Abstract: A method and device for decoding a signal, where the method includes: obtaining an average quantity of allocated bits per spectral coefficient of a sub-band of a current frame of the audio signal, wherein the sub-band includes a plurality of spectral coefficients; obtaining a noise filling gain for the sub-band when the average quantity of allocated bits per spectral coefficient is less than a classification threshold; reconstructing, according to the noise filling gain, at least some of the spectral coefficients to generate reconstructed spectral coefficients when the average quantity of allocated bits per spectral coefficient is less than a classification threshold; obtaining a frequency domain signal according to the reconstructed spectral coefficients; and generating a time domain signal based on the frequency domain signal. 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 combination of a block-oriented encoder and decoder with a modified dataset identifier that is associated with an encoded block size are used to perform block-based encoding and decoding operations. The encoding process may generate optional metadata that includes an array of encoded block sizes to support random access into the stream or group of encoded blocks during the decoding process. The modified dataset identifier associates the original dataset identifier with the block size used by the encoder.

Abstract: The system of the present technology includes an embodiment that provides a host audio, video and control operating system configured to establish or interact with one or more virtual machines, each with a guest operating system.

Abstract: System and techniques for reduced multiplicative complexity discrete cosine transform (DCT) circuitry are described herein. An input data set can be received and, upon the input data set, a self-recursive DCT technique can be performed to produce a transformed data set. Here, the self-recursive DCT technique is based on a product of factors of a specified type of DCT technique. Recursive components of the technique are of the same DCT type as that of the DCT technique. The transformed data set can then be produced to a data consumer.

Abstract: Personal audio systems and methods are disclosed. A personal audio system includes a voice activity detector to determine whether or not an ambient audio stream contains voice activity, a pitch estimator to determine a frequency of a fundamental component of an annoyance noise contained in the ambient audio stream, and a filter bank to attenuate the fundamental component and at least one harmonic component of the annoyance noise to generate a personal audio stream. The filter bank implements a first filter function when the ambient audio stream does not contain voice activity, or a second filter function when the ambient audio stream contains voice activity.

Abstract: A method for pyramid vector quantization deindexing of coefficients derived from an audio or video signal includes receiving a first sign and an input index as codewords from an ingoing bit stream. The first sign and the input index represent the coefficients, and the first sign is a sign of a first or last non-zero coefficient in an output vector to be created. The method includes deindexing the input index into the output vector with a pyramid vector quantization de-enumeration scheme, assigning a sign of the first or last non-zero coefficient in the output vector according to the received first sign, and outputting the output vector.

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.

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 example apparatus includes a memory configured to store the audio data; and one or more processors in communication with the memory, the one or more processors configured to: obtain, for each of a plurality of subbands of audio data, a respective energy scalar and a respective residual identifier; determine overall distortion levels for a plurality of candidate subband pulse allocations for performing pyramid vector dequantization (PVdQ) of the residual identifiers; select, from the plurality of subband pulse allocations and based on the overall distortion levels, a candidate subband pulse allocation; and perform, using the candidate subband pulse allocation, PVdQ on the residual identifiers to reconstruct a residual vector for each subband.

Abstract: An encoder for encoding frequency transform coefficients of a harmonic audio signal include the following elements: A peak locator configured to locate spectral peaks having magnitudes exceeding a predetermined frequency dependent threshold. A peak region encoder configured to encode peak regions including and surrounding the located peaks. A low-frequency set encoder configured to encode at least one low-frequency set of coefficients outside the peak regions and below a crossover frequency that depends on the number of bits used to encode the peak regions. A noise-floor gain encoder configured to encode a noise-floor gain of at least one high-frequency set of not yet encoded coefficients outside the peak regions.

