Abstract: The present invention relates in one aspect to a voice coil temperature protector for electrodynamic loudspeakers. The voice coil temperature protector comprises an audio signal input for receipt of an audio signal supplied by an audio signal source and a probe signal source for generation of a low-frequency probe signal. A signal combiner is configured to combine the audio signal with the low-frequency probe signal to provide a composite loudspeaker drive signal comprising an audio signal component and a probe signal component. The voice coil temperature protector comprises a current detector configured for detecting a level of a probe current component flowing through the voice coil in response to the composite loudspeaker drive signal and a current comparator which is configured to comparing the detected level of the probe current component with a predetermined probe current threshold.

Abstract: Methods and apparatus are disclosed for subwoofer calibration. An example method includes determining a crossover frequency of a playback system, and the playback system includes the subwoofer and a playback device. The example method includes the subwoofer and the playback device configured to output a multimedia content in synchronization. The example method also includes outputting from the subwoofer and the playback device a series of tones near the crossover frequency. The example method also includes prompting a user to select a tone set from the series of tones, and includes automatically adjusting a phase of the subwoofer in relation to the playback device based on the user selection.

Abstract: An electronic device for buzz reduction is described. The electronic device is to be used with a speaker driver that is built onto an enclosure and is to be driven by an audio signal which could cause the enclosure to produce buzz. The electronic device includes a filter that is to attenuate a frequency component of the audio signal before driving the speaker driver. The electronic device also includes a controller that is to configure the filter to attenuate the frequency component of the audio signal in response to determining that strength of the audio signal at the frequency component exceeds a threshold. Other embodiments are also described and claimed.

Abstract: An apparatus is disclosed for inputting digital data on the output channel(s) of an audio subsystem in an audio device, without interfering with normal operation of the audio subsystem. The described circuit includes a resistive element in parallel with the expected load device, such as a headphone or speaker. The resistive element receives a modulated digital signal from a data source or a switch, and the instantaneous current through the resistive element due to the modulated digital signal is reflected in a current feedback mechanism of the audio subsystem. Demodulation logic retrieves the digital signal from the current measured by the current feedback mechanism. A capacitor is provided to prevent the current in the resistive element from the digital signal from impacting the average DC current that the feedback mechanism uses to evaluate the load device.

Abstract: The present invention is directed towards a method for collecting information pertaining to an audio notification system comprising a plurality of audio devices and a database, wherein each of the plurality of audio devices comprises a loudspeaker and a microphone, the method comprising: sequentially for each of the plurality of audio devices: emitting, by means of the loudspeaker of a currently emitting audio device, an audible test sound; monitoring, by means of microphones of the audio devices in the system, for the audible test sound; and for each monitoring audio device, if the audible test sound is detected by the monitoring audio device, storing, in the database, information pertaining to the currently emitting audio device and information pertaining to the monitoring audio device detecting the audible test sound.

Abstract: A loudspeaker control system is disclosed. The loudspeaker control system includes a loudspeaker, a sensor for measuring a voltage and current and a processor. The processor is adapted to calculate an input-voltage-to-excursion transfer function over time from an admittance function, blocked electrical impedance and force factor, use the input-voltage-to-excursion transfer function over time to predict an excursion and use the excursion to control audio processing of the loudspeaker.

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include creating at least two iterative filters based on at least two separate frequency ranges of the original frequency response, calculating an error difference between the frequency response modified by the at least two iterative filters and the original frequency response, and applying the error difference to the audio signal.

Abstract: A method and device for detecting a force factor of a loudspeaker is provided. The method includes the steps of: providing the loudspeaker with a dynamic driving voltage signal; continuously measuring a current signal flowing through the loudspeaker; observing the current signal and if the current signal shows that the diaphragm excursion of the loudspeaker is greater than an displacement limit, decreasing the driving voltage signal until the current signal shows that the diaphragm excursion of the loudspeaker is less than or equal to the displacement limit; and substituting the current driving voltage signal, the current signal, the displacement limit, and an electrical impedance of the loudspeaker into a function so as to compute the force factor of the loudspeaker.

