Abstract: A method includes: processing the digital audio input signal to generate M delayed digital audio signal samples; converting the delayed digital audio signal samples to frequency domain representation in N number of frequency bands to compute respective signal spectrum values; determining respective signal level estimates; computing respective frequency domain gain coefficients based on the respective signal level estimates and band gain laws; transforming the frequency domain gain coefficients to time domain representation to produce M time-varying filter coefficients of a processing filter; convolving the M delayed digital audio signal samples with the time-varying filter coefficients to produce the processed digital output signal; and updating the delayed digital audio signal samples in accordance with a sample-by-sample or a predetermined block rate; wherein two of the signal spectrum values for at least two of the frequency bands are updated at different rates; and wherein M and N are positive integer num

Abstract: A method for processing an encoded audio signal in a binary stream by MICDA predictive coding. The method includes the following steps: determining a signal assessed from quantification indices of the binary stream; determining unencoded parameters representative of the audio signal from the assessed signal; and processing the encoded audio signal using the determined parameters. Also provided is a device implementing the method.

Abstract: An audio watermarking system conveys information using an audio channel by modulating an audio signal to produce a modulated signal by embedding additional information into the audio signal. Modulating the audio signal includes segmenting the audio signal into overlapping time segments using a non-rectangular analysis window function produce a windowed audio signal, processing the windowed audio signal for a time segment to produce frequency coefficients representing the windowed time segment and having phase values and magnitude values, selecting one or more of the frequency coefficients, modifying phase values of the selected frequency coefficients using the additional information to map the phase values onto a known phase constellation, and processing the frequency coefficients including the modified phase values to produce the modulated signal.

Abstract: In general, techniques are described for closed loop quantization of HOA coefficients that provide a three-dimensional representation of the sound field. An audio encoding device may perform closed loop quantization of an audio object based at least in part on a result of performing quantization of directional information associated with the audio object. An audio decoding device may obtain an audio object that has been closed loop quantized based at least in part on a result of performing quantization of directional information associated with the audio object, and may dequantize the audio object.

Abstract: An encoding method and encoder is provided for transparent lossless audio watermarking by quantizing an original PCM audio signal twice, each quantization quantizing to a quantization grid. As a PCM signal is inherently already quantized, there are three quantization grids to consider, the first being the quantization grid of the original PCM signal, the second being that of the watermarked signal and the third being that of an intermediate signal. The technique reduces the amount of introduced quantization error, spectrally shapes the error and fully decorrelates signal alterations from the original audio, thus making the error more similar to additive noise. A decoding method and decoder is also provided, as is a method of altering the watermark without fully decoding the encoded signal.

Abstract: A method is provided for detecting a predetermined frequency band in an audio data signal which has previously been coded according to a succession of data blocks, among which at least certain blocks contain respectively at least one set of spectral parameters representing a linear prediction filter. Such a method of detection implements, for a current block among the at least certain blocks and for which at least a plurality of spectral parameters of the set have been previously decoded, acts of: determining, among the plurality of previously decoded spectral parameters, the index of the first spectral parameter closest to a threshold frequency; calculating at least one criterion on the basis of the determined index; and deciding whether the predetermined frequency band is detected in the current block, as a function of the criterion calculated.

Abstract: In an encoding method that is expected to produce a smaller code amount out of a periodicity-based encoding method and a non-periodicity-based encoding method, the amount of code or an estimated value of the amount of code of an integer value sequence which is derived from an audio signal is obtained while adjusting gain. In the other encoding method, an integer value sequence obtained in this process is substituted to obtain the amount of code or an estimated value of the amount of code of the integer value sequence. The obtained code amounts or estimated values are compared to choose one of the encoding methods and the integer value sequence is encoded using the chosen encoding method to obtain and output an integer signal code.

Abstract: An audio processing system (100) accepts an audio bitstream having one of a plurality of predefined audio frame rates. The system comprises a front-end component (110), which receives a variable number of quantized spectral components, corresponding to one audio frame in any of the predefined audio frame rates, and performs an inverse quantization according to predetermined, frequency-dependent quantization levels. The front-end component may be agnostic of the audio frame rate. The audio processing system further comprises a frequency-domain processing stage (120) and a sample rate converter (130), which provide a reconstructed audio signal sampled at a target sampling frequency independent of the audio frame rate. By its frame-rate adaptability, the system can be configured to operate frame-synchronously in parallel with a video processing system that accepts plural video frame rates.

