Abstract: Disclosed is an apparatus and method for processing a multichannel audio signal. A multichannel audio signal processing method may include: generating an N-channel audio signal of N channels by down-mixing an M-channel audio signal of M channels; and generating a stereo audio signal by performing binaural rendering of the N-channel audio signal.

Abstract: The invention relates to a method for centralized control of multiple active noise cancellation devices. The method includes identifying a trigger event. Also, the method includes identifying, in response to identifying the trigger event, two or more zones of a mapped area. Further, the method includes identifying, based on the two or more zones, two or more devices. Still yet, the method includes transmitting a command to disable active noise cancellation on each of the two or more devices.

Abstract: A method includes sampling a common signal over a plurality of iterations, concatenating the sampled iterations, summing samples of the concatenated iterations within a window of a predetermined sample width, at each of a plurality of window positions along the concatenated iterations, and estimating noise of the common signal from the summed sample values at the window positions.

Abstract: The present disclosure enables applications of a computing system to coordinate object-based audio resources by the use of a minimum resource working set. The minimum resource working set encourages an application to be fair in its requirements since specifying a large number will most likely result in the application receiving zero resources, or losing all of its resources to another application. A working set, which can include a minimum and a maximum working set, also provides a useful metric for the spatial audio resource manager to use when balancing demand. In addition, a minimum working set provides a performance metric for resource balancing since it exposes what the minimum functional requirement is from the maxim requested resource claim.

Abstract: A device includes a receiver configured to receive an audio frame of an audio stream. The device also includes a decoder configured to generate first decoded speech associated with the audio frame and to determine a count of audio frames classified as being associated with band limited content. The decoder is further configured to output second decoded speech based on the first decoded speech. The second decoded speech may be generated according to an output mode of the decoder. The output mode may be selected based at least in part on the count of audio frames.

Abstract: This disclosure falls into the field of audio coding, in particular it is related to the field of spatial audio coding, where the audio information is represented by multiple signals, where the signals may comprise audio channels or/and audio objects. In particular the disclosure provides a method and apparatus for reconstructing audio objects in an audio decoding system. Furthermore, this disclosure provides a method and apparatus for encoding such audio objects.

Abstract: An image decoding method which can improve both image quality and coding efficiency is an image decoding method for decoding a coded stream which includes a plurality of processing units and a header for the processing units, the coded stream being generated by coding a moving picture, the processing units including at least one processing unit layered to be split into a plurality of smaller processing units, the image decoding method including specifying a hierarchical layer having a processing unit in which a parameter necessary for decoding is stored, by parsing hierarchy depth information stored in the header, and decoding the processing unit using the parameter stored in the processing unit located at the specified hierarchical layer.

Abstract: A loudspeaker includes a horn including a first end panel, a second end panel, a first side panel, and a second side panel. Edges of at least the first and second side panels define a diffraction slot opening. The first and second side panels are each fabricated from a sheet of flexible material held in a stressed, curved shape by at least a rigid support member. The panels are designed by an automated process based on a number of electro-acoustic transducers to be used in a loudspeaker, horizontal and vertical coverage angles for the loudspeaker, and a wall length for a horn of the loudspeaker.

Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which at least comprises one or more audio objects (106a). The encoder (108, 110) generates a bit stream (116) which comprises downmix signals (112) and side information which includes individual matrix elements (114) of a reconstruction matrix which enables reconstruction of the one or more audio objects (106a) in the decoder (120).

Abstract: Some methods may involve receiving a block of audio data, the block including N pulse code modulated (PCM) audio channels, including audio samples for each of the N channels, receiving metadata associated with the block of audio data and receiving a first set of values corresponding to reference audio samples. A second set of values, corresponding to audio samples from the block of audio data, may be determined. The first and second set of values may be compared. Based on the comparison, it may be determined whether the block of audio data is synchronized with the metadata.

Abstract: The present invention discloses a speech/audio signal processing method and apparatus.

Abstract: A system for generating one or more audio output signals is provided, having a decomposition module, a signal processor, and an output interface. The signal processor is configured to receive the direct component signal, the diffuse component signal and direction information. Moreover, the signal processor is configured to generate one or more processed diffuse signals depending on the diffuse component signal. For each audio output signal of the one or more audio output signals, the signal processor is configured to determine a direct gain, the signal processor is configured to apply the direct gain on the direct component signal to obtain a processed direct signal, and the signal processor is configured to combine the processed direct signal and one of the one or more processed diffuse signals to generate the audio output signal. The signal processor further has a gain function computation module and a signal modifier.