Abstract: An audio signal transmission device for encoding an audio signal includes an audio encoding unit that encodes an audio signal and a side information encoding unit that calculates and encodes side information from a look-ahead signal. An audio signal receiving device for decoding an audio code and outputting an audio signal includes: an audio code buffer that detects packet loss based on a received state of an audio packet, an audio parameter decoding unit that decodes an audio code when an audio packet is correctly received, a side information decoding unit that decodes a side information code when an audio packet is correctly received, a side information accumulation unit that accumulates side information obtained by decoding a side information code, an audio parameter missing processing unit that outputs an audio parameter upon detection of audio packet loss, and an audio synthesis unit that synthesizes decoded audio from the audio parameter.

Abstract: A method and device for decoding a signal, where the method for decoding the signal includes obtaining spectral coefficients of sub-bands from a received bitstream by 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 decoding and is in the sub-band with unsaturated bit allocation to restore the spectral coefficient that has not been obtained by decoding, and obtaining a frequency domain signal according to the spectral coefficients obtained by 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: Personal audio systems and methods are disclosed. A personal audio system includes a voice activity detector to determine whether or not an ambient audio stream contains voice activity, a pitch estimator to determine a frequency of a fundamental component of an annoyance noise contained in the ambient audio stream, and a filter bank to attenuate the fundamental component and at least one harmonic component of the annoyance noise to generate a personal audio stream. The filter bank implements a first filter function when the ambient audio stream does not contain voice activity, or a second filter function when the ambient audio stream contains voice activity.

Abstract: The present technology relates to an encoding device and a method, a decoding device and a method, and a program that enables acquisitions of high-quality sound even in a resource-poor setting. A demultiplexer demultiplexes a supplied code string, to obtain the quantized low-band spectrum, the spectral characteristic code, and the quantized expansion coefficient(s). At this point, the code string includes a single quantized expansion coefficient or quantized expansion coefficients of the respective bands in the high band depending on the spectral characteristic code. A spectral inverse quantization unit obtains the low-band spectrum by inversely quantizing the quantized low-band spectrum. An expansion coefficient inverse quantization unit obtains the expansion coefficient(s) by inversely quantizing the quantized expansion coefficient(s).

Abstract: Techniques for using machine learning models to approximate a user in a communication are described. For example, a method of initiating a communication link with an edge device to exchange audio data; receiving a compressed audio data from the edge device; re-encoding the received compressed audio data using a re-encoding machine learning model to approximate a voice; and outputting the approximated voice is detailed.

Abstract: A method for mixing, processing and enhancing signals using signal decomposition is presented. A method for improving sorting of decomposed signal parts using cross-component similarity is also provided.

Abstract: An importance metric, based at least in part on an energy metric, may be determined for each of a plurality of received audio objects. Some methods may involve: determining a global importance metric for all of the audio objects, based, at least in part, on a total energy value calculated by summing the energy metric of each of the audio objects; determining an estimated quantization bit depth and a quantization error for each of the audio objects; calculating a total noise metric for all of the audio objects, the total noise metric being based, at least in part, on a total quantization error corresponding with the estimated quantization bit depth; calculating a total signal-to-noise ratio corresponding with the total noise metric and the total energy value; and determining a final quantization bit depth for each of the audio objects by applying a signal-to-noise ratio threshold to the total signal-to-noise ratio.

Abstract: A plurality of bits are received, and the plurality of bits are grouped and transcoded into a plurality of bit blocks, where the plurality of bit blocks includes a first set of bit blocks and a second set of bit blocks. A set of transcoded bit blocks is generated by transcoding the first set of bit blocks, where the set of transcoded bit blocks contains fewer bits than the first set of bit blocks. An aggregated bit block is generated at least by aggregating the set of transcoded bit blocks and the second set of bit blocks. All bits in the aggregated bit block are encoded, and the encoded bits in the aggregated bit block are modulated to generate a communication frame.

Abstract: Embodiments are generally directed to a low-overhead mechanism to detect address faults in ECC-protected memories. An embodiment of an apparatus includes a memory array; an error correction code (ECC) encoder for the memory array to encode ECC values based on a data value and a respective address value for the data value; and an ECC decoder for the memory array to decode ECC values that are based on data values and respective addresses for the data values; wherein the apparatus is to detect and correct an error in an address value based on an ECC value, address value, and data value stored in the memory.

