5083results about "Transducer casings/cabinets/supports" patented technology

The apparatus for constructing a multi-channel output signal using an input signal and parametric side information, the input signal including the first input channel and the second input channel derived from an original multi-channel signal, and the parametric side information describing interrelations between channels of the multi-channel original signal uses base channels for synthesizing first and second output channels on one side of an assumed listener position, which are different from each other. The base channels are different from each other because of a coherence measure. Coherence between the base channels (for example the left and the left surround reconstructed channel) is reduced by calculating a base channel for one of those channels by a combination of the input channels, the combination being determined by the coherence measure. Thus, a high subjective quality of the reconstruction can be obtained because of an approximated original front/back coherence.

Sound processing methods and apparatus are provided. A sound capture unit is configured to identify one or more sound sources. The sound capture unit generates data capable of being analyzed to determine a listening zone at which to process sound to the substantial exclusion of sounds outside the listening zone. Sound captured and processed for the listening zone may be used for interactivity with the computer program. The listening zone may be adjusted based on the location of a sound source. One or more listening zones may be pre-calibrated. The apparatus may optionally include an image capture unit configured to capture one or more image frames. The listening zone may be adjusted based on the image. A video game unit may be controlled by generating inertial, optical and/or acoustic signals with a controller and tracking a position and/or orientation of the controller using the inertial, acoustic and/or optical signal.

Various dynamic audio ducking techniques are provided that may be applied where multiple audio streams, such as a primary audio stream and a secondary audio stream, are being played back simultaneously. For example, a secondary audio stream may include a voice announcement of one or more pieces of information pertaining to the primary audio stream, such as the name of the track or the name of the artist. In one embodiment, the primary audio data and the voice feedback data are initially analyzed to determine a loudness value. Based on their respective loudness values, the primary audio stream may be ducked during the period of simultaneous playback such that a relative loudness difference is generally maintained with respect to the loudness of the primary and secondary audio streams. Accordingly, the amount of ducking applied may be customized for each piece of audio data depending on its loudness characteristics.

Techniques for canceling echo and suppressing noise using an array microphone and signal processing. In one system, at least two microphones form an array microphone and provide at least two microphone input signals. Each input signal may be processed by an echo canceller unit to provide a corresponding intermediate signal having some echo removed. An echo cancellation control unit receives the intermediate signals and derives a first gain used for echo cancellation. A noise suppression control unit provides at least one control signal used for noise suppression based on background noise detected in the intermediate signals. An echo cancellation and noise suppression unit derives a second gain based on the control signal(s), cancels echo in a designated intermediate signal based on the first gain, and suppresses noise in this intermediate signal based on the second gain. The signal processing may be performed in the frequency domain.

Acoustic device including a member extending transversely of its thickness and capable of sustaining bending waves at least over an intendedly consequentially acoustically active area of the transverse extent of said member, the member having, by reason of orderly design methodology disclosed and claimed, a distribution of resonant modes of its natural bending wave vibration at least over said area that is dependent on values of particular parameters of said members, including geometrical configuration and directional bending stiffness(es), which values have been selected to predetermine said distribution of natural resonant modes being consonant with required achievable acoustic action of said member for operation of said device over a desired operative acoustic frequency range.

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.

At least one exemplary embodiment is directed to an earpiece comprising: an ambient microphone; an ear canal microphone; an ear canal receiver; a sealing section; a logic circuit; a communication module; a memory storage unit,; and a user interaction element, where the user interaction element is configured to send a play command to the logic circuit when activated by a user where the logic circuit reads registration parameters stored on the memory storage unit and sends audio content to the ear canal receiver according to the registration parameters.

A prefilter (5) for an audio system comprising a loudspeaker (1) in a room (2), which corrects both amplitude and phase errors due to the loudspeaker (1) by a linear phase correction filter response and corrects the amplitude response of the room (2) whilst introducing the minimum possible amount of extra phase distortion by employing a minimum phase correction filter stage. A test signal generator (8) generates a signal comprising a periodic frequency sweep with a greater phase repetition period than the frequency repetition period. A microphone (7) positioned at various points in the room (2) measures the audio signal processed by the room (2) and loudspeaker (1), and a coefficient calculator (6) (e.g. a digital signal processor device) derives the signal response of the room and thereby a requisite minimum phase correction to be cascaded with the linear phase correction already calculated for the loudspeaker (1). Filter (5) may comprise the same digital signal processor as the coefficient calculator (6). Applications in high fidelity audio reproduction, and in car stereo reproduction.

