Abstract: The present invention relates to hAM15-52 analogues with improved amylin receptor (hAMY3R) potency (hAMY3R-EC50?250 pM) and which are largely based on the sequence of the human adrenomedullin fragment hAM15-52. The invention further relates to hAM15-52 analogues that are selective amylin receptor (hAMY3R) agonists (hAMY3R-EC50?250 pM and an hAM1R-EC50?25 nM) and which are largely based on the sequence of the human adrenomedullin fragment hAM15-52. The hAM15-52 analogues according to the invention maintain the good physical stability of hAM15-52. The invention further relates to pharmaceutical compositions comprising such polypeptides and their use in the treatment of a medical condition such as obesity, NASH and/or diabetes.

Abstract: Methods and apparatuses for automatic analysis and optimization of an audio signal are described herein. An example method may comprise receiving, by an audio signal optimizer, a first indication to perform an audio signal optimization, receiving an audio signal from an input device, recording a sample of the audio signal, analyzing the sample of the audio signal for at least one audio parameter, and performing, based on an analysis of the sample of the audio signal, the audio signal optimization of the audio signal, wherein the audio signal optimization comprises a configuration of a gain level of the audio signal.

Abstract: A microphone may comprise a microphone element for detecting sound, and a digital signal processor configured to process a first audio signal that is based on the sound in accordance with a selected one of a plurality of digital signal processing (DSP) modes. Each of the DSP modes may be for processing the first audio signal in a different way. For example, the DSP modes may account for distance of the person speaking (e.g., near versus far) and/or desired tone (e.g., darker, neutral, or bright tone). At least some of the modes may have, for example, an automatic level control setting to provide a more consistent volume as the user changes their distance from the microphone or changes their speaking level, and that may be associated with particular default (and/or adjustable) values of the parameters attack, hold, decay, maximum gain, and/or target gain, each depending upon which DSP is being applied.

Abstract: A microphone may comprise a microphone element for detecting sound, and a digital signal processor configured to process a first audio signal that is based on the sound in accordance with a selected one of a plurality of digital signal processing (DSP) modes. Each of the DSP modes may be for processing the first audio signal in a different way. For example, the DSP modes may account for distance of the person speaking (e.g., near versus far) and/or desired tone (e.g., darker, neutral, or bright tone). At least some of the modes may have, for example, an automatic level control setting to provide a more consistent volume as the user changes their distance from the microphone or changes their speaking level, and that may be associated with particular default (and/or adjustable) values of the parameters attack, hold, decay, maximum gain, and/or target gain, each depending upon which DSP is being applied.

Abstract: A microphone having a multi-channel mixer and one or more connectors usable to receive and/or output audio. Any of the connectors may be used as an input connector, as an output connector, or configured to be switchable between being an input and an output connector. The user of the microphone may be able to conveniently use one or more of the connectors to expand the microphone to become part of a larger setup that uses multiple microphones. For example, an XLR connector of the microphone may be passive, and may be configured such that a user can daisy chain the output from an XLR connector of another microphone into an XLR connector of the microphone. In such an arrangement, an output based on one or both of the microphones may be output through another connector of the microphone, such as a USB connector.

Abstract: A microphone may comprise a microphone element for detecting sound, and a digital signal processor configured to process a first audio signal that is based on the sound in accordance with a selected one of a plurality of digital signal processing (DSP) modes. Each of the DSP modes may be for processing the first audio signal in a different way. For example, the DSP modes may account for distance of the person speaking (e.g., near versus far) and/or desired tone (e.g., darker, neutral, or bright tone). At least some of the modes may have, for example, an automatic level control setting to provide a more consistent volume as the user changes their distance from the microphone or changes their speaking level, and that may be associated with particular default (and/or adjustable) values of the parameters attack, hold, decay, maximum gain, and/or target gain, each depending upon which DSP is being applied.

Abstract: A communication system, a communication device and related method, the communication device comprising a processor, a source interface comprising a first source interface for a first external source, and an output interface, wherein the processor is configured to obtain, e.g. receive, a first input signal via the first source interface, obtain a first complementing signal, combine the first input signal and the first complementing signal to a first combined signal, apply a first filter function to the first combined signal for provision of a first left output signal and a first right output signal, the first filter function comprising one or more head related transfer functions, and output a left output signal and a right output signal via the output interface, wherein the left output signal is based on the first left output signal and the right output signal is based on the first right output signal.

Abstract: A communication system, a communication device and related method, the communication device comprising a processor, a source interface comprising a first source interface for a first external source, and an output interface, wherein the processor is configured to obtain, e.g. receive, a first input signal via the first source interface, obtain a first complementing signal, combine the first input signal and the first complementing signal to a first combined signal, apply a first filter function to the first combined signal for provision of a first left output signal and a first right output signal, the first filter function comprising one or more head related transfer functions, and output a left output signal and a right output signal via the output interface, wherein the left output signal is based on the first left output signal and the right output signal is based on the first right output signal.

Abstract: The present invention relates to an on-line method and a flow system as well as a cuvette for carrying out IR spectrometry for analysis of liquid food products, possibly containing dissolved gases, in a process line in a liquid food product processing plant, especially a dairy processing milk and milk products. A liquid sample is extracted from the process line to a measuring branch, the sample is thermostated and passed to a measurement cuvette. The IR-absorbance spectrum is measured, e.g. in the MID-IR or NIR-range. In order to obtain an on-line monitoring of the process line the liquid food sample is extracted directly from the process line into the measurement branch, in which the pressure is maintaining at least as high as in the adjacent process line. The high pressure ensures that dissolved air will stay dissolved in the liquid food. Before each new sample the measurement branch and cuvette are flushed by high flow rates with a part of the new sample to clean the cuvette.