Abstract: Aspects of the disclosed technology provide a method comprising executing, at a first processor and over a predetermined time period, first and second instructions in repeated alternations, wherein each alternation comprises executing the first instruction a predetermined number of times followed by executing the second instruction the predetermined number of times. Further, the method comprises measuring, via a measuring apparatus, a side-channel signal that results from the first processor executing the first and second instructions in repeated alternations. Additionally, the method comprises filtering, by a second processor, a spectral component of the measured side-channel signal, and analyzing, by the second processor, the filtered spectral component of the measured side-channel signal to determine power spectral density within a frequency range of the filtered spectral component.

Abstract: An input signal includes a channel-based audio signal and an object-based audio signal, and an audio encoding device includes an audio scene analysis unit configured to determine an audio scene from the input signal and detect audio scene information; a channel-based encoder that encodes the channel-based audio signal output from the audio scene analysis unit; an object-based encoder that encodes the object-based audio signal output from the audio scene analysis unit; and an audio scene encoding unit configured to encode the audio scene information.

Abstract: A device of updating a media file for an increased resolution includes an I/O interface, a processor, and a storage device. The I/O interfaces acquires a first media file having first data samples and a second media file having second data samples. Each first data sample is sampled from an analog media signal and encoded with P bits and each second data sample is sampled from the analog media signal and encoded with Q bits, wherein “P” and “Q” are integers, and a resolution in bits of Q is greater than a resolution in bits of. The processor divides each second data sample by a corresponding one of the first data samples to calculate third data samples. The storage device stores the calculated third data samples as a third media file.

Abstract: An audio apparatus and an audio providing method thereof are provided. The audio providing method includes receiving an audio signal including a plurality of channels, applying an audio signal having a channel, from among the plurality of channels, giving a sense of elevation to a filter to generate a plurality of virtual audio signals to be respectively output to a plurality of speakers, applying a combination gain value and a delay value to the plurality of virtual audio signals so that the plurality of virtual audio signals respectively output through the plurality of speakers form a sound field having a plane wave, and respectively outputting the plurality of virtual audio signals, to which the combination gain value and the delay value are applied, through the plurality of speakers. The filter processes the audio signal to have a sense of elevation.

Abstract: The present document discloses a method and apparatus for compensating for a lost frame in a transform domain, comprising: calculating frequency-domain coefficients of a current lost frame using frequency-domain coefficients of one or more frames prior to the current lost frame, and performing frequency-time transform to obtain an initially compensated signal; and performing waveform adjustment, to obtain a compensated signal. Alternatively, extrapolation is performed for all or part of frequency points of the current lost frame using phases and amplitudes of corresponding frequency points of a plurality of previous frames to obtain phases and amplitudes of the corresponding frequency points of the current lost frame, to obtain frequency-domain coefficients of the corresponding frequency points, and frequency-time transform is performed to obtain a compensated signal.

Abstract: An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided, having: a receiving interface for receiving one or more frames, a coefficient generator, and a signal reconstructor. The coefficient generator is configured to determine one or more first audio signal coefficients, and one or more noise coefficients. Moreover, the coefficient generator is configured to generate one or more second audio signal coefficients, depending on the one or more first audio signal coefficients and depending on the one or more noise coefficients.

Abstract: An example playback device may identify a first version of audio content for play back from a first source, and further identify a first value of an audio characteristic of the first version of the audio content, where the playback device includes a capability corresponding to the audio characteristic. The playback device may determine that the first value of the audio characteristic of the first version of the audio content does not match the capability of the playback device. The playback device may then identify a second version of the audio content from a second source, where the second version of the audio content includes the audio characteristic having a second value that is closer to the capability of the playback device. The playback device may then cause playback of the second version of the audio content.

Abstract: A control device is provided which mixes and outputs two types of audio signals processed under standards different from each other; in particular, an audio signal of ASIO standard and an audio signal of WDM standard. An audio interface is connected to a computer, and an audio signal is input to the computer. A mixer module of the computer mixes an audio signal which is effect-processed by an ASIO application and an audio signal reproduced by a WDM application, outputs the mixed audio signal to the audio interface, and also outputs the mixed audio signal to the WDM application and records the sound. In each step of the process, double buffering is used to reduce sound drop-out.

