for receiving side information. Moreover, the apparatus includes a downmixer for downmixing the three or more audio input channels depending on the side information to obtain the two or more audio output channels. The number of the audio output channels is smaller than the number of the audio input channels. The side information indicates a characteristic of at least one of the three or more audio input channels, or a characteristic of one or more sound waves recorded within the one or more audio input channels, or a characteristic of one or more sound sources which emitted one or more sound waves recorded within the one or more audio input channels.

Abstract: The invention improves HOA sound field representation compression. The HOA representation is analyzed for the presence of dominant sound sources and their directions are estimated. Then the HOA representation is decomposed into a number of dominant directional signals and a residual component. This residual component is transformed into the discrete spatial domain in order to obtain general plane wave functions at uniform sampling directions, which are predicted from the dominant directional signals. Finally, the prediction error is transformed back to the HOA domain and represents the residual ambient HOA component for which an order reduction is performed, followed by perceptual encoding of the dominant directional signals and the residual component.

Abstract: The present invention relates to a method and device for encoding or decoding an object audio signal or rendering the object audio signal in a three-dimensional space. The method for processing an audio signal, according to one aspect of the present invention, comprises the steps of: generating a first object signal group and a second object signal group obtained by classifying a plurality of object signals according to a determined method; generating a first down-mix signal for the first object signal group; generating a second down-mix signal for the second object signal group; generating first object extraction information in correspondence with the first down-mix signal with respect to object signals included in the first object signal group; and generating second object extraction information in correspondence with the second down-mix signal with respect to object signals included in the second object signal group.

Abstract: Higher Order Ambisonics (HOA) represents three-dimensional sound. HOA provides high spatial resolution and facilitates analyzing of the sound field with respect to dominant sound sources. The invention aims to identify independent dominant sound sources constituting the sound field, and to track their temporal trajectories. Known applications are searching for all potential candidates for dominant sound source directions by looking at the directional power distribution of the original HOA representation, whereas in the invention all components which are correlated with the signals of previously found sound sources are removed. By such operation the problem of erroneously detecting many instead of only one correct sound source can be avoided in case its contributions to the sound field are highly directionally dispersed.

Abstract: According to the invention, a system for converting sound to electrical signals is disclosed. The system may include a gaze tracking device and a microphone. The gaze tracking device may determine a gaze direction of a user. The microphone may be more sensitive in a selected direction than at least one other direction and alter the selected direction based at least in part on the gaze direction determined by the gaze tracking device.

Abstract: It is inter alia disclosed to provide a left signal representation associated with a left audio channel and a right signal representation associated with a right audio channel, each of the left and right signal representations being associated with a plurality of subbands of a frequency range, and to provide directional information associated with at least one subband of the plurality of subbands associated with the left and the right signal representation, the directional information being at least partially indicative of a direction of a sound source with respect to the left and right audio channel.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding a stereo audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and a discrete representation of the stereo audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

Abstract: A transceiver system may be configured to provide tunable bandwidths. The transceiver may comprise a signal processing component and a filtering component, which may comprise a plurality of filters. The signal processing component may determine one or more adjustments that are applicable to one or both of a first filter that is configured for filtering signals corresponding to a first frequency band associated with a first stream, and a second filter that is configured for filtering signals corresponding to a second frequency band associated with a second stream. The one or more adjustments may correspond to modifications in one or both of the first frequency band and the second frequency band. The one or more adjustments may be communicated to the filtering component, which may apply the adjustments to one or more of the plurality of filters.

Abstract: The present invention relates to an encoding device and method, and a decoding device and method, and a program which enable music signals to be played with higher sound quality by expanding a frequency band. A band pass filter divides an input signal into multiple subband signals, a feature amount calculating circuit calculates feature amount using at least any one of the divided multiple subband signals and the input signal, a high-frequency subband power estimating circuit calculates an estimated value of high-frequency subband power based on the calculated feature amount, and a high-frequency signal generating circuit generates a high-frequency signal component based on the multiple subband signals divided by the band pass filter and the estimated value of the high-frequency subband power calculated by the high-frequency subband power estimating circuit.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed.

Abstract: In general, techniques are described for indicating frame parameter reusability for decoding vectors. A device comprising a processor and a memory may perform the techniques. The processor may be configured to obtain a bitstream comprising a vector representative of an orthogonal spatial axis in a spherical harmonics domain. The bitstream may further comprise an indicator for whether to reuse, from a previous frame, at least one syntax element indicative of information used when compressing the vector. The memory may be configured to store the bitstream.