Abstract: A power management system coupled to a speaker module includes a monitor module for measuring a current signal and a voltage signal of the speaker module to obtain a real-time impedance information of the speaker module; a reception module for receiving a time-domain audio analogy signal to be transformed into a frequency-domain audio digital signal; a prediction module for generating a power prediction information according to an initial audio information, the real-time impedance information and the audio digital signal; a control module for generating a control signal according to the audio digital signal and a human hearing model information; and a power adjustment module for outputting an adjustment audio signal to the speaker module according to the power prediction information, the audio digital signal and the control signal, so as to perform a broadcast operation of the speaker module.

Abstract: An integrated circuit for processing audio and voice signals in a telephone is disclosed. The integrated circuit comprises an audio output port having a first and a second output terminal for connection to a first and a second input terminal, respectively, of a speaker. The integrated circuit further comprises an audio input port having a first and a second input terminal for connection to a first and a second output terminal of an audio input source. The integrated circuit also comprises at least one audio output path and one audio input path. Moreover, the integrated circuit comprises a test impedance, such as a resistor, (RT) for measuring an electrical impedance of the speaker.

Abstract: An apparatus for measuring a plurality of loudspeakers arranged at different positions includes a generator of a test signal for a loudspeaker; a microphone device configured for receiving a plurality of different sound signals in response to one or more loudspeaker signals emitted by one of the loudspeakers in response to the test signal; a controller for controlling emissions of the loudspeaker signals by the loudspeakers and for handling the different sound signals so that a set of sound signals recorded by the microphone device is associated with each loudspeaker in response to the test signal; and an evaluator for evaluating the set of sound signals for each loudspeaker to determine at least one loudspeaker characteristic for each loudspeaker and for indicating a loudspeaker state using the at least one loudspeaker characteristic. This scheme allows automatic, efficient and accurate measurement of loudspeakers arranged in a three-dimensional configuration.

Abstract: Techniques for improved audio localization for visual effects are described. In one embodiment, for example, an apparatus may comprise a processor circuit and an audio management module, and the audio management module may be operable by the processor circuit to determine a position of a user interface element in a presentation area, determine an audio effect corresponding to the user interface element, determine audio location information for the audio effect based on the position of the user interface element, the audio location information defining an apparent position for the audio effect, and generate audio playback information for the audio effect based on the audio location information. Other embodiments are described and claimed.

Abstract: A directional sound apparatus is described. The apparatus comprises a receiver operable to receive audio data associated with a video stream and to receive, over a network, ancillary audio data containing supplemental audio data associated with the video stream; a storage device operable to store a plurality of profiles, each profile being associated with a different user of the sound system and defining the audio to be provided to the user; a user location device defining a first positional location of a first user having a stored profile; an audio output operable to connect to a plurality of speakers; and an audio controller operable to control the audio output to the plurality of speakers to simultaneously provide the audio data and the supplemental audio data to the first user at the first positional location and said audio data to a second user located at a second, different, positional location.

Abstract: A method for crest-factor based gain modification comprises sampling audio inputs to determine an energy for each input. The inputs are amplified to produce output channels. An average energy is determined for each frame by adding a percentage of a last average energy to a residual percentage of the energy. The last average energy is equal to zero for a first sampling of the audio inputs. The residual percentage is equal to one hundred minus the percentage of the last average energy. The amplifier gain is iteratively reduced by a decrease-step for a fixed period until a number of exceeding-inputs is less than or equal to a maximum. The exceeding-inputs have an average energy that exceeds a maximum-energy. The amplifier gain is iteratively increased by an increase-step for another fixed period until a number of exceeding-inputs is greater than a maximum.

Abstract: A method of virtually tuning an audio system that incorporates an acoustic compensation system, where the audio system is adapted to play audio signals in a listening environment over one or more sound transducers. The acoustic compensation system has an audio sensor located at a sensor location in the listening environment. The transfer functions from each sound transducer to the audio sensor location are inherent. The method contemplates recording noise at the sensor location, and creating virtual transfer functions from each sound transducer to the sensor location based on the inherent transfer functions from each sound transducer to the sensor location. Audio signals are processed through the virtual sound transducer to sensor location transfer functions. A virtual sensor signal is created by combining the audio signals processed through the virtual sound transducer to sensor location transfer functions with the noise recorded at the sensor location.

