Abstract: A scanning device for an air filter being arranged to be mounted in a circular duct, comprising: a driving unit and a sampling unit, wherein the driving unit comprises an inner magnetic member and an inner magnetic member support element and an outer magnetic member and a circular guiding rail wherein the sampling unit comprises a central rotation axis and a probe nozzle, the probe nozzle is driven to rotate around the central rotation axis by a force of attraction between the inner magnetic member and the outer magnetic member.

Abstract: A system and method can determine a time-varying position of a sound in a multi-channel audio signal. At least one processor can: receive a multi-channel audio signal representing a sound, each channel of the multi-channel audio signal providing audio associated with a corresponding channel position around a perimeter of a soundstage; determine a time-varying volume level for each channel of the multi-channel audio signal; determine, from the time-varying volume levels and the channel positions, a time-varying position in the soundstage of the sound; and generate a location data signal representing the time-varying position of the sound. The channel positions can be time-invariant. The position magnitude can be scaled to provide a unit magnitude as a sound pans from a channel to an adjacent channel. The position azimuth angle can be scaled to account for center location bias.

Abstract: Systems and methods to provide distortion sensing, prevention, and/or distortion-aware bass enhancement in audio systems can be implemented in a variety of applications. Sensing circuitry can generate statistics based on an input signal received for which an acoustic output is generated. In various embodiments, the statistics can be used such that a multi-notch filter can be used to provide input to a speaker to generate the acoustic output. In various embodiments, the statistics from the sensing circuitry can be provided to a bass parameter controller coupled to bass enhancement circuitry to operatively provide parameters to the bass enhancement circuitry. The bass enhancement circuitry can provide a bass enhanced signal for generation of the acoustic output, based on the parameters. Various combinations of a multi-notch filter and bass enhancement circuitry using statistics from sensing circuitry can be implemented to provide an enhanced acoustic output.

Abstract: Systems and methods to provide distortion sensing, prevention, and/or distortion-aware bass enhancement in audio systems can be implemented in a variety of applications. Sensing circuitry can generate statistics based on an input signal received for which an acoustic output is generated. In various embodiments, the statistics can be used such that a multi-notch filter can be used to provide input to a speaker to generate the acoustic output. In various embodiments, the statistics from the sensing circuitry can be provided to a bass parameter controller coupled to bass enhancement circuitry to operatively provide parameters to the bass enhancement circuitry. The bass enhancement circuitry can provide a bass enhanced signal for generation of the acoustic output, based on the parameters. Various combinations of a multi-notch filter and bass enhancement circuitry using statistics from sensing circuitry can be implemented to provide an enhanced acoustic output.

Abstract: A system and method can determine a time-varying position of a sound in a multi-channel audio signal. At least one processor can: receive a multi-channel audio signal representing a sound, each channel of the multi-channel audio signal providing audio associated with a corresponding channel position around a perimeter of a soundstage; determine a time-varying volume level for each channel of the multi-channel audio signal; determine, from the time-varying volume levels and the channel positions, a time-varying position in the soundstage of the sound; and generate a location data signal representing the time-varying position of the sound. The channel positions can be time-invariant. The position magnitude can be scaled to provide a unit magnitude as a sound pans from a channel to an adjacent channel. The position azimuth angle can be scaled to account for center location bias.

Abstract: Loudness control systems or methods may normalize audio signals to a predetermined loudness level. If the audio signal includes moderate background noise, then the background noise may also be normalized to the target loudness level. Noise signals may be detected using content-versus-noise classification, and a loudness control system or method may be adjusted based on the detection of noise. Noise signals may be detected by signal analysis in the frequency domain or in the time domain. Loudness control systems may also produce undesirable audio effects when content shifts from a high overall loudness level to a lower overall loudness level. Such loudness drops may be detected, and the loudness control system may be adjusted to minimize the undesirable effects during the transition between loudness levels.