Abstract: An apparatus for generating an audio signal envelope from one or more coding values is provided. The apparatus includes an input interface for receiving the one or more coding values, and an envelope generator for generating the audio signal envelope depending on the one or more coding values. The envelope generator is configured to generate an aggregation function depending on the one or more coding values, wherein the aggregation function includes a plurality of aggregation points. Furthermore, the envelope generator is configured to generate the audio signal envelope such that the envelope value of each of the envelope points of the audio signal envelope depends on the aggregation value of at least one aggregation point of the aggregation function.

Abstract: An audio decoder for providing at least four audio channel signals on the basis of an encoded representation is configured to provide a first residual signal and a second residual signal on the basis of a jointly encoded representation of the first residual signal and of the second residual signal using a multi-channel decoding. The audio decoder is configured to provide a first audio channel signal and a second audio channel signal on the basis of a first downmix signal and the first residual signal using a residual-signal-assisted multi-channel decoding. The audio decoder is configured to provide a third audio channel signal and a fourth audio channel signal on the basis of a second downmix signal and the second residual signal using a residual-signal-assisted multi-channel decoding. An audio encoder is based on corresponding considerations.

Abstract: A method for processing an audio signal includes: decomposing an audio signal comprising spatial information into a set of audio signal components; and processing a first subset of the set of audio signal components according to a first processing scheme and processing a second subset of the set of audio signal components according to a second processing scheme different from the first processing scheme, wherein the first subset comprises audio signal components corresponding to at least one frontal signal source and the second subset comprises audio signal components corresponding to at least one ambient signal source; and wherein the second processing scheme is based on crosstalk cancellation.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: An audio decoder for providing at least four audio channel signals on the basis of an encoded representation is configured to provide a first residual signal and a second residual signal on the basis of a jointly encoded representation of the first residual signal and of the second residual signal using a multi-channel decoding. The audio decoder is configured to provide a first audio channel signal and a second audio channel signal on the basis of a first downmix signal and the first residual signal using a residual-signal-assisted multi-channel decoding. The audio decoder is configured to provide a third audio channel signal and a fourth audio channel signal on the basis of a second downmix signal and the second residual signal using a residual-signal-assisted multi-channel decoding. An audio encoder is based on corresponding considerations.

Abstract: The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency Q?+r?0 is generated on the basis of existing components at ? and ?+?0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

Abstract: The application relates to HFR (High Frequency Reconstruction/Regeneration) of audio signals. In particular, the application relates to a method and system for performing HFR of audio signals having large variations in energy level across the low frequency range which is used to reconstruct the high frequencies of the audio signal. A system configured to generate a plurality of high frequency subband signals covering a high frequency interval from a plurality of low frequency subband signals is described.

Abstract: There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary encoding method includes inter alia calculating M downmix signals by forming combinations of N audio objects, wherein M?N, and calculating parameters which allow reconstruction of a set of audio objects formed on basis of the N audio objects from the M downmix signals. The calculation of the M downmix signals is made according to a criterion which is independent of any loudspeaker configuration.

Abstract: In order to process a subband signal of a plurality of real subband signals which are a representation of a real discrete-time signal generated by an analysis filter bank, a weighter for weighting a subband signal by a weighting factor determined for the subband signal is provided to obtain a weighted subband signal. In addition, a correction term is calculated by a correction term determiner, the correction term determiner being implemented to calculate the correction term using at least one other subband signal and using another weighting factor provided for the other subband signal, the two weighting factors differing. The correction term is then combined with the weighted subband signal to obtain a corrected subband signal, resulting in reduced aliasing, even if subband signals are weighted to a different extent.

Abstract: The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively.

Abstract: An apparatus for generating a stereo side signal having a first side channel and a second side channel from a stereo input signal having a first input channel and a second input channel is provided. The apparatus has a modification information generator for generating modification information based on mid-side information. Furthermore, the apparatus has a signal manipulator being adapted to manipulate the first input channel based on the modification information to obtain the first side channel and being adapted to manipulate the second input channel based on the modification information to obtain the second side channel. The modification information generator has a spectral weights generator for generating the modification information by generating a first spectral weighting factor based on a mono mid signal and on a mono side signal of the stereo input signal.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: A wireless-communication device includes a user interface configured to receive audio signals; a cryptographic module configured to encrypt the audio signals into a binary data sequence; a generator module configured to: receive a bit pattern representative of the binary data sequence; and generate a byte sequence representative of the bit pattern by generating a byte that is greater than the preceding byte when the current bit is of a first value and that is lesser than the preceding byte when the current bit is of a second value; an audio codec configured to generate transmission bits by generating a transmission bit having the first value when the current byte is greater than the preceding byte and having the second value when the current byte is lesser than the preceding byte; and a transceiver configured to convey the transmission bits to a receiver.

