1300results about "Tone control" patented technology

Systems and methods for adjusting playback and graphical representations of media files are provided. The systems and methods can monitor playback and access of media files to create usage metadata. The usage metadata can then be used to adjust the playback of the media file. For example, the usage metadata may indicate that a user skips, on average, the first 22 seconds of a particular song so the next time that song is played, the first 22 seconds will automatically be skipped. The usage metadata can additionally or alternatively be used to adjust a graphical representation of the media file. For example, the usage metadata may indicate that a user rarely accesses a particular song so the graphical representation of that song will be small and faded. This change in graphical representation can help a user find more commonly used media files.

A speaker includes at least one transducer and at least one memory device. The at least one transducer is adapted to receive an audio signal. The at least one memory device is adapted to store data related to the speaker. A method includes reading data from a memory device of at least one speaker. An audio signal is provided from an audio system.

A method and a control circuit for controlling an output of an audio signal of a battery-powered device are described.

A system and method are provided for smoothing the in-room acoustic magnitude response of an audio reproduction system. An in-room acoustic magnitude response analysis is performed to determine a room resonance induced peak associated with an audio signal. A replica of the audio signal is filtered at the room resonance induced peak. The filtered replica signal is added with the audio signal. Through this, smoothing of the room resonance induced peak may be achieved, such that a subjective impression of transient response and dynamics of the audio signal are preserved.

The invention includes a method, apparatus, and computer program to selectively suppress wind noise while preserving narrow-band signals in acoustic data. Sound from one or several microphones is digitized into binary data. A time-frequency transform is applied to the data to produce a series of spectra. The spectra are analyzed to detect the presence of wind noise and narrow band signals. Wind noise is selectively suppressed while preserving the narrow band signals. The narrow band signal is interpolated through the times and frequencies when it is masked by the wind noise. A time series is then synthesized from the signal spectral estimate that can be listened to. This invention overcomes prior art limitations that require more than one microphone and an independent measurement of wind speed. Its application results in good-quality speech from data severely degraded by wind noise.

Improved approaches to flexibly implementing graphic equalizers on media players are disclosed. These approaches provide dynamic order reduction of a multi-band graphic equalizer so that equalizer effects can be timely performed with only limited computational resources. In one embodiment, a media player receives a media item and associated equalizer settings for a multi-band graphic equalizer. The media player can then automatically (i.e., without user action) approximate the multi-band graphic equalizer with the equalizer settings for the media item using a fewer number of filters. Fewer filters means order reduction, and thus reduction in computational requirements. After the multi-band graphic equalizer is approximated, the media player can present the media item to its user in accordance with the reduced complexity, approximated equalizer.

A method and an apparatus for reproducing a sound signal are provided. The method includes generating an output sound signal to be transmitted to speakers by transmitting a first input sound signal through a filter; acquiring magnitude information of the output sound signal; determining frequency response parameters related to frequency responses of the filter based on the magnitude information; and adaptively adjusting coefficients of the filter based on the determined frequency response parameters.

A method of adjusting volume removes an audio signal of a first frequency band from audio signals, and increases the volume of the audio signal from which the signal of the first frequency band is removed.

To improve the consumer experience with electronic devices, an improved system and method is provided for selecting and applying optimal settings for listening to digital music. In exemplary embodiments, optimal equalizer settings for a given song or music category may be stored in the metadata of the music file. When the song is played by an electronic device, the optimal equalizer settings are extracted from the music metadata. The equalizer of the electronic device then may adjust automatically to the optimal settings as extracted from the metadata. The optimal settings may be defined by a content provider or selected by a user as a series of personal preferences. By adjusting the equalizer settings automatically to the optimal settings, enjoyment of the music may be enhanced.

The present invention describes techniques that can be used to provide novel methods of spatial audio rendering using adapted M-S matrix shuffler topologies. Such techniques include headphone and loudspeaker-based binaural signal simulation and rendering, stereo expansion, multichannel upmix and pseudo multichannel surround rendering.

The envelope of a decorrelated signal derived from an original signal can be shaped without introducing additional distortion, when a spectral flattener is used to spectrally flatten the spectrum of the decorrelated signal and the original signal prior to using the flattened spectra for deriving a gain factor describing the energy distribution between the flattened spectra, and when the so derived gain factor is used by an envelope shaper to timely shape the envelope of the decorrelated signal.

Loudspeakers can be damaged by high drive signals. One reason for this damage is an excess vibration displacement of the coil-diaphragm assembly. This invention describes a novel method for limiting this displacement by a signal processor. In the present invention, a low frequency shelving and notch filter is used to attenuate low frequencies according to a prediction of the loudspeaker displacement. A novel method for calculating coefficient values for a digital implementation of the low frequency shelving and notch filter according to the predicted displacement is described.

A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, filters, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions. The system also provides a methodology for selecting loudspeaker locations, the number of loudspeakers, the types of loudspeakers, correction factors, listening positions, crossover filters or a combination of these schemes in an audio system that has a single listening position or multiple listening positions.

