Abstract: The present disclosure may provide a cooling plant system, a method of operating a cooling plant system and an energy storage and scheduling control system, the cooling plant system comprising one or more cooling towers; one or more thermal energy storage systems coupled to the one or more cooling towers; one or more electrical energy storage systems, the one or more electrical energy storage systems being coupled to the one or more thermal energy storage systems for selectively providing electrical energy to the one or more thermal energy storage systems; and an energy storage and scheduling control system coupled to the one or more thermal energy storage systems and the one or more electrical energy storage systems.

Abstract: The present disclosure may provide a cooling plant system, a method of operating a cooling plant system and an energy storage and scheduling control system, the cooling plant system comprising one or more cooling towers; one or more thermal energy storage systems coupled to the one or more cooling towers; one or more electrical energy storage systems, the one or more electrical energy storage systems being coupled to the one or more thermal energy storage systems for selectively providing electrical energy to the one or more thermal energy storage systems; and an energy storage and scheduling control system coupled to the one or more thermal energy storage systems and the one or more electrical energy storage systems, the energy storage and scheduling control system being configured to control operations of the one or more thermal energy storage systems and the one or more electrical energy storage systems via a scheduling of the operations per time slot, the operations including the selective provision of el

Abstract: According to various embodiments, a testing apparatus may be provided. The testing apparatus may include: a cell pair comprising two l-bit memory cells configured to represent a stored pattern of l-bit; and a converter configured to convert a query pattern of l-bit into a pair of voltages defined such that when applied to gates of the cell pair, the voltages make the cell pair into high resistance mode when the query pattern matches the stored pattern and into low resistance mode when the query pattern does not match the stored pattern.

Abstract: A compression device which may include: a subset determination circuit configured to determine a subset of a traffic indication map, the traffic indication map including a plurality of bits, each bit indicating whether data to be transmitted to a respective pre-determined radio communication terminal is present in an access point; a pre-determined bit value determination circuit configured to determine whether the subset includes a bit of a pre-determined bit value; and a compressed string generation circuit configured to insert, if the subset includes a bit of the pre-determined bit value, into a compressed string the subset and an indicator indicating that the subset includes a bit of the pre-determined bit value, and further configured to include, if the subset does not includes a bit of the pre-determined bit value, into the compressed string an indicator indicating that the subset does not include a bit of the pre-determined bit value.

Abstract: A compression device which may include: a subset determination circuit configured to determine a subset of a transmission indication map, the transmission indication map including a plurality of bits, each bit indicating whether data to be transmitted to a respective pre-determined radio communication terminal is present in an access point; a pre-determined bit value determination circuit configured to determine whether the subset includes a bit of a pre-determined bit value; and a compressed string generation circuit configured to insert, if the subset includes a bit of the pre-determined bit value, into a compressed string an indicator indicating that the subset includes a bit of the pre-determined bit value and the subset and further configured to include, if the subset does not includes a bit of the pre-determined bit value, into the compressed string an indicator indicating that the subset does not include a bit of the pre-determined bit value.

Abstract: An electroacoustic channel soundfield is altered. An audio signal is applied by an electromechanical transducer to an acoustic space, causing air pressure changes therein. Another audio signal is obtained by a second electromechanical transducer, responsive to air pressure changes in the acoustic space. A transfer function estimate of the electroacoustic channel is established, responsive to the second audio signal and part of the first audio signal. The transfer function estimate is derived to be adaptive to temporal variations in the electroacoustic channel transfer function. Filters are obtained with transfer functions based on the transfer function estimate. Part of the first audio signal is filtered therewith.

Abstract: A method for enhancing speech components of an audio signal composed of speech and noise components processes subbands of the audio signal, the processing including controlling the gain of the audio signal in ones of the subbands, wherein the gain in a subband is controlled at least by processes that convey either additive/subtractive differences in gain or multiplicative ratios of gain so as to reduce gain in a subband as the level of noise components increases with respect to the level of speech components in the subband and increase gain in a subband when speech components are present in subbands of the audio signal, the processes each responding to subbands of the audio signal and controlling gain independently of each other to provide a processed subband audio signal.

Abstract: Speech enhancement based on a psycho-acoustic model is disclosed that is capable of preserving the fidelity of speech while sufficiently suppressing noise including the processing artifact known as “musical noise”.

Abstract: A dual microphone voice activity detector system is presented. A voice activity detector system estimates the signal level and noise level at each microphone. A level differential between the two microphones of nearby sounds such as the signal is greater than the level differential of more distant sounds such as the noise. Thus, the voice activity detector detects the presence of nearby sounds.

Abstract: Enhancing speech components of an audio signal composed of speech and noise components includes controlling the gain of the audio signal in ones of its subbands, wherein the gain in a subband is reduced as the level of estimated noise components increases with respect to the level of speech components, wherein the level of estimated noise components is determined at least in part by (1) comparing an estimated noise components level with the level of the audio signal in the subband and increasing the estimated noise components level in the subband by a predetermined amount when the input signal level in the subband exceeds the estimated noise components level in the subband by a limit for more than a defined time, or (2) obtaining and monitoring the signal-to-noise ratio in the subband and increasing the estimated noise components level in the subband by a predetermined amount when the signal-to-noise ratio in the subband exceeds a limit for more than a defined time.