Abstract: Provided is a tag insertion method performed by an apparatus for inserting a tag into a stereo audio signal, the method including receiving an original stereo audio signal, analyzing an energy distribution of the original stereo audio signal based on an azimuth, determining valid azimuths for control information and for a plurality of pieces of tag information based on the energy distribution, wherein the control information is used to control tag information, modulating the plurality of pieces of tag information and the control information generated based on the valid azimuths, generating a left signal and a right signal based on the modulated control information and the plurality of pieces of modulated tag information, and generating a multi-tagged stereo audio signal by mixing the generated left signal and the generated right signal with the original stereo audio signal.

Abstract: A coding efficiency of coding spectral coefficients of a spectrum of an audio signal is increased by en/decoding a currently to be en/decoded spectral coefficient by entropy en/decoding and, in doing so, performing the entropy en/decoding depending, in a context-adaptive manner, on a previously en/decoded spectral coefficient, while adjusting a relative spectral distance between the previously en/decoded spectral coefficient and the currently en/decoded spectral coefficient depending on an information concerning a shape of the spectrum.

Abstract: In general, techniques are described for obtaining audio rendering information in a bitstream. A device configured to render higher order ambisonic coefficients comprising a processor and a memory may perform the techniques. The processor may be configured to obtain sign symmetry information indicative of sign symmetry of a matrix used to render the higher order ambisonic coefficients to generate a plurality of speaker feeds. The memory may be configured to store the sparseness information.

Abstract: An acoustic processor for a mobile device is described the mobile device comprising a speaker including a speaker membrane, the speaker membrane having a first speaker membrane side and second speaker membrane side opposite the first speaker membrane side, a mobile device housing for providing a first acoustic path between the first speaker membrane side and the exterior of the mobile device and a second acoustic path between the second speaker membrane side and the exterior of the mobile device, the acoustic processor being configured and arranged to sense a signal on an acoustic processor input, the signal being induced on at least one terminal of the speaker in response to acoustic waves process the induced signal to discriminate between acoustic waves from different directions; and output the processed signal on an acoustic processor output.

Abstract: The present document relates to audio encoding and decoding. In particular, the present document relates to audio coding schemes which make use of high frequency reconstruction (HFR) methods. A system configured to determine a master scale factor band table of a highband signal (105) of an audio signal is described. The highband signal (105) is to be generated from a lowband signal (101) of the audio signal using a high frequency reconstruction (HFR) scheme. The master scale factor band table is indicative of a frequency resolution of a spectral envelope of the highband signal (105).

Abstract: Recordings from microphones that provide 1st order Ambisonics signals, so-called B-format signals, offer a limited cognition of sound directivity. Sound sources are perceived broader than they actually are, especially for off-center listening positions, and the sound sources are often located to be coming from the closest speaker positions. In a method and apparatus for enhancing the directivity of 1st order Ambisonics signals, additional directivity information is extracted (SFA) from the lower order Ambisonics input signal. The additional directivity information is used to estimate higher order Ambisonics coefficients, which are then combined with the coefficients of the input signal. Thus, the directivity of the Ambisonics signal is enhanced, which leads to an increased accuracy of spatial source localization when the Ambisonics signal is decoded to loud speaker signals. The resulting output signal has more energy than the input signal.

Abstract: A method and device for decoding a signal. The method for decoding a signal includes: obtaining spectral coefficients of sub-bands from a received bitstream by means of decoding; classifying sub-bands in which the spectral coefficients are located into a sub-band with saturated bit allocation and a sub-band with unsaturated bit allocation; performing noise filling on a spectral coefficient that has not been obtained by means of decoding and is in the sub-band with unsaturated bit allocation, so as to restore the spectral coefficient that has not been obtained by means of decoding; and obtaining a frequency domain signal according to the spectral coefficients obtained by means of decoding and the restored spectral coefficient. Therefore, a sub-band with unsaturated bit allocation in a frequency domain signal may be obtained by classification, thereby improving signal decoding quality.