Abstract: A system can be provided for increasing loudness of an audio signal to present a perceived loudness to a listener that is greater than a loudness provided natively by a loudspeaker. The system can include one or more of the following: a frequency suppressor, a loudness adjuster, an equalizer, and a distortion control module. The frequency suppressor can increase headroom in the audio signal by filtering out low and/or high frequencies. The loudness adjuster can calculate a loudness of the audio signal and apply a gain to the audio signal to increase the loudness. The equalizer can further increase headroom by attenuating portions of a passband of the loudspeaker's frequency response. The distortion control module can induce partial harmonic distortion in the audio signal to further increase loudness.

Abstract: Loudness control systems or methods may normalize audio signals to a predetermined loudness level. If the audio signal includes moderate background noise, then the background noise may also be normalized to the target loudness level. Noise signals may be detected using content-versus-noise classification, and a loudness control system or method may be adjusted based on the detection of noise. Noise signals may be detected by signal analysis in the frequency domain or in the time domain. Loudness control systems may also produce undesirable audio effects when content shifts from a high overall loudness level to a lower overall loudness level. Such loudness drops may be detected, and the loudness control system may be adjusted to minimize the undesirable effects during the transition between loudness levels.

Abstract: Loudness control systems or methods may normalize audio signals to a predetermined loudness level. If the audio signal includes moderate background noise, then the background noise may also be normalized to the target loudness level. Noise signals may be detected using content-versus-noise classification, and a loudness control system or method may be adjusted based on the detection of noise. Noise signals may be detected by signal analysis in the frequency domain or in the time domain. Loudness control systems may also produce undesirable audio effects when content shifts from a high overall loudness level to a lower overall loudness level. Such loudness drops may be detected, and the loudness control system may be adjusted to minimize the undesirable effects during the transition between loudness levels.

Abstract: There is disclosed methods and apparatus for decomposing a signal having a plurality of channels into direct and diffuse components. The correlation coefficient between each pair of signals from the plurality of signals may be estimated. A linear system of equations relating the estimated correlation coefficients and direct energy fractions of each of the plurality of channels may be constructed. The linear system may be solved to estimate the direct energy fractions. A direct component output signal and a diffuse component output signal may be generated based in part on the direct energy fractions.

Abstract: Loudness control systems or methods may normalize audio signals to a predetermined loudness level. If the audio signal includes moderate background noise, then the background noise may also be normalized to the target loudness level. Noise signals may be detected using content-versus-noise classification, and a loudness control system or method may be adjusted based on the detection of noise. Noise signals may be detected by signal analysis in the frequency domain or in the time domain. Loudness control systems may also produce undesirable audio effects when content shifts from a high overall loudness level to a lower overall loudness level. Such loudness drops may be detected, and the loudness control system may be adjusted to minimize the undesirable effects during the transition between loudness levels.

Abstract: There is disclosed methods and apparatus for decomposing a signal having a plurality of channels into direct and diffuse components. The correlation coefficient between each pair of signals from the plurality of signals may be estimated. A linear system of equations relating the estimated correlation coefficients and direct energy fractions of each of the plurality of channels may be constructed. The linear system may be solved to estimate the direct energy fractions. A direct component output signal and a diffuse component output signal may be generated based in part on the direct energy fractions.

Abstract: A system for generating parametric stereo data from phase modulated stereo data is provided. A phase difference system receives left channel data and right channel data and determines a phase difference between the left channel data and the right channel data. A phase difference weighting system receives the phase difference data and generates weighting data to adjust left channel amplitude data and right channel amplitude data based on the phase difference data. A magnitude modification system adjusts the left channel amplitude data and the right channel amplitude data using the weighting data to eliminate phase data in the left channel data and the right channel data.