Abstract: A method for down-mixing of a m-channel audio signal (L, R, C, Ls, Rs, Rss, Lss) into a n-channel audio signal (Ro, Lo, Rso, Lso), where m is an integer for which holds m>n and n is an integer for which holds n?2, including the step of generating one of the n-channel audio signals of one side (right or left) of a listener (Ro, Lo, Rso, Lso), by a combination of: a first term including a signal component (R, L, Rs, Ls) of the m-channel audio signal of the same side only, and a second term dependent of m, including one or more of further signal components of the m-channel audio signal (C, Ls, Rs, Rss, Lss) of the same side only, multiplied by at least one respective filtering function (H1, H2, H3, H4, H5, H6, H7, H8), said filtering function being dependent on: a frequency characteristic of the transmission path between the position of the loudspeaker of the respective signal component of the further m-channel audio signal, and a position of the right ear or left ear, respectively, of a listener in an m-chan

Abstract: In general, techniques are described for coding of spherical harmonic coefficients representative of a three dimensional soundfield. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may be configured to store a plurality of spherical harmonic coefficients. The one or more processors may be configured to perform an energy analysis with respect to the plurality of spherical harmonic coefficients to determine a reduced version of the plurality of spherical harmonic coefficients.

Abstract: By copying to a high-frequency band portion (extension band) a low-frequency band portion in which peaking has been set to a sufficiently low state, this encoding device is capable of preventing generation of a spectrum with overly high peaking in the high-frequency band portion, and of generating a high-quality extension band spectrum. This device comprises: a maximum value search unit which searches, in each of multiple sub-bands obtained by dividing the low-frequency band portion of an audio signal and/or music signal below a prescribed frequency, for the maximum value of the amplitude of a first spectrum obtained by decoding first encoded data, which is encoded data in the low-frequency band portion; and an amplitude normalization unit which obtains a normalized spectrum by normalizing, at the maximum values of the amplitude of each sub-band, the first spectrum contained in each sub-band.

Abstract: Provided is an apparatus and method for converting a 10.2 channel signal into a multi-channel signal having a relatively few number of channels, thereby encoding and decoding the 10.2 channel signal. An apparatus for encoding/decoding a multi-channel signal may include a multi-channel signal converter to convert a first multi-channel signal into a second multi-channel signal having a fewer number of channels when compared to the first multi-channel signal, a remaining signal generator to generate a remaining signal using a difference between the first multi-channel signal and the second multi-channel signal, and an encoder to encode the second multi-channel signal and the remaining signal.

Abstract: Provided is an apparatus for detecting a speech/non-speech section. The apparatus includes an acquisition unit which obtains inter-channel relation information of a stereo audio signal, a separation unit which separates each element of the stereo audio signal into a center channel element and a surround element on the basis of the inter-channel relation information, a calculation unit which calculates an energy ratio value between a center channel signal composed of center channel elements and a surround channel signal composed of surround elements, for each frame, and an energy ratio value between the stereo audio signal and a mono signal generated on the basis of the stereo audio signal, and a judgment unit which determines a speech section and a non-speech section from the stereo audio signal by comparing the energy ratio values.

Abstract: An audio enhancement technique comprises an audio source. The audio source is split into at least two branches and each of the branches is processed by a filter bank. Outputs from the filer banks are summed up to create a summed output signal. The summed output signal is splitting into two or more segments to create a split summed output signal. The split summed output signal is processed by one or more audio processing modules. Audio processed by each of the audio processing modules is summed to create a summed audio processed signal. The summed audio processed signal is provided to a balance module to select a balance mix of the audio source and the summed audio processed signal to create a balanced mixed audio signal. The balanced mixed audio signal is fed to a stereo output gain module for gain adjustment to create a gain adjusted stereo signal.

Abstract: A method and apparatus for the parametric encoding of a stereo digital-audio signal. The method includes encoding a mono signal produced by downmixing applied to the stereo signal and encoding spatialization information of the stereo signal. Downmixing includes determining, for a predetermined set of frequency sub-bands, a phase difference between two stereo channels; obtaining an intermediate channel by rotating a first predetermined channel of the stereo signal through an angle obtained by reducing the phase difference; determining the phase of the mono signal from the phase of the signal that is the sum of the intermediate channel and the second stereo signal, and from a phase difference between, on the one hand, the signal that is the sum of the intermediate channel and the second channel and, on the other hand, the second channel of the stereo signal. Also provided are a decoding method, an encoder and a decoder.