Abstract: The present invention pertains to an encoding device and method, a decoding device and method, and to a program, with which sound of an appropriate volume level can be obtained with a smaller quantity of codes. A first gain calculation circuit calculates a first gain for volume level correction of an input time series signal, and a second gain calculation circuit calculates a second gain for volume level correction of a downmixed signal obtained by downmixing of the input time series signal. A gain encoding circuit computes the gain differential between the first gain and the second gain, the gain differential between time frames, and the gain differential within time frames, and encodes the first gain and the second gain. The present invention can be applied in encoding devices and decoding devices.

Abstract: A method comprising acquiring an analog first sound signal, performing analog-to-digital conversion on the first sound signal to generate a digital second sound signal, performing reverberation processing, at a system bottom layer, on the second sound signal to generate a digital third sound signal, performing digital sound mixing processing on the third sound signal and a background sound signal sent from an application layer to generate a digital fourth sound signal, performing digital-to-analog conversion on the fourth sound signal to generate an analog fifth sound signal, performing analog sound mixing processing on the first sound signal and the fifth sound signal to generate an analog sixth sound signal, and playing the sixth sound signal.

Abstract: The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor S.

Abstract: An example device for receiving audio data includes an audio decoder implemented using digital logic circuitry and configured to decode audio data conforming to MPEG-H or AC-4 part 2, and an audio data processing unit implemented in digital logic circuitry and configured to receive a manifest file describing a plurality of audio objects conforming to MPEG-H or AC-4 part 2, wherein the manifest file indicates whether each of the audio objects is available via broadcast, broadband, or both broadcast and broadband, receive selection data indicating which of the audio objects are to be presented, receive streamed media data including the audio objects based on the selection data, and provide the audio objects indicated by the selection data to the audio decoder.

Abstract: A system, method, and computer program product are provided for maintaining service continuity associated with a mobile device during travel, by providing alternative navigational routes based on user defined inputs and preferences. In use, at least one input parameter is received from a mobile device associated with at least one communication network, the at least one input parameter corresponding to a navigational task to be performed utilizing the mobile device. Further, a starting point associated with the navigational task is identified. Additionally, an ending point associated with the navigational task is identified. Furthermore, at least one route is determined from the starting point to the ending point, based at least in part on a current service continuity associated with the mobile device, a historical service continuity associated with the mobile device, and the at least one input parameter.

Abstract: An audio encoder encodes a digital audio recording having a number of audio channels or audio objects. A Dynamic Range Control (DRC) processor produces a sequence of encoder DRC gain values, by applying a selected one of a number of DRC characteristics to a group of one or more of the audio channels or audio objects. The encoder DRC gain values are to be applied to adjust the group of audio channels or audio objects, upon decoding them from the encoded digital audio recording. A bitstream multiplexer combines a) the encoded digital audio recording with b) the sequence of encoder DRC gain values, an indication of the selected DRC characteristic, and an indication of an alternate DRC characteristic, the latter as metadata associated with the encoded digital audio recording. Other embodiments are also described including a system for decoding the encoded audio recording and performing DRC adjustment upon it.

Abstract: An audio encoding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute: calculating a similarity in phase of a first channel signal and a second channel signal contained in a plurality of channels of an audio signal; and selecting, based on the similarity, a first output that outputs one of the first channel signal and the second channel signal, or a second output that outputs both of the first channel signal and the second channel signal.

Abstract: An apparatus and method for encoding and decoding a signal for high frequency bandwidth extension are provided. An encoding apparatus may down-sample a time domain input signal, may core-encode the down-sampled time domain input signal, may transform the core-encoded time domain input signal to a frequency domain input signal, and may perform bandwidth extension encoding using a basic signal of the frequency domain input signal.