Abstract: A device receives first and second data streams representing an audio and/or video content. The device is configured beforehand so as to disseminate the content from the first stream. Following a command for switching from the first stream to the second stream, the device determines by correlation a reference point in the first stream and in the second stream, and determines an offset in time from time-stamping information associated with the reference point in each of the first and second streams. Before switching, the device: pauses the second stream until the offset in time is absorbed when the offset in time indicates that the first stream is late; decreases a frame rate of the first stream until the offset in time is absorbed when the offset in time indicates that the first stream is in advance.

Abstract: An audio signal encoding and decoding method, an audio signal encoding and decoding apparatus, a transmitter, a receiver, and a communications system, which can improve encoding and/or decoding performance. The audio signal encoding method includes dividing a to-be-encoded time domain signal into a low band signal and a high band signal; encoding the low band signal to obtain a low frequency encoding parameter; calculating a voiced degree factor, and predicting a high band excitation signal; weighting the high band excitation signal and random noise using the voiced degree factor, so as to obtain a synthesized excitation signal; and obtaining a high frequency encoding parameter based on the synthesized excitation signal and the high band signal. Technical solutions in the embodiments of the present invention can improve an encoding or decoding effect.

Abstract: Apparatuses and methods are described to identify desired audio. A first input of an apparatus is configured to receive a main signal. A second input of the apparatus is configured to receive a reference signal. A normalizer is configured to normalize a compressed main signal by a compressed reference signal to create a normalized main signal. A single channel normalized voice threshold comparator is configured to receive as an input the normalized main signal and to output a desired voice activity detection signal.

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: In general, techniques are described for obtaining spherical harmonic coefficients (SHC). A device comprising a processor and a memory may be configured to perform the techniques. The processor may obtain a set of coefficients of a vector representative a distinct component of a sound field, the vector having been decomposed from SHC representative of the sound field. The processor may obtain a configuration mode by which to extract the coefficients, where the configuration mode indicates that the coefficients include coefficients corresponding to an order greater than an order of a basis function to which one or more of the spherical harmonic coefficients correspond and exclude at least one of the coefficients corresponding to a greater order. The processor may extract the coefficients of the vector based on the obtained configuration mode. The memory may be configured to store the non-zero set of the coefficients of the vector.

Abstract: There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary decoding method described is for reconstructing audio objects based on a data stream, wherein the data stream corresponds to a plurality of time frames, wherein the data stream comprises a plurality of side information instances, wherein the data stream further comprises, for each side information instance, transition data including two independently assignable portions which in combination define a point in time to begin a transition from a current reconstruction setting to a desired reconstruction setting specified by the side information instance, and a point in time to complete the transition.

Abstract: Present disclosure provide an encoding method and apparatus, which relate to the communications field and can perform proper quantization bit allocation for spectral coefficients of an audio signal, thereby improving quality of a signal obtained by a decoder by means of decoding. The method includes: after splitting spectral coefficients of a current data frame into subbands, acquiring quantized frequency envelope values of the subbands; modifying quantized frequency envelope values of subbands of a first quantity in the subbands; allocating quantization bits to the subbands according to modified quantized frequency envelope values of the subbands of the first quantity; quantizing a spectral coefficient of a subband to which a quantization bit is allocated in the subbands; and writing the quantized spectral coefficient of the subband to which a quantization bit is allocated into a bitstream.

Abstract: In general, techniques are described for indicating reusability of an index that determines a Huffman codebook used to code data associated with a vector in a spherical harmonics domain. The bitstream may comprise an indicator for whether to reuse, from a previous frame, at least one syntax element indicative of the index. The memory may be configured to store the bitstream.

Abstract: In general, techniques are described for obtaining decomposed versions of spherical harmonic coefficients. A device comprising a processor and a memory may be configured to perform the techniques. The processor may obtain a non-zero set of coefficients of a vector representative a distinct component of a sound field. The vector may have been decomposed from a plurality of spherical harmonic coefficients that describe the sound field. The processor may also obtain one of a plurality of configuration modes by which to extract the non-zero set of coefficients of the vector, where the one of the configuration modes indicates that the coefficients include all of the coefficients except for at least one of the coefficients. The processor may further extract the coefficients of the vector based on the obtained one of the configuration modes. The memory may be configured to store the non-zero set of the coefficients of the vector.