Abstract: A device includes an audio processor, a first speaker, and a second speaker. The audio processor is configured to receive an audio signal. The first speaker is operatively coupled to the audio processor and is configured to produce a first sound wave. The first sound wave is associated with the audio signal. The second speaker is operatively coupled to the audio processor and is configured to produce a second sound wave. The second sound wave is configured to at least partially cancel the first sound wave.

Abstract: In accordance with these and other embodiments of the present disclosure, systems and methods may include a controller configured to be coupled to an audio speaker, wherein the controller receives an audio input signal, and based on a displacement transfer function associated with the audio speaker, processes the audio input signal to generate an output audio signal communicated to the audio speaker, wherein the displacement transfer function correlates an amplitude and a frequency of the audio input signal to an expected displacement of the audio speaker in response to the amplitude and the frequency of the audio input signal.

Abstract: An audio communications device has a handset in which a touch sensing ear piece region is coupled to an acoustic leakage analyzer. The acoustic leakage analyzer is to analyze signals from the touch sensing ear piece region and on that basis adjust an audio processing parameter. The latter configures an audio processor which generates an audio receiver input signal for the device. Other embodiments are also described and claimed.

Abstract: An exemplary embodiment illustrates an equalization pre-processing method, adapted for characterizing a second sound receptor unit in a sound receiving system based on knowing the internal structure parameters of a first sound receptor unit.

Abstract: An audio appliance includes two external connections for connection to external loudspeakers, and includes means for establishing an electrical signal between the external terminals that is representative of an audio source in order to inject an excitation electric current into each external loudspeaker. Means are provided for incorporating a component at a “detection” frequency that is situated outside the audible spectrum in the electrical signal that is established between the external terminals, together with means for detecting an electric current flowing between the external terminals and having a frequency corresponding to the detection frequency. This is applicable in particular to systems comprising a mobile appliance and a fixed docking station incorporating a woofer and designed to receive the mobile appliance.

Abstract: Method and, apparatus for implementing the method, the method comprising determining control signal data for an array of loudspeakers, the control signal data being such as to control the loudspeakers to produce a desired sound field associated with an audio signal, the method comprises determining control signal data for different frequency components of the desired sound field in respect of respective different positions in a listening volume of the loudspeaker array, wherein determination of the control signal data comprises sampling the desired sound field at the surface of a control volume (V).

Abstract: The application discloses a system and method for playback of audio signals though a large variety of headphone devices. To optimize playback, the specific headphone device is identified so the proper amplifier gain and equalization settings can be applied. Such identification is determined by measuring the headphone impedance as a function of frequency, and comparing the impedance data with data of know devices in a database. Once a match is found, the appropriate audio gain and equalization settings can be applied.

Abstract: Methods and systems of automatically identifying left-right earpieces may provide for determining an orientation of a device, and determining an earpiece orientation of a headset relative to the orientation of the device. Additionally, an audio output of the device may be configured based on the earpiece orientation. In one example, the earpiece orientation indicates whether the earpiece is facing either left or right with respect to the device.

Abstract: The present invention relates to methods and systems for determining an equalization filter for one or more loudspeakers provided in an enclosure, such as a listening room or an automobile cabin. A method for determining an equalization filter for one or more loudspeakers provided in an enclosure comprises the steps of providing the one or more loudspeakers with an audio input signal whereby a sound field is generated in the enclosure; determining an acoustic contribution or room gain of the enclosure to the generated sound field; forming a speaker equalization filter as the square root of the ratio between a target acoustic power output and the actual acoustic power output from the loudspeaker driver(s), which actual acoustic power output is calculated as the square of the power averaged sound pressure divided by the room gain; and determining the equalization filter as the speaker equalization filter.