Abstract: A method includes determining whether a rate distortion cost of a compressed video is above a cost threshold, the compressed video being encoded using a first constant rate factor (CRF). Upon determining the quality of a compressed video is above a cost threshold calculating a second CRF based on the first CRF, and encoding a video associated with the compressed video using the second CRF. Upon determining the quality of a compressed video is below a cost threshold encoding the video using the first CRF and a target bitrate.

Abstract: A system for generating comfort noise for a stream of frames carrying an audio signal includes frame characterizing logic configured to generate a set of filter parameters characterising the frequency content of a frame; an analysis filter adapted using the filter parameters and configured to filter the frame so as to generate residual samples; an analysis controller configured to cause the residual samples to be stored in a store responsive to receiving an indication that the frame does not comprise speech; and a synthesis controller operable to select stored residual samples from the store and cause a synthesis filter, inverse to the analysis filter and adapted using filter parameters generated by the frame characterizing logic for one or more frames not comprising speech, to filter the selected residual samples so as to generate a frame of comfort noise.

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: The present invention relates to a multi-channel digital microphone, comprising: an array pickup unit and a processing unit, the array pickup unit comprises a plurality of pickup modules, arranged in array form, wherein each of the pick-up modules picks up the audio signals of the spatial region component, the array pickup unit converts a plurality of the audio signals of the spatial region component to a plurality of digital signal and outputs it; the pickup unit is connected to the processing unit, configured to extract and denoise the plurality of digital signals. The present invention adds a spatial region based on the time region and frequency region so that a pick-up de-noising process can be done for these received signals from different spatial directions.

Abstract: A method for spectrum recovery in spectral decoding of an audio signal, comprises obtaining of an initial set of spectral coefficients representing the audio signal, and determining a transition frequency. The transition frequency is adapted to a spectral content of the audio signal. Spectral holes in the initial set of spectral coefficients below the transition frequency are noise filled and the initial set of spectral coefficients are bandwidth extended above the transition frequency. Decoders and encoders being arranged for performing part of or the entire method are also illustrated.

Abstract: Described are techniques for determining whether a received audio signal is corrupted or a product of noise, prior to decoding the entire signal. A received audio signal may include both initialization data and encoded payload data. Correspondence between the initialization data and confirmation data may be determined. If the correspondence indicates a match, the audio receiver may continue decoding the audio signal to access the payload data. If the correspondence does not indicate a match, the audio receiver may cease decoding, which may conserve power, computing resources, and time, ensuring that subsequent audio signals are not missed while the audio receiver decodes a false or corrupted signal.

Abstract: Embodiments of the present disclosure provide signal encoding and decoding methods and devices. The method includes: determining, a quantity k of subbands to be encoded, where i is a positive number, and k is a positive integer; selecting, according to quantized envelopes of all subbands, k subbands from all the subbands, or selecting k subbands from all subbands according to a psychoacoustic model; and performing a first-time encoding operation on spectral coefficients of the k subbands. In the embodiments of the present disclosure, the quantity k of subbands to be encoded is determined according to the quantity of available bits and the first saturation threshold, and encoding is performed on the k subbands that are selected from all the subbands, instead of on an entire frequency band, which can reduce spectrum holes of a signal obtained through decoding, and therefore, can improve auditory quality of an output signal.

Abstract: Methods, apparatuses, and systems for using Digital Signal Processing (DSP) to detect, measure, and/or compensate for, antenna load mismatch are described. One method provides a test sweep signal as digital input to a Digital-to-Analog Converter (DAC) in a transmit path ending with an antenna, and determines a transmit transfer function based on the digital input and digital output from an Analog-to-Digital Converter (ADC) in a loopback path which is electrically connected in parallel with a load impedance of the antenna. The determined transmit transfer function may be used for any of detecting, measuring, and compensating for, antenna mismatch.