Abstract: A method and apparatus for preserving matrix-surround information in encoded audio/video includes a receiver operative to receive matrix-surround encoded audio signals via a modem, separate the audio signals into a frequency spectrum having discrete audio frequencies, and determine a cutoff threshold used to encode the matrix-surround encoded audio signals. The method and apparatus further includes a decoder operative to decode a first set of the audio frequencies below the determined cutoff threshold using a first matrix-surround preserving audio encoding method and to decode a second set of audio frequencies above the cutoff threshold using a second non matrix-surround preserving audio encoding method.

Abstract: A decoder and decoding method for use in surround sound system wherein at least four audio input signals representing an original sound field are encoded into two channel signals and said encoded signals are decoded into at least four audio output signals corresponding to the four audio input signals and have an amplitude ratio and a phase relationship. The decoder and method including means for: compensating the said encoded signals for variations in perceived loudness relative to frequency associated with the encoded two channel signals due to non linearity in human hearing response at least at some frequencies; producing steering signals in responsive to the phase relationship of the said compensated signals; decoding said encoded signals to produce audio output signals corresponding to audio input signals by varying at least the amplitude ratio of said encoded signals contained in each of the output signals in response to said steering signals.

Abstract: A multi-channel audio system including first combining circuitry, for combining a first spectral band of a first plurality channels to provide a first bass audio signal stream; second combining circuitry, for combining the first spectral band of a second plurality channels to provide a second bass audio signal stream; and third combining circuitry, for combining a second spectral band, the second spectral band including lower frequencies than the first spectral band, of the first plurality of channels and the second plurality of channels to provide a third bass audio signal stream.

Abstract: A decoding apparatus (10) is disclosed which includes: a storing means (11) for storing encoded audio signals including multi-channel audio signals; a transforming means (40) for transforming the encoded audio signals to generate transform block-based audio signals in a time domain; a window processing means (41) for multiplying the transform block-based audio signals by a product of a mixture ratio of the audio signals and a first window function, the product being a second window function; a synthesizing means (43) for overlapping the multiplied transform block-based audio signals to synthesize audio signals of respective channels; and a mixing means (14) for mixing audio signals of the respective channels between the channels to generate a downmixed audio signal. Furthermore, an encoding apparatus is also disclosed which downmixes the multi-channel audio signals, encodes the downmixed audio signals, and generates the encoded, downmixed audio signals.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving a downmix signal and side information; extracting control restriction information from the side information; receiving control information for controlling gain or panning at least one object signal; generating at least one of first multi-channel information and first downmix processing information based on the control information and object information, without using the control restriction information; and, generating an output signal by applying the at least one of the first multichannel information and the first downmix processing information to the downmix signal, wherein the control restriction information relates to a parameter indicating limiting degree of the control information.

Abstract: The present document relates to audio encoding/decoding. In particular, the present document relates to a method and system for improving the quality of encoded multi-channel audio signals. An audio encoder configured to encode a multi-channel audio signal according to a total available data-rate is described. The multi-channel audio signal is representable as a basic group (121) of channels for rendering the multi-channel audio signal in accordance to a basic channel configuration, and as an extension group (122) of channels, which—in combination with the basic group (122)—is for rendering the multi-channel audio signal in accordance to an extended channel configuration. The basic channel configuration and the extended channel configuration are different from one another.

Abstract: A method for encoding audio signals, for later reproduction in arbitrary three-dimensional loudspeaker layouts, based on the generation of an intermediate channel-independent representation, which enables the creation, manipulation and reproduction of sounds with complex apparent size and shape, including multiple disconnected shapes.

Abstract: Provided is an apparatus and method of encoding and decoding multiple channel signals based upon phase information and one or more residual signals.

Abstract: An audio signal decoder for providing a decoded multi-channel audio signal representation on the basis of an encoded multi-channel audio signal representation has a time warp decoder configured to selectively use individual audio channel specific time warp contours or a joint multi-channel time warp contour for a reconstruction of a plurality of audio channels represented by the encoded multi-channel audio signal representation.

Abstract: An apparatus for enhancing a multichannel audio signal comprising at least two channels configured to: estimate a value representing a direction of arrival associated with a first audio signal from at least a first channel and a second audio signal from at least a second channel of at least two channels of a multichannel audio signal; determine a scaling factor dependent on the direction of arrival associated with the first audio signal and the second audio signal; and apply the scaling factor to a parameter associated with a difference in audio signal levels between the first audio signal and the second audio sign.