Abstract: A method for audio sample rate conversion may include receiving an audio signal at a first rate, the audio signal having a fundamental frequency, determining absolute derivatives of the audio signal at the first rate, and generating a weighted sum of the absolute derivatives to arrive at a combined absolute derivative. The combined absolute derivative may be analyzed to locate a local minimum of the combined absolute derivative, a location in the audio signal may be selected based on the local minimum of the combined absolute derivative and at least one audio sample in the audio signal may be altered at the selected location to develop an audio signal at a second rate.

Abstract: Provided is a sound reproduction device including: a first loudspeaker having directionality utilizing a parametric effect; a second loudspeaker having directionality broader than that of the first loudspeaker; an orientation adjustment unit configured to change an orientation of the first loudspeaker; an information obtaining device configured to obtain positional information of a listener; and a drive controller electrically connected to the first loudspeaker, the second loudspeaker, the orientation adjustment unit, and the information obtaining device, and configured to control the orientation of the first loudspeaker based on the positional information of the listener.

Abstract: A method comprising the steps of: determining one or more impedance values of a loudspeaker; and determining the polarity of a loudspeaker based on the one or more impedance values.

Abstract: An array loudspeaker includes a plurality of drivers arranged in an array configuration. A digital signal processor-based control system processes direct audio signal and indirect audio signal inputs for the loudspeaker to simultaneously produce direct sound in the form of a directed beam or wavefront, and indirect sound as a perceptually diffuse soundfield.

Abstract: Devices and methods are adapted to characterize a multi-channel loudspeaker configuration, to correct loudspeaker/room delay, gain and frequency response or to configure sub-band domain correction filters.

Abstract: Systems and methods are described for clamping a headset in a calibration system using a clamp system that includes a clamp, platform, and one or more spindles (e.g., cushion spindles) to minimize or eliminate issues associated with positioning of headsets. The clamp system comprises a mount having a receptacle. When a device is introduced to the mount the receptacle receives at least a portion of a device. The clamp system includes a clamp attached to the mount and having a first arm rotateably coupled to a second arm that controls the first arm between an open position and a closed position. A platform and at least one spindle are connected to the first arm. When the device is present in the receptacle and the first arm is in the closed position the spindle contacts the device and seats or secures the device in the receptacle.

Abstract: A method of estimating diaphragm excursion of an electrodynamic loudspeaker may be performed using audio signals. An audio output signal may be applied to a voice coil of the electrodynamic loudspeaker through an output amplifier to produce sound. A detected voice coil current and a determined voice coil voltage may be applied to a linear adaptive digital loudspeaker model that has a plurality of adaptive loudspeaker parameters. The parameter values of the adaptive loudspeaker parameters may be computed based on the linear adaptive digital loudspeaker model and applied to a non-linear state-space model of the electrodynamic loudspeaker. For the non-linear state-space model, a predetermined non-linear function may be applied to at least one of the plurality of received parameter values to compute at least one non-linearity compensated parameter value of the adaptive loudspeaker parameters, to determine an instantaneous excursion of the diaphragm.

Abstract: A circuit for driving a loudspeaker is disclosed. The circuit includes: a signal generator connected to the loudspeaker, generating a first signal comprising a first pulse with positive value and a second pulse with negative value; a detection unit connected to the loudspeaker, detecting a second signal produced by the loudspeaker in response to the first signal; and a processing unit connected to the signal generator and the detection unit, calculating the impedance of the loudspeaker according to the second signal; wherein each pulse width of the first and second pulses is between 100 ns and 900 ns.

Abstract: Aspects for monitoring audible tones indicative of operational status of each planar in a multiple planar chassis are described. Included in the aspects is the monitoring of a speaker channel of each planar of a plurality of planars in a common chassis for state changes of beep tones. An operational status of a specific planar emitting the beep tones is identified based on the state changes.

Abstract: In one embodiment, an apparatus for testing sound transducers includes a test socket having at least one acoustic generator and at least one sound monitoring device integrated therein. In one embodiment, the test socket includes a well for holding the sound transducer during test, the well being in communication with the at least one acoustic generator and the at least one sound receiving device.