A method and apparatus for intelligently routing and managing audio signals within an electronic device is disclosed. The routing is responsive to a set of logical and physical policies which are stored in data tables which can be updated as needed.

An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier for modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

One embodiment of the present invention provides a system that manages the acoustic noise produced by a device. During operation, the system receives a set of acoustic characteristics for the device. The system then uses these acoustic characteristics to estimate the acoustic noise being generated by the device. Next, the system uses the estimated acoustic noise to adjust a setting in the device to manage the acoustic noise produced by the device.

Circuits and methods for providing zero-gap playback of consecutive data streams in portable electronic devices, such as media players, are described. In some embodiments, a circuit includes a decoder circuit configured to receive encoded audio data and to output decoded audio data including data streams associated with a data file and a subsequent data file. Moreover, a predictive circuit, which is electrically coupled to the decoder circuit, is configured to selectively generate additional samples based on samples in the data file, where the additional samples correspond to times after the end of a data stream associated with the data file. Additionally, a filter circuit, which is electrically coupled to the decoder circuit and selectively electrically coupled to the predictive circuit, is configured to selectively combine or blend samples at a beginning of the subsequent data file with the additional samples. Note that the circuit may be included in an integrated circuit.

A technique for enhancing audio signals generated from compressed digital audio files is described. The technique uses a Bass Maximizer module, a Harmonic Exciter module and a Quasi Stereo module. The Bass Exciter module enhances the intensity, depth and punch of the bass audio content by creating harmonic sequences from low frequency components contained in the original input signal. The Harmonic Exciter module adds to the treble audio content of the original input signal by generating harmonic series from the high frequency components contained in the input signal. The Quasi Stereo Module creates a stereo image of the enhanced input signal by adding and subtracting delayed and filtered versions of the enhanced input signal with itself to create left and right channeled stereo-like outputs. The technique provides a useful tool to regenerate from an audio signal more pleasant and joyful sounds.

One embodiment of the present invention provides a system that uses a sound sensor to adjust the audio output of a device. During operation, the system uses the sound sensor to determine an ambient sound level for the environment in the proximity of the device. The system then adjusts a volume setting for the device adaptively based on the determined ambient sound level. Adaptively adjusting the volume setting allows the device to adapt to its audio environment and ameliorates potentially-disruptive audio outputs.

Systems and methods for providing single microphone noise suppression fallback are provided. In exemplary embodiments, primary and secondary acoustic signals are received. A single microphone noise estimate may be generated based on the primary acoustic signal, while a dual microphone noise estimate may be generated based on the primary and secondary acoustic signals. A combined noise estimate based on the single and dual microphone noise estimates is then determined. Using the combined noise estimate, a gain mask may be generated and applied to the primary acoustic signal to generate a noise suppressed signal. Subsequently, the noise suppressed signal may be output.

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.

At least one exemplary embodiment is directed to a method of situation awareness facilitation for a vehicle driver comprising: receiving a first acoustic signal from outside a vehicle that the driver is in and converting the first acoustic signal into a first acoustic electronic signal; sending the first acoustic electronic signal to a processor; matching the first acoustic electronic signal by the processor to stored reference electronic signals; and sending an acoustic message associated with a matched reference electronic signal to a speaker in the cab of the vehicle.

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.

A sound control system and method applicable to an electronic device having a timing unit. A setting module is provided to set predetermined conditions and corresponding sound volume parameters. A time session control module retrieves the sound volume parameter corresponding to a present condition obtained by the timing unit. Then, a corresponding sound output signal to be outputted by a speaker unit connected to the sound control system is set via a sound effects module according to the sound volume parameter and a sound signal around the electronic device that is received and recognized by a sound recognition module. By such arrangement, the sound control system and method allow the electronic device to provide the user with optimal sound effects depending on the environment and the user's preferences.

A noise suppression device receives data representative of a noise-corrupted signal which contains a speech signal and a noise signal, divides the received data into data frames, and then passes the data frames through a pre-filter to remove a dc-component and the minimum phase aspect of the noise-corrupted signal. The noise suppression device appends adjacent data frames to eliminate boundary discontinuities, and applies fast Fourier transform to the appended data frames. A voice activity detector of the noise suppression device determines if the noise-corrupted signal contains the speech signal based on components in the time domain and the frequency domain. A smoothed Wiener filter of the noise suppression device filters the data frames in the frequency domain using different sizes of a window based on the existence of the speech signal. Filter coefficients used for Wiener filter are smoothed before filtering. The noise suppression device modifies magnitude of the time domain data based on the voicing information outputted from the voice activity detector.