Abstract: The present document relates to the determination of a 200 bitrate related to an encoded bitstream, and describes a method for determining an estimate of a bitrate of a bitstream comprising a sequence of frames comprising a varying number of bitsand corresponding to excerpts of an audio and/or video signal. At least two frames of the sequence of frames comprise a parameter indicative of a processing delay for the corresponding frame. The method comprises determining: a total number of bits for a subsequence of frames from the bitstream; a corrected number 201 of frames based on a number of frames comprised within the subsequence and the parameters of at least two frames of the subsequence; and a lower bitrate bound and an upper bitrate bound of the bitrate based on the total number of bits, the corrected number of frames and a frame rate of the bitstream.

Abstract: The present technology relates to an encoding device and a method, a decoding device and a method, and a program that enables acquisitions of high-quality sound even in a resource-poor setting. A demultiplexer demultiplexes a supplied code string, to obtain the quantized low-band spectrum, the spectral characteristic code, and the quantized expansion coefficient(s). At this point, the code string includes a single quantized expansion coefficient or quantized expansion coefficients of the respective bands in the high band depending on the spectral characteristic code. A spectral inverse quantization unit obtains the low-band spectrum by inversely quantizing the quantized low-band spectrum. An expansion coefficient inverse quantization unit obtains the expansion coefficient(s) by inversely quantizing the quantized expansion coefficient(s).

Abstract: A device includes a decoder configured to receive an encoded audio signal at a decoder and to generate a synthesized signal based on the encoded audio signal. The device further includes a classifier configured to classify the synthesized signal based on at least one parameter determined from the encoded audio signal.

Abstract: An automatic speech recognition system and a method performed by an automatic speech recognition system are provided. The method includes performing at least two passes of speech activity detection on an acoustic utterance uttered by a speaker. The at least two passes include an initial pass and a subsequent pass. The method further includes estimating at least one of feature statistics and transforms for acoustic feature extraction and acoustic modeling based on an output of an initial pass. The method further includes performing automatic speech recognition using an output of the subsequent pass while bypassing an output of the initial pass to recognize the acoustic utterance.

Abstract: The invention relates to a method for extending the frequency band of an audio signal during a decoding or improvement process comprising a step of decoding or extracting, in a first so-called low frequency band, an excitation signal and coefficients of a linear prediction filter. The method comprises the following steps: —obtaining a signal extended in at least a second frequency band higher than the first frequency band from an oversampled excitation signal extended in at least a second frequency band; —scaling the extended signal by means of a gain defined by subframe on the basis of an energy ratio of a frame and of a subframe; —filtering said scaled extended signal with a linear prediction filter of which the coefficients are derived from the coefficients of the low frequency band filter. The invention also relates to a frequency band extension device implementing the described method and a decoder comprising such a device.

Abstract: An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighborhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams.

Abstract: A method and computer program product for dynamically balancing the active noise cancellation value at a first and second earphone interface of a headphone device. The method enables determination of a first active noise cancellation value of an anti-noise output signal at a first interface. The method determines a second active noise cancellation value of the anti-noise output signal at a second interface. A comparison is made between the first active noise cancellation value and the second active noise cancellation value. In response to the first active noise cancellation value and the second active noise cancellation value being outside of a predetermined margin of each other, at least one adjustable parameter associated with the anti-noise output signal is dynamically adjusted to balance active noise cancellation values at the first and second headphone nterface.

Abstract: An audio signal transmission device for encoding an audio signal includes an audio encoding unit that encodes an audio signal and a side information encoding unit that calculates and encodes side information from a look-ahead signal. An audio signal receiving device for decoding an audio code and outputting an audio signal includes: an audio code buffer that detects packet loss based on a received state of an audio packet, an audio parameter decoding unit that decodes an audio code when an audio packet is correctly received, a side information decoding unit that decodes a side information code when an audio packet is correctly received, a side information accumulation unit that accumulates side information obtained by decoding a side information code, an audio parameter missing processing unit that outputs an audio parameter upon detection of audio packet loss, and an audio synthesis unit that synthesizes decoded audio from the audio parameter.