Abstract: A method and a device for encoding a high frequency signal, and a method and a device for decoding a high frequency signal are provided, which relate to encoding and decoding technology. The method for encoding a high frequency signal includes: determining a signal type of a high frequency signal of a current frame; smoothing and scaling time envelopes of the high frequency signal of the current frame and obtaining time envelopes of the high frequency signal of the current frame that require to be encoded, if the high frequency signal of the current frame is a non-transient signal and a high frequency signal of the previous frame is a transient signal; and quantizing and encoding the time envelopes of the high frequency signal of the current frame that require to be encoded, and frequency information and signal type information of the high frequency signal of the current frame.

Abstract: Systems, methods and articles of manufacture for outputting an audio effect on a remote device are disclosed. Embodiments select a device from a plurality of devices within a physical environment for use in outputting an audio effect. Upon determining that transmitting the audio effect as uncompressed data to the selected device would violate a predefined performance criteria, the audio effect is modified by determining, for each of a plurality of portions of the audio effect, a respective priority. Additionally, upon determining that a first portion of the plurality of portions of the audio effect is a lower priority, relative to a second portion of the audio effect, embodiments compress the first portion of the audio effect, while the second portion of the audio effect remains uncompressed. The modified audio effect is then transmitted to the selected device for playback.

Abstract: Multiple audio files may be synchronized using energy vectors produced from energy portions of individual frequency energy representations. Individual energy samples and time values of individual energy vectors may be compared using a multi-resolution framework to correlate energy samples and time values of multiple audio tracks to one another.

Abstract: A method and apparatus for efficiently processing data in various formats in a single instruction multiple data (“SIMD”) architecture is presented. Specifically, a method to unpack a fixed-width bit values in a bit stream to a fixed width byte stream in a SIMD architecture is presented. A method to unpack variable-length byte packed values in a byte stream in a SIMD architecture is presented. A method to decompress a run length encoded compressed bit-vector in a SIMD architecture is presented. A method to return the offset of each bit set to one in a bit-vector in a SIMD architecture is presented. A method to fetch bits from a bit-vector at specified offsets relative to a base in a SIMD architecture is presented. A method to compare values stored in two SIMD registers is presented.

Abstract: A device for embedding data upon a prediction coding of a multi-channel signal includes a storage unit to store a code book that includes a plurality of prediction parameter sets, each of the plurality of prediction parameter sets including a plurality of kinds of prediction parameters for a processing regarding the prediction coding. The device extracts a plurality of candidates of a prediction parameter set for the multi-channel signal from the code book, wherein the plurality of candidates are capable of suppressing a prediction error in the prediction coding within a predetermined range, converts an embedding object that is at least part of the data in accordance with a number corresponding to a number of the candidates, selects, from the plurality of candidates, the prediction parameter set corresponding to the converted embedding object, and multiplexes the selected prediction parameter set with coded data which are down-mixed from the multi-channel signal.

Abstract: In one embodiment of the present invention, a method of decoding an encoded audio bitstream and generating frequency bandwidth extension includes decoding the audio bitstream to produce a decoded low band audio signal and generate a low band excitation spectrum corresponding to a low frequency band. A sub-band area is selected from within the low frequency band using a parameter which indicates energy information of a spectral envelope of the decoded low band audio signal. A high band excitation spectrum is generated for a high frequency band by copying a sub-band excitation spectrum from the selected sub-band area to a high sub-band area corresponding to the high frequency band. Using the generated high band excitation spectrum, an extended high band audio signal is generated by applying a high band spectral envelope. The extended high band audio signal is added to the decoded low band audio signal to generate an audio output signal having an extended frequency bandwidth.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: The present disclosure envisages a computer implemented system for identifying significant speech frames within speech signals for facilitating speech recognition. The system receives an input speech signal having a plurality of feature vectors which is passed through a spectrum analyzer. The spectrum analyzer divides the input speech signal into a plurality of speech frames and computes a spectral magnitude of each of the speech frames. There is provided a suitability engine which is enabled to compute a suitability measure for each of the speech frames corresponding to spectral flatness measure (SFM), energy normalized variance (ENV), entropy, signal-to-noise ratio (SNR) and similarity measure. The suitability engine further computes a weighted suitability measure for each of the speech frames.