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 and system for selectively and variably attenuating audio data are disclosed. A high-volume control value as selected by a user is first received, and this high-volume control value defines a volume output level for high amplitude audio samples. An attenuation factor is then determined by utilizing the high-volume control value. Each sample from an incoming audio data stream is conditionally attenuated with the attenuation factor such that high amplitude audio data get compressed while low amplitude audio data remain unaffected. Finally, the attenuated samples are sent to an output.

Active noise control system and method for controlling an acoustic noise generated by a noise source at a listening location, in which system and method sound is picked up in the surroundings of the listening location by a sound sensor; an electrical noise signal which corresponds to the acoustic noise of the noise source is generated and filtered adaptively in accordance with control signals. The adaptively filtered noise signal is irradiated into the surroundings of the listening location by a sound reproduction device, where a secondary path transfer function extends between the sound reproduction device and sound sensor. The noise signal is filtered with a transfer function that models the secondary path transfer function. The signals which provided by the sound sensor after first filtering serve as control signals for the adaptive filtering.

Techniques are provided to suppress noise and interference using an array microphone and a combination of time-domain and frequency-domain signal processing. In one design, a noise suppression system includes an array microphone, at least one voice activity detector (VAD), a reference generator, a beam-former, and a multi-channel noise suppressor. The array microphone includes multiple microphones—at least one omni-directional microphone and at least one uni-directional microphone. Each microphone provides a respective received signal. The VAD provides at least one voice detection signal used to control the operation of the reference generator, beam-former, and noise suppressor. The reference generator provides a reference signal based on a first set of received signals and having desired voice signal suppressed. The beam-former provides a beam-formed signal based on a second set of received signals and having noise and interference suppressed. The noise suppressor further suppresses noise and interference in the beam-formed signal.

Techniques for personalizing audio levels, in accordance with embodiments of the present technology, provide different audio volumes to different locations in a room allowing for two or more users to enjoy the same audio content at different volumes. Differential level and delay compensation filtering based on a position of each of a plurality of speakers, the location of each user and the preferred relative audio volume of each user are utilized to produce different effective audio levels in localized regions of a room.

A multi-dimensional audio processor receives as an input either a single channel signal or a two channel signal from an audio signal source; for example a musical instrument or an audio mixer. The processor is programmable to divide the input among at least 3 output channels in a user-defined manner. The processor is also user programmable to provide a variety of effect and mixing functions for the output channel signals.

Control system for devices such as an audio reproduction system, an actuator device, an electromechanical device and a telephony device. The system includes control circuitry which receives an input signal and a signal indicative of a position of a portion of the controlled apparatus. The control circuit provides an output signal to the controlled apparatus to affect an operation of the controlled apparatus. The output signal provides control of the apparatus to compensate for one or more of: motor factor; spring factor; back electromotive force; and impedance of a coil in a driver of the controlled apparatus. The signal indicative of position is derived by one or more position indicator techniques such as an infrared LED and PIN diode combination, position dependent capacitance of one portion of the controlled apparatus with respect to another portion of the controlled apparatus, and impedance of a coil in the controlled apparatus. The control circuitry is configurable to control transconductance and/or transduction of the system being controlled. A technique is disclosed to detect and measure a cant of a voice coil transducer, the technique including measuring a capacitance between one portion of the voice coil transducer with respect to another portion of the voice coil transducer over a range of movement of the voice coil during operation.

In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.

A hearing-aid device and a method for transmitting sound through bone conduction are disclosed. The hearing-aid device comprises a piezoelectric-type actuator, housing and connector. The piezoelectric actuator is preferably a circular flextensional-type actuator mounted along its peripheral edge in a specifically designed circular structure of the housing. During operation, the bone-conduction transducer is placed against the mastoid area behind the ear of the patient. When the device is energized with an alternating electrical voltage, it flexes back and forth like a circular membrane sustained along its periphery and thus, vibrates as a consequence of the inverse piezoelectric effect. Due to the specific and unique designs proposed, these vibrations are directly transferred trough the human skin to the bone structure (the skull) and provide a means for the sound to be transmitted for patients with hearing malfunctions. The housing acts as a holder for the actuators, as a pre-stress application platform, and as a mass which tailors the frequency spectrum of the device. The apparatus exhibits a performance with a very flat response in the frequency spectrum 200 Hz to 10 kHz, which is a greater spectrum range than any other prior art devices disclosed for bone-conduction transduction which are typically limited to less than 4 kHz.