Abstract: The invention relates to an encoder and a decoder and methods therein for supporting split gain shape vector encoding and decoding. The method performed by an encoder, where the encoding of each vector segment is subjected to a constraint related to a maximum number of bits, BMAX, allowed for encoding a vector segment. The method comprises, determining an initial number, Np_init, of segments for a target vector x; and further determining an average number of bits per segment, BAVG, based on a vector bit budget and Np_init. The method further comprises determining a final number of segments to be used, for the vector x, in the gain shape vector encoding, based on energies of the Np_init segments and a difference between BMAX and BAVG. The performing of the method enables an efficient allocation of the bits of the bit budget over the target vector.

Abstract: A method and apparatus for providing signal processing coefficients for processing an input signal at a predetermined signal processing sampling rate, wherein the input signal is received at an input signal sampling rate, the method comprising the steps of computing a correlation or covariance function based on the received input signal at the input signal sampling rate to provide correlation or covariance coefficients at the input signal sampling rate, re-sampling the computed correlation or covariance coefficients having the input signal sampling rate to provide correlation or covariance coefficients at the predetermined signal processing sampling rate, and calculating the signal processing coefficients based on the correlation or covariance coefficients at the predetermined signal processing sampling rate.

Abstract: A better compromise between a too high bitstream and decoding overhead on the one hand and flexibility of frame element positioning on the other hand is achieved by arranging that each of the sequence of frames of the bitstream has a sequence of N frame elements and, on the other hand, the bitstream has a configuration block having a field indicating the number of elements N and a type indication syntax portion indicating, for each element position of the sequence of N element positions, an element type out of a plurality of element types with, in the sequences of N frame elements of the frames, each frame element being of the element type indicated, by the type indication portion, for the respective element position at which the respective frame element is positioned within the sequence of N frame elements of the respective frame in the bitstream.

Abstract: A method, medium, and system scalably encoding/decoding audio/speech. The method includes splitting an input signal into a low frequency band signal that is lower than a predetermined frequency and a high frequency band signal that is higher than the predetermined frequency, scalably encoding the split low frequency band signal into a core layer and one or more extension layers and then decoding the encoded core layer and the encoded extension layers, generating an error signal by using the split low frequency band signal and a decoded signal of the encoded core layer and the encoded extension layers, and encoding the error signal and the high frequency band signal into a signal-to-noise ratio (SNR) enhancement layer and a bandwidth extension layer.

Abstract: Respiratory rate can be calculated from an acoustic input signal using time domain and frequency domain techniques. Confidence in the calculated respiratory rate can also be calculated using time domain and frequency domain techniques. Overall respiratory rate and confidence values can be obtained from the time and frequency domain calculations. The overall respiratory rate and confidence values can be output for presentation to a clinician.

Abstract: The present document relates to audio coding systems. In particular, the present document relates to efficient methods and systems for parametric multi-channel audio coding. An audio encoding system (500) configured to generate a bitstream (564) indicative of a downmix signal and spatial metadata for generating a multi-channel upmix signal from the downmix signal is described. The system (500) comprises a downmix processing unit (510) configured to generate the downmix signal from a multi-channel input signal (561); wherein the downmix signal comprises m channels and wherein the multi-channel input signal (561) comprises n channels; n, m being integers with m<n. Furthermore, the system (500) comprises a parameter processing unit (520) configured to determine the spatial metadata from the multi-channel input signal (561).

Abstract: Systems, apparatuses, and methods for implementing a low power decimator. A decimator may receive a plurality of input samples from a digital microphone. The decimator may include one or more coefficient tables for storing values combining two or more filter coefficients for filtering the received samples. The decimator may utilize a concatenation of multiple samples to perform a lookup of a corresponding coefficient table. The coefficient tables may store only the necessary non-redundant values for all coefficient combinations which can be applied to the multiple samples. The result of the lookup of the coefficient table may have its sign inverted or be zeroed based on the values of the multiple samples.