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: An apparatus for providing a speech probability estimation is provided. The apparatus includes a first speech probability estimator for estimating speech probability information indicating a first probability on whether a sound field of a scene includes speech or on whether the sound field of the scene does not include speech. Moreover, the apparatus includes an output interface for outputting the speech probability estimation depending on the speech probability information. The first speech probability estimator is configured to estimate the first speech probability information based on at least spatial information about the sound field or spatial information on the scene.

Abstract: Methods and apparatuses for addressing open space noise are disclosed. In one example, a method for masking open space noise includes outputting a test signal from a speaker, the test signal operable to measure a speech transmission quality of a transmission channel. The method includes receiving the test signal at a microphone, outputting a detected test signal, and processing the detected test signal to determine the speech transmission quality of the transmission channel. The method further includes adjusting an output level of a noise masking signal from the speaker responsive to the speech transmission quality.

Abstract: A computing device may receive or otherwise access a base audio layer and one or more enhancement audio layers. The computing device can reconstruct the retrieved base layer and/or enhancement layers into a single data stream or audio file. The local computing device may process audio frames in a highest enhancement layer retrieved in which the data can be validated (or a lower layer if the data in audio frames in the enhancement layer(s) cannot be validated) and build a stream or audio file based on the audio frames in that layer.

Abstract: The quality of encoded signals can be improved by reclassifying AUDIO signals carrying non-speech data as VOICE signals when periodicity parameters of the signal satisfy one or more criteria. In some embodiments, only low or medium bit rate signals are considered for re-classification. The periodicity parameters can include any characteristic or set of characteristics indicative of periodicity. For example, the periodicity parameter may include pitch differences between subframes in the audio signal, a normalized pitch correlation for one or more subframes, an average normalized pitch correlation for the audio signal, or combinations thereof. Audio signals which are re-classified as VOICED signals may be encoded in the time-domain, while audio signals that remain classified as AUDIO signals may be encoded in the frequency-domain.

Abstract: An apparatus for processing a decoded audio signal including a filter for filtering the decoded audio signal to obtain a filtered audio signal, a time-spectral converter stage for converting the decoded audio signal and the filtered audio signal into corresponding spectral representations, each spectral representation having a plurality of subband signals, a weighter for performing a frequency selective weighting of the filtered audio signal by a multiplying subband signals by respective weighting coefficients to obtain a weighted filtered audio signal, a subtractor for performing a subband-wise subtraction between the weighted filtered audio signal and the spectral representation of the decoded audio signal, and a spectral-time converter for converting the result audio signal or a signal derived from the result audio signal into a time domain representation to obtain a processed decoded audio signal.

Abstract: Described are methods and systems of identifying one or more fundamental frequency component(s) of an audio signal. The methods and systems may include any one or more of an audio event receiving step, a signal discretization step, a masking step, and/or a transcription step.

Abstract: An exemplary noise reduction system and method processes a speech signal that is delivered in a noisy channel or with ambient noise. Some exemplary embodiments of the system and method use filters to extract speech information, and focus on a subset of harmonics that are least corrupted by noise. Some exemplary embodiments disregard signal harmonics with low signal-to-noise ratio(s), and disregard amplitude modulations that are inconsistent with speech. An exemplary system and method processes a signal that focuses on a subset of harmonics that are least corrupted by noise, disregards the signal harmonics with low signal-to-noise ratio(s), and disregards amplitude modulations that are inconsistent with speech.

Abstract: An encoder for digital audio signals at a higher sample rate creates a stream for consumer distribution at a lower sampling rate, with compatibility for standard PCM players without a decoder. In conjunction with a suitable decoder, two enhanced playback options are supported, the first option allowing full lossless reconstruction of a noise-shaped higher sampling rate signal, the second option allowing lossy bandwidth extension even if an intervening transmission chain has truncated the least-significant-bits of the encoder's output signal.