Abstract: Embodiments of the invention are directed to systems, methods and computer program products for interpolating impedance data associated with an electronic device. The present invention enables faster electronic device impedance analysis which in turn will have an impact on memory allocation associated with a computing system that controls the electronic device. An exemplary method comprises receiving complex impedance data; converting the complex impedance data to polar impedance data, wherein the polar impedance data defines a Kennelly circle; normalizing the polar impedance data based on at least one parameter associated with the Kennelly circle; and interpolating the polar impedance data for a selected frequency.

Abstract: A system for headphone equalization includes a stored set of predetermined tone burst reference signals and a stored set of predetermined tone burst test signals that form a range of frequencies used in a user specific audio test to develop a headphone correction filter. A predetermined tone burst reference signal and a predetermined tone burst test signal may intermittently and sequentially drive a transducer included in the headphone. A loudness of the predetermined tone burst reference signal may be fixed and a loudness of the predetermined tone burst test signal may be variable with a gain setting. The gain setting may be used to generate the headphone correction filter.

Abstract: An acoustic apparatus including circuitry to correct for acoustic cross-coupling of acoustic drivers mounted in a common acoustic enclosure. A plurality of acoustic drivers are mounted in the acoustic enclosure so that motion of each of the acoustic drivers causes motion in each of the other acoustic drivers. A canceller cancels the motion of each of the acoustic drivers caused by motion of each of the other acoustic drivers. A cancellation adjuster cancels the motion of each of the acoustic drivers that may result from the operation of the canceller.

Abstract: Disclosed is subject matter that proposes a system and method for a media device to automatically detect the characteristics of an attached speaker. Speakers have many different characteristics, for example, power, impedance, frequency response, etc. With knowledge of the speaker characteristics, audio output can be equalized appropriately, and an amplifier of the media device, for example, can prevent exceeding the maximum power handling capability of the speaker. Described is a device and method for retrieving information about the speaker from a memory that is coupled to the speaker. A media device can read the data from the memory over existing speaker wires. Software and/or hardware in the media device can optimize the output to the attached speaker. Accordingly, the media device can interrogate the speaker directly over speaker wire to obtain the characteristics of the speaker.

Abstract: A wireless speaker audio system configured to receive audio information wirelessly transmitted by an audio source including first and second wireless transceivers. The first wireless transceiver establishes a bidirectional secondary wireless link with the audio source for receiving and acknowledging receipt of the audio information. The first and second wireless transceivers communicate with each other via a primary wireless link. A wireless audio system including an audio source and first and second wireless transceivers. The first and second wireless transceivers communicate via a primary wireless link. The audio source communicates audio information to the first wireless transceiver via a secondary wireless link which is configured according to a standard wireless protocol. The first wireless transceiver is configured to acknowledge successful reception of audio information via the secondary wireless link.

Abstract: A sound system 100 for producing sound on a loudspeaker 130 is provided wherein the temperature of a voice coil of the loudspeaker may be monitored continuously. The sound system comprises a signal generator 150 for generating an evaluation signal which is combined with an input sound signal 120 to obtain a loudspeaker signal and a monitor 140 for monitoring an electric response of the voice coil to the loudspeaker signal. The loudspeaker signal may be adapted 180 to control the temperature of the voice coil in dependency upon the monitored response. The system can determine the temperature of the voice coil 170 even if the input sound signal comprises only little sound energy.

Abstract: A signal processing device includes: a detection unit generating a digital detection signal corresponding to motion of a diaphragm of a speaker to output the digital detection signal; a gain adjustment unit generating a digital feedback signal by multiplying the outputted digital detection signal by again coefficient to output the generated digital feedback signal; a combining unit combining the outputted digital feedback signal with a digital audio signal; a storage unit storing plural gain coefficients; and a control unit performing control so that a given gain coefficient is selected from the plural gain coefficients and that the selected gain coefficient is used for the multiplication.

Abstract: A loudspeaker drive circuit comprises an input for receiving an audio signal and a signal processor for processing the audio signal before application to the loudspeaker. The signal processor processes the audio signal to derive a loudspeaker drive signal which results in the loudspeaker membrane reaching its maximum displacement in both directions of diaphragm displacement.