Abstract: The invention provides a highly efficient technique for encoding a multi-channel audio signal. The invention relies on the basic principle of encoding a first signal representation of one or more of the multiple channels in a first encoder and encoding a second signal representation of one or more of the multiple channels in a second, multi-stage, encoder. This procedure is significantly enhanced by providing a controller for adaptively allocating a number of encoding bits among the different encoding stages of the second, multi-stage, encoder in dependence on multi-channel audio signal characteristics.

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: Personal audio systems and methods are disclosed. A personal audio system includes a voice activity detector to determine whether or not an ambient audio stream contains voice activity, a pitch estimator to determine a frequency of a fundamental component of an annoyance noise contained in the ambient audio stream, and a filter bank to attenuate the fundamental component and at least one harmonic component of the annoyance noise to generate a personal audio stream. The filter bank implements a first filter function when the ambient audio stream does not contain voice activity, or a second filter function when the ambient audio stream contains voice activity.

Abstract: A quasi-cyclic LDPC encoding apparatus is disclosed wherein a matrix H of the form [0 T; D E] is used, where T is a triangular matrix and D and E are arbitrary matrices selected to improve encoding performance. T and E vary with the size of an encoded data word whereas D is maintained constant. T and E are sparse such that encoding operations performed on them are computationally simple. Likewise D and its inverse are constant and pre-computed further reducing computation. T, E, and D and the inverse of D may be constrained to be quasi-cyclic, which reduces storage required to represent them and enables the performance of encoding operations using shift registers.

Abstract: The present invention relates to a signal processing apparatus and a signal processing method, an encoder and an encoding method, a decoder and a decoding method, and a program capable of reproducing music signal having a better sound quality by expansion of frequency band. A sampling frequency conversion unit converts a sampling frequency of an input signal, and a sub-band division circuit divides the input signal after the sampling conversion into sub-band signals of sub-bands having the number corresponding to the sampling frequency. A pseudo high band sub-band power calculation circuit calculates pseudo high band sub-band powers based on low band signals of the input signal and coefficient tables having coefficients for the respective high band sub-bands. A pseudo high band sub-band power difference calculation circuit compares high band sub-band powers and the pseudo high band sub-band powers to each other and selects a coefficient table from plural coefficient tables.

Abstract: There is provided a communication device including a network interface connected to a network, a packet collection unit configured to collect packets transmitted from the network interface and packets received by the network interface, a packet recording unit configured to select a packet to be recorded from the packets collected by the packet collection unit and to record the packet as recorded information, and a packet analysis unit configured to detect discontinuity of a packet ID of the packets collected by the packet collection unit. When the packet analysis unit detects discontinuity of a packet ID, the packet recording unit retains the recorded information that has been recorded.

Abstract: The present document relates to methods and systems for audio encoding. In particular, the present document relates to methods and systems for fast audio encoding using a parallel system architecture. A frame-based audio encoder (300, 400, 500, 600) comprising K parallel transform units (303, 403) is described; wherein each of the K parallel transform units (303, 403) is configured to transform a respective one of a group of K frames (305) of an audio signal (101) into a respective one of K sets of frequency coefficients; wherein K>1; wherein each of the K frames (305) comprises a plurality of samples of the audio signal (101).

Abstract: A personal listening system has an active noise control (ANC) controller that produces an anti-noise signal. A head worn audio device for a user has a speaker to convert the anti-noise signal into anti-noise, an error microphone, and a reference microphone. The controller uses signals from the error and reference microphones to produce the anti-noise signal in accordance with an adaptive filter algorithm that has an adjustable parameter which changes so as to move the point at which acoustic cancellation occurs from the error microphone and closer to the user's eardrum. Other embodiments are also described and claimed.