Abstract: In an aspect, a user equipment (UE) is positioned within a driver operation zone of a vehicle, one or more UE-based user interfaces (UIs) are restricted and forwarded to a vehicle-based UI controller to permit access to the one or more UE-based features via a vehicle-based UI. In another aspect, the UE in the driver operation zone is engaged in hands-free speakerphone mode via a vehicle audio system of the vehicle, and an attempt to transition the UE to handset-based audio mode is blocked. In another aspect, when a handset-based audio capture and/or playback attempt of the UE is detected, the UE interacts with a vehicle audio system to temporarily reduce volume being output by one or more proximal speakers. In another aspect, the UE streams media to a media presentation device in its own zone or another zone of the vehicle for presentation thereon.

Abstract: Embodiments of client device and method for audio or video conferencing are described. An embodiment includes an offset detecting unit, a configuring unit, an estimator and an output unit. The offset detecting unit detects an offset of speech input to the client device. The configuring unit determines a voice latency from the client device to every far end. The estimator estimates a time when a user at the far end perceives the offset based on the voice latency. The output unit outputs a perceivable signal indicating that a user at the far end perceives the offset based on the time estimated for the far end. The perceivable signal is helpful to avoid collision between parties.

Abstract: Noise filling in perceptual transform audio codecs is improved by performing the noise filling with a spectrally global tilt, rather than in a spectrally flat manner.

Abstract: The present document relates to audio signal processing in general, and to the concealment of artifacts that result from loss of audio packets during audio transmission over a packet-switched network, in particular. A method (200) for concealing one or more consecutive lost packets is described. A lost packet is a packet which is deemed to be lost transform-based audio decoder. Each of the one or more lost packets comprises a set of transform coefficients. A set of transform coefficients is used by the transform-based audio decoder to generate a corresponding frame of a time domain audio signal. The method (200) comprises determining (205) for a current lost packet of the one or more lost packets a number of preceding lost packets from the one or more lost packets; wherein the determined number is referred to as a loss position.

Abstract: A device for producing a frequency-shifted audio signal based on an audio input signal is provided. The device has an interface and a frequency-shifting unit. The interface is configured for receiving the audio input signal. The frequency-shifting unit is configured for producing the frequency-shifted audio signal. The frequency-shifting unit is additionally configured to produce one of the second subband values based on one of the first subband values such that the second phase angle of this second subband value differs from the first phase angle of this first subband value by a phase angle difference, the phase angle difference being dependent on frequency information indicating by which frequency difference the audio input signal is to be shifted in order to obtain the frequency-shifted audio signal, and the phase angle difference being dependent on a frequency bandwidth of one of the first subbands.

Abstract: An audio coding device that performs predictive coding on a third-channel signal included in a plurality of channels in an audio signal according to a first-channel signal and a second-channel signal, which are included in the plurality of channels, and to a plurality of channel prediction coefficients included in a coding book, the device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, selecting channel prediction coefficients corresponding to the first-channel signal and the second-channel signal so that an error, which is determined by a difference between the third-channel signal before predictive coding and the third-channel signal after predictive coding, is minimized; and controlling the first-channel signal or the second-channel signal so that the error is further reduced.

Abstract: A method for encoding sound field signals includes allocating coding rate by application of a uniform criterion to all subbands of all signals in a joint process. An allocation criterion may be based on a comparison, in a given subband, between a spectral envelope of the signals to be encoded and a coding noise profile, wherein the noise profile may be a sum of a noise shape and a noise offset, which noise offset is computed on the basis of the coding bit budget. The rate allocation process may be combined with an energy-compacting orthogonal transform, for which there is proposed a parameterization susceptible of efficient coding and having adjustable directivity. In a further aspect, the invention provides a corresponding decoding method.

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

Abstract: Provided are a method and apparatus for encoding and decoding a high frequency signal by using a low frequency signal. The high frequency signal can be encoded by extracting a coefficient by linear predicting a high frequency signal, and encoding the coefficient, generating a signal by using the extracted coefficient and a low frequency signal, and encoding the high frequency signal by calculating a ratio between the high frequency signal and an energy value of the generated signal. Also, the high frequency signal can be decoded by decoding a coefficient, which is extracted by linear predicting a high frequency signal, and a low frequency signal, and generating a signal by using the decoded coefficient and the decoded low frequency signal, and adjusting the generated signal by decoding a ratio between the generated signal and an energy value of the high frequency signal.