Abstract: Frequency equalization system substantially equalizes the room frequency responses generated by at least one loudspeaker within a listening area so that the frequency responses in the listening area are substantially constant and flat within a desired frequency range. The frequency equalization system uses multiple microphones to measure the impulse responses of the room and uses the impulse responses to design filters to process the audio signals of one or more subwoofers to achieve an improved bass response that is flat across the relevant frequency range. The system employs an algorithm that is a closed-form, non-iterative, mathematical solution and features very short computation time.

Abstract: An impedance matched resonant circuit uses inductors coupling an audio source and a speaker array in order to reduce feedback reflection induced by an audio signal traveling from the audio source to the array of speakers. An RC circuit could also be coupled to the positive and negative terminals of the speaker array to reduce the slope differential of the audio signal at certain frequencies. The circuit can be packaged together in a single module with switches to activate and deactivate portions of the circuit to alter the effectiveness of the circuit depending upon need.

Abstract: An in-ear audio device has a casing on which is disposed one or more bags are positioned to be filled with fillable material during a customization process in which a test sound is acoustically output into an ear canal by an acoustic driver of the in-ear audio device while the one or more bags are being filled, and a microphone acoustically coupled to the ear canal is employed to detect sounds within the ear canal that are indicative of the frequency response of the acoustic output of the acoustic driver to determine when the degree of sealing of the ear canal by the one or more bags is sufficient to achieve a desired quality of frequency response.

Abstract: A fiber-reinforced composite part comprises structural fiber strands and linear electromagnetic-to-acoustic transducers embedded in a polymeric matrix. When these internal transducers are activated in sequence, the propagating acoustic waves are detected by an array of external acoustic-to-electric transducers acoustically coupled to external surfaces of the part. These external transducers convert impinging acoustic waves into electrical signals that carry information concerning acoustic wave amplitudes and phase shifts relative to the excitation of the internal transducers. The electrical signals are processed by a computer which is programmed to determine the location and orientation of each internal transducer and ultimately the structural integrity of the composition.

Abstract: An audio power management system manages operation of audio devices in an audio system. The audio power management system includes a parameter computer, a threshold comparator and a limiter. Audio signals generated with the audio system may be provided to the audio power management system. Based on a measured actual parameter of the audio signal, such as a real-time actual voltage and/or a real-time actual current, the parameter computer can derive estimated operational characteristics of audio devices, such as a loudspeaker included in the audio system. The threshold comparator may use the estimated operational characteristics to develop a threshold and manage operation of one of more devices in the audio system by monitoring the measured actual parameter, and selectively directing the limiter to adjust the audio signal, or another device in the audio system to protect or optimize performance.

Abstract: In a method for operating a loudspeaker device having at least one loudspeaker, at least one actual membrane state parameter of a membrane of the loudspeaker is detected by a detecting device. An actual membrane state of the membrane based on the following actual membrane state parameters: actual membrane position (xactual), actual membrane speed (vactual) and actual membrane acceleration (aactual), is determined from the at least one detected actual membrane state parameter (xactual, aactual) and is directly used to determine a driving signal (U(t)) that is applied to the voice coil of the loudspeaker. The voice coil is operatively connected to the membrane. A loudspeaker device being operated with this method and a device for noise compensation are also disclosed.

Abstract: This accessory is apparatus capable of implementing a method that can affect one or more features of a portable electronic device when brought into a vehicle having a radio with a right and left speaker. The accessory has a sender for initiating transmission of an RF test signal at the portable electronic device The RF test signal is arranged to cause the radio to produce a distinct pair of audio responses from the right and left speaker. The accessory has a discriminator for determining for the right and left speaker the corresponding return delay between transmission of the RF test signal and arrival of the distinct pair of audio responses at the portable electronic device.

Abstract: An electronic device and methods are disclosed. A cover panel is located on a front surface of the electronic device. A piezoelectric vibration module configured to vibrate the cover panel. A drive module configured to vibrate the piezoelectric vibration module based on a sound signal. A pressure intensity acquiring module configured to acquire pressure intensity information. The pressure intensity information indicates an intensity at which an ear of a user is pressed onto the cover panel. A sound quality controller configured to control a sound quality of the sound signal based on the pressure intensity information.