Abstract: A system for identifying audio data is provided. The system includes a transform system receiving left channel audio data and right channel audio data and generating a plurality of frequency bins of left channel magnitude data, left channel phase data, right channel magnitude data and right channel phase data. A watermarking system receives watermarking data and modifies predetermined frequency bins of the left channel phase data and the right channel phase data to encode the watermarking data. A magnitude system receives the left channel magnitude data and the right channel magnitude data and increases the left channel magnitude data and the right channel magnitude data for one or more of the predetermined frequency bins to a threshold level if the left channel magnitude data and the right channel magnitude data for the corresponding frequency bin is less than the threshold level.

Abstract: An audio spatial environment engine for converting from an N channel audio system to an M channel audio system, where N is an integer greater than M, is provided. The audio spatial environment engine includes one or more correlators receiving two of the N channels of audio data and eliminating delays between the channels that are irrelevant to an average human listener. One or more Hilbert transform systems each perform a Hilbert transform on one or more of the correlated channels of audio data. One or more summers receive at least one of the correlated channels of audio data and at least one of the Hilbert transformed correlated channels of audio data and generate one of the M channels of audio data.

Abstract: An audio spatial environment engine for flexible and scalable up-mixing from an M channel audio system to an N channel audio system, where M and N are integers and N is greater than M, is provided. The input M channel audio is provided to an analysis filter bank which converts the time domain signals into frequency domain signals. Relevant inter-channel spatial cues are extracted from the frequency domain signals on a sub-band basis and are used as parameters to generate adaptive N channel filters which control the spatial placement of a frequency band element in the up-mixed sound field. The N channel filters are smoothed across both time and frequency to limit filter variability which could cause annoying fluctuation effects. The smoothed N channel filters are then applied to adaptive combinations of the frequency domain input signals and are provided to a synthesis filter bank which generates the N channel time domain output signals.

Abstract: A system for identifying audio data is provided. The system includes a transform system receiving left channel audio data and right channel audio data and generating a plurality of frequency bins of left channel magnitude data, left channel phase data, right channel magnitude data and right channel phase data. A watermarking system receives watermarking data and modifies predetermined frequency bins of the left channel phase data and the right channel phase data to encode the watermarking data. A magnitude system receives the left channel magnitude data and the right channel magnitude data and increases the left channel magnitude data and the right channel magnitude data for one or more of the predetermined frequency bins to a threshold level if the left channel magnitude data and the right channel magnitude data for the corresponding frequency bin is less than the threshold level.

Abstract: An audio spatial environment engine for converting from an N channel audio system to an M channel audio system, where N is an integer greater than M, is provided. The audio spatial environment engine includes one or more correlators receiving two or more of the N channels of audio data and eliminating delays between the channels that are irrelevant to an average human listener. One or more Hilbert transform systems each perform a Hilbert transform on one or more of the correlated channels of audio data. One or more summers receive at least one of the correlated channels of audio data and at least one of the Hilbert transformed correlated channels of audio data and generate one of the M channels of audio data.

Abstract: An audio spatial environment engine is provided for converting between different formats of audio data. The audio spatial environment engine (100) allows for flexible conversion between N-channel data and M-channel data and conversion from M-channel data back to N?-channel data, where N, M, and N? are integers and where N is not necessarily equal to N?. For example, such systems could be used for the transmission or storage of surround sound data across a network or infrastructure designed for stereo sound data. The audio spatial environment engine provides improved and flexible conversions between different spatial environments due to an advanced dynamic down-mixing unit (102) and a high-resolution frequency band up-mixing unit (104). The dynamic down-mixing unit includes an intelligent: analysis and correction loop (108, 110) capable of correcting for spectral, temporal, and spatial inaccuracies common to many down-mixing methods.

Abstract: An audio processing application is provided which utilizes an audio codec encode/decode simulation system and a psychoacoustic model to estimate audible quantization noise that may occur during lossy audio compression. Mask-to-noise ratio values are computed for a plurality of frequency bands and are used to intelligently process an audio signal specifically for a given audio codec. In one exemplary embodiment, the mask-to-noise ratio values are used to reduce the extent of perceived artifacts for lossy compression, such as by modifying the energy and/or coherence of frequency bands in which quantization noise is estimated to exceed the masking threshold.