Abstract: A method for determining a number of bits for encoding an audio signal comprising a core audio signal portion and a residual audio signal portion is described that comprises selecting, from the residual audio signal portion, a reference residual audio signal portion and at least one candidate residual audio signal portion; comparing the reference residual audio signal portion with the candidate residual audio signal portion; and determining the number of bits for encoding the audio signal depending on the result of the comparison.

Abstract: A method for encoding a digital signal into a scalable bitstream comprising quantizing the digital signal, and encoding the quantized signal to form a core-layer bitstream, performing an error mapping based on the digital signal and the core-layer bitstream to remove information that has been encoded into the core-layer bitstream, resulting in an error signal, bit-plane coding the error signal based on perceptual information of the digital signal, resulting in an enhancement-layer bitstream, wherein the perceptual information of the digital signal is determined using a perceptual model, and multiplexing the core-layer bitstream and the enhancement-layer bitstream, thereby generating the scalable bitstream.

Abstract: In one embodiment, a method for providing an encoded digital signal is described comprising determining, for each data frame of a plurality of data frames of a digital signal, a plurality of pairs of an encoding data volume and an encoding quality, wherein each pair of an encoding data volume and an encoding quality specifies the encoding data volume required for achieving the encoding quality; determining for each data frame at least one or more interpolations between the plurality of determined pairs; determining a multi-frame relationship between encoding quality and encoding data volume required to encode the plurality of data frames at the encoding quality based on a combination of the at least one or more interpolations for the plurality of data frames; determining an encoding quality for the plurality of data frames based on the relationship; and providing at least one data frame of the plurality of data frames encoded at the determined encoding quality.

Abstract: The invention relates to the coding of audio signals that may include both speech-like and non-speech-like signal components. It describes methods and apparatus for code excited linear prediction (CELP) audio encoding and decoding that employ linear predictive coding (LPC) synthesis filters controlled by LPC parameters, a plurality of codebooks each having codevectors, at least one codebook providing an excitation more appropriate for non-speech-like signals and at least one codebook providing an excitation more appropriate for speech-like signals, and a plurality of gain factors, each associated with a codebook. The encoding methods and apparatus select from the codebooks codevectors and/or associated gain factors by minimizing a measure of the difference between the audio signal and a reconstruction of the audio signal derived from the codebook excitations. The decoding methods and apparatus generate a reconstructed output signal from the LPC parameters, codevectors, and gain factors.

Abstract: According to one embodiment, a method for transmitting a digital signal is provided that includes dividing data representing the digital signal into a plurality of data blocks, processing each data block in accordance with a desired amount of data included in the data block, determining, for each processed data block, the size of the processed data block, generating a message including, in a message body of the message, the processed data blocks and, for each data block, a message field specifying the size of the processed data block and transmitting the message.

Abstract: A speech enhancement method operative for devices having limited available memory is described. The method is appropriate for very noisy environments and is capable of estimating the relative strengths of speech and noise components during both the presence as well as the absence of speech.

Abstract: A method for transforming a digital signal from the time domain into the frequency domain and vice versa using a transformation function comprising a transformation matrix, the digital signal comprising data symbols which are grouped into a plurality of blocks, each block comprising a predefined number of the data symbols. The method includes the process of transforming two blocks of the digital signal by one transforming element, wherein the transforming element corresponds to a block-diagonal matrix comprising two sub matrices, wherein each sub-matrix comprises the transformation matrix and the transforming element comprises a plurality of lifting stages and wherein each lifting stage comprises the processing of blocks of the digital signal by an auxiliary transformation and by a rounding unit.

Abstract: A method for performing a domain transformation of a digital signal from the time domain into the frequency domain and vice versa, the method including performing the transformation by a transforming element, the transformation element comprising a plurality of lifting stages, wherein the transformation corresponds to a transformation matrix and wherein at least one lifting stage of the plurality of lifting stages comprises at least one auxiliary transformation matrix and a rounding unit, the auxiliary transformation matrix comprising the transformation matrix itself or the corresponding transformation matrix of lower dimension. The method further comprising performing a rounding operation of the signal by the rounding unit after the transformation by the auxiliary transformation matrix.

Abstract: Systems and methods for scalably encoding and decoding coded data are presented. One exemplary method for scalably coding data includes classifying, based upon at least one predetermined criteria, each of the plurality of data received as either (i) perceptually relevant data or (ii) perceptually irrelevant data. The perceptually relevant data is scalably coded, and the perceptually irrelevant data is non-scalably coded. Subsequently, the scalably coded perceptually relevant data and the non-scalably coded perceptually irrelevant are combined into a coded data stream for transmission.

Abstract: An electroacoustic channel soundfield is altered. An audio signal is applied by an electromechanical transducer to an acoustic space, causing air pressure changes therein. Another audio signal is obtained by a second electromechanical transducer, responsive to air pressure changes in the acoustic space. A transfer function estimate of the electroacoustic channel is established, responsive to the second audio signal and part of the first audio signal. The transfer function estimate is derived to be adaptive to temporal variations in the electroacoustic channel transfer function. Filters are obtained with transfer functions based on the transfer function estimate. Part of the first audio signal is filtered therewith.