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 ({tilde over (C)}AMB (k?1)). The ambient HOA component ({tilde over (C)}AMB (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: An apparatus comprising a processor component to: provide, to a control device, an indication of availability to perform a processing task with one or more data set portions as a node device; perform a processing task specified by the control device with the one or more data set portions; and request a pointer to a location at which to store the one or more data set portions as a data block within a data file. In response to the data set including partitioned data, for each data set portion, include a data sub-block size of the data set portion and a hashed identifier derived from a partition label of a partition in the request; receive, from the control device, the requested pointer to the location; and store each data set portion as a data sub-block within the data block starting at the location within the data file.

Abstract: The present technique relates to an encoding device and a method, a decoding device and a method, and a program capable of obtaining higher quality audio. An encoding unit encodes position information and a gain of an object in a current frame in multiple encoding modes. A compressing unit generates, for each combination of encoding modes of each pieces of position information and gains, encoded meta data including encoding mode information indicating the encoding modes and encoded data which are the encoded position information and gains, and compresses the encoding mode information included in the encoding meta data. A determining unit selects encoded meta data of which amount of data is the least from among the encoded meta data generated for each combination, thus determining the encoding mode of each pieces of position information and gains. The present technique can be applied to an encoder and a decoder.

Abstract: Content identification methods for consumer devices determine robust audio fingerprints that are resilient to audio distortions. One method generates signatures representing audio content based on a constant Q-factor transform (CQT). A 2D spectral representation of a 1D audio signal facilitates generation of region based signatures within frequency octaves and across the entire 2D signal representation. Also, points of interest are detected within the 2D audio signal representation and interest regions are determined around selected points of interest. Another method generates audio descriptors using an accumulating filter function on bands of the audio spectrum and generates audio transform coefficients. A response of each spectral band is computed and transform coefficients are determined by filtering, by accumulating derivatives with different lags, and computing second order derivatives.

Abstract: The invention concerns a conference bridge server (30) for establishing a conference telephone call, and a corresponding method and a computer program product. The conference bridge server (30) comprises input and output means (31, 33) for a plurality of communication connections connecting conference terminals (11 to 16, 20) with the conference bridge server (30) and a conference core (32) for distributing input signals from conference terminals (11 to 16, 20) to at least some of the remaining conference terminals (11 to 16, 20). The conference bridge server (30) comprises means for issuing a special address to be utilized, by one or more participants (111 to 116) at a first location (1) using a common conference phone (10), to connect one or more laptop computers (11 to 16) as additional conference terminals (11 to 16).

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for receiving audio data encoding an utterance and environmental data, obtaining a transcription of the utterance, identifying an entity using the environmental data, submitting a query to a natural language query processing engine, wherein the query includes at least a portion of the transcription and data that identifies the entity, and obtaining one or more results of the query.

Abstract: Lossy compression and transmission of a downmixed composite signal having multiple tracks and objects, including a downmixed signal, is accomplished in a manner that reduces the bit-rate requirement as compared to redundant transmission or lossless compression, while reducing upmix artifacts. A compressed residual signal is generated and transmitted along with a compressed total mix and at least one compressed audio objects. In the reception and upmix aspect the invention decompresses a downmixed signal and other compressed objects, calculates an approximate upmix signal, and corrects specific base signals derived from the upmix, by subtracting a decompressed residual signal. The invention thus allows lossy compression to be used in combination with downmixed audio signals for transmission through a communication channel (or for storage).

Abstract: An input signal includes a channel-based audio signal and an object-based audio signal, and an audio encoding device includes an audio scene analysis unit configured to determine an audio scene from the input signal and detect audio scene information; a channel-based encoder that encodes the channel-based audio signal output from the audio scene analysis unit; an object-based encoder that encodes the object-based audio signal output from the audio scene analysis unit; and an audio scene encoding unit configured to encode the audio scene information.

Abstract: A method for pyramid vector quantization indexing of audio/video signals comprises obtaining of an integer input vector representing the audio/video signal samples. A leading sign is extracted from the integer input vector. The leading sign is a sign of a terminal non-zero coefficient in the integer input vector. The terminal non-zero coefficient is one of a first non-zero coefficient and a last non-zero coefficient in the integer input vector. The integer input vector is indexed with a pyramid vector quantization enumeration scheme into an output index representing the audio/video signal samples. The pyramid vector quantization enumeration scheme is designed for neglecting the sign of the terminal non-zero coefficient. The output index and the leading sign are outputted. A corresponding method for de-indexing, an encoder, a decoder, and computer programs therefore are also disclosed.