Multiple passes are executed in the setup of an equalizer, and modification of the equalization is performed after each pass of an analysis phase. After an initial pass, the equalization is adjusted, based upon the location of peaks and valleys in the system response. This initial adjustment of equalization may tend to flatten most of the peaks and valleys to produce the desired uniform linear response. Inexact application of equalization corrections may introduce other artifacts into the system response and / or may not sufficiently normalize equalization. A second pass is then performed to measure the system response using the new equalization settings. The new peaks and valleys are measured, and the equalization adjusted to try to flatten response further. A proximity range may be applied to each pass, to reduce the likelihood that adjustment of one equalizer coefficient will create artifacts in the resulting system response.

A dynamic equalization system 12 for use in audio reproduction systems. The apparatus includes a chirp tone generator 38 which produces a tone having multiple frequencies. The chirp tone is broadcast into the listening space 10 from a transducer 14. The broadcast chirp tone is monitored by a second transducer 22 at the listening position to produce a received chirp tone. The received tone and the original tone are compared in a coefficient computer 44 connected to a programmable equalizer 42. The equalizer 42 uses the signal from the coefficient computer to compensate for irregularities 47 in listening space 10 and transducer 14 to produce a substantially undistorted listening experience from source 28 in listening space 10. The first step of the method of the invention is generation of a chirp tone. The chirp tone includes multiple frequencies. The chirp tone is broadcast into the listening space from a transducer placed at the selected transducer position. The broadcast chirp tone is next monitored by a transducer at the listening position to produce a received chirp tone. The received tone is then compared to the generated chirp tone and differences noted. The differences are used to program an equalizer for correction of sound. The process is done for each position where a transducer is located. Finally, sound from a program source is routed through the equalizer to the transducers for a corrected sound.

Automated creation of new music by listening is disclosed. A method to create new music may comprise listening to a plurality of music, learning from the plurality of music, and performing concatenative synthesis based on the listening and the learning to create the new music. The method may be performed on a computing device having an audio interface, such as a personal computer.

A method and apparatus to reproduce a media file including reading mapping information from a predetermined position of the media file, searching for a sound field effect file of the media file by referring to the mapping information, and setting an equalizer of the audio data on the basis of the sound field effect file. A user can reproduce a media file by using an optimized equalizer of each media file even though the user himself does not set the equalizer. Also, since an equalizer setting of the reproducing apparatus can be applied to another reproducing apparatus, a user can enjoy listening to a media file in the other reproducing apparatus under the same reproducing conditions as those of the reproducing apparatus.

An automatic sound field correcting device executes a signal process to the plurality of audio signals on respective correspondent signal transmission paths, and outputs them to a plurality of correspondent speakers to correct sound characteristics on the respective signal transmission paths. Namely, a measurement signal is supplied to each signal transmission path, and a measurement sound corresponding to it is outputted from the speaker to a sound space. The outputted measurement sound is detected as a detecting signal. The frequency characteristic of the audio signal on each signal transmission path is corrected by an equalizer, and a gain value of the equalizer is determined by a correction amount determining unit. A frequency characteristics correction is performed predetermined times. At a first correction, the correction amount determining unit determines the correction amount by performing a frequency analysis, based on the detecting signal, i.e. base on the detecting signal corresponding to the measurement sound actually outputted to the sound space. On the contrary, at and after a second correction, the correction amount determining unit determines the correction amount based on the detecting signal or an output signal of the equalizer. Namely, at and after the second correction, the output signal of the equalizer is supplied to the correction amount determining unit in a signal processing circuit as the need arises, and the frequency characteristics correction is performed without actually outputting the measurement sound to the sound space.

A method and apparatus for de-noising weak bio-signals having a relatively low signal to noise ratio utilizes an iterative process of wavelet de-noising a data set comprised of a new set of frames of wavelet coefficients partially generated through a cyclic shift algorithm. The method preferably operates on a data set having 2N frames, and the iteration is performed N−1 times. The resultant wavelet coefficients are then linearly averaged and an inverse discrete wavelet transform is performed to arrive at the de-noised original signal. The method is preferably carried out in a digital processor.

An apparatus for monitoring speaker cone displacement in an audio speaker includes: (a) an electromagnetic coil structure; (b) a ferrous core structure; the ferrous core structure and the electromagnetic coil structure being mounted with the speaker to effect variable electromagnetic coupling between the ferrous core structure and the electromagnetic coil structure as the speaker cone moves; (c) a signal injecting circuit coupled with the electromagnetic coil structure for injecting a predetermined input signal into the electromagnetic coil structure; and (d) a signal monitoring circuit coupled with the electromagnetic coil structure; the signal monitoring circuit receiving an output signal from the electromagnetic coil structure and generating an indicating signal based upon the output signal; at least one signal characteristic of the indicating signal being related with the cone displacement.

An automated audio tuning system may optimize an audio system for power efficiency when performing automated tuning of the audio system to optimize acoustic performance. The system may establish any number of different power efficiency weighting factors to provide a balance between acoustic performance and power efficiency during operation. The power efficiency weighting factors may range from representing optimizing power efficiency with constrained optimization of acoustic performance to optimized acoustic performance with minimized regard for power efficiency. For each of the efficiency weighting factors, the system may generate operational parameters, such as filter parameters, to achieve a target acoustic response while maintaining a determined level of power efficiency.