Abstract: A method of generating a signal for driving a first linear array of sound sources. The first linear array of sound sources comprises a primary sound source and one or more secondary sound sources. The method comprises the steps of receiving an audio signal for a first channel of an audio system, deriving, from the audio signal, a first signal and a second signal, applying a low-pass filter to the second signal to generate a second drive signal for driving the one or more secondary sound sources, and applying a corresponding high-frequency shelving filter to the first signal to generate a first drive signal for driving the primary sound source. A computer program product and an audio system for generating a levelled sound field is also provided.

Abstract: An in-vehicle acoustic system includes a first amplifier that outputs a first audio signal to a first mid-range speaker; a second amplifier that outputs a second audio signal to a second mid-range speaker; a third amplifier that outputs a third audio signal to a first high-range speaker and a first low-range speaker; and a fourth amplifier that outputs a fourth audio signal to a second high-range speaker and a second low-range speaker, in which the third audio signal is inputted to the first high-range speaker, the fourth audio signal is inputted to the second high-range speaker, the first audio signal is inputted to a deep-bass speaker, the third audio signal is inputted to the first low-range speaker, the fourth audio signal is inputted to the second low-range speaker.

Abstract: A system may include a vibrating surface, a single mechanical transducer mechanically coupled to the vibrating surface, a signal processing subsystem configured to receive an audio signal and a haptic signal, process the audio signal and the haptic signal to generate a combined audio-haptic signal, and drive the combined audio-haptic signal to the single mechanical transducer in order to generate concurrent audio playback and haptic effects on the vibrating surface; and a control subsystem configured to, responsive to a haptic stimulus, modify at least one parameter of at least one of the audio signal and the haptic signal to accommodate the concurrent audio playback and haptic effects on the vibrating surface within at least one operational limit of the system.

Abstract: In some embodiments, a method for performing enhancement on an audio signal to generate an enhanced audio signal in response to feedback indicative of amount of compression applied to at least one frequency band of the enhanced audio signal. In typical embodiments, the enhancement is or includes bass enhancement. Examples of other types of enhancement performed in other embodiments include dialog enhancement, upmixing, frequency shifting, harmonic injection or transposition, subharmonic injection, virtualization, and equalization. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook, tablet, mobile phone, or other device with small speakers) configured to perform any embodiment of the method.

Abstract: A method for reducing gain for audio amplification by an audio system having a tweeter channel, an optional midrange channel, and a woofer channel. Gain of the woofer channel is reduced simultaneously with reducing gain of the tweeter channel, both responsive to detecting the same instance of overloading (overdriving) the woofer channel. Other aspects are also described and claimed.

Abstract: A method for estimating an embedding capacity includes receiving, at a deterministic reference encoder, a reference audio signal, and determining a reference embedding corresponding to the reference audio signal, the reference embedding having a corresponding embedding dimensionality. The method also includes measuring a first reconstruction loss as a function of the corresponding embedding dimensionality of the reference embedding and obtaining a variational embedding from a variational posterior. The variational embedding has a corresponding embedding dimensionality and a specified capacity. The method also includes measuring a second reconstruction loss as a function of the corresponding embedding dimensionality of the variational embedding and estimating a capacity of the reference embedding by comparing the first measured reconstruction loss for the reference embedding relative to the second measured reconstruction loss for the variational embedding having the specified capacity.

Abstract: An apparatus and method of excursion protection of a loudspeaker. The method includes attenuating selected bands in a transform domain, controlled by a feedback signal resulting from an excursion transfer function that has been modified according to the real-time operational characteristics of the loudspeaker. In this manner, the system reduces the amount of wideband attenuation needed to address the predicted excursion, resulting in a better listening experience.

Abstract: A technique for enabling a person to experience music haptically includes splitting an audio signal representing music into an original low-frequency band and at least one high-frequency band, and translating each high-frequency band into a corresponding synthesized low-frequency band. The original low-frequency-band signal and each synthesized low-frequency-band signal is used to excite a respective exciter, which can produce vibrations corresponding to the received signal. By placing the exciters in contact with or proximate to the body of a person, the person can experience the music, including not only the low-frequency components of the music but also, at least to some extent, the high-frequency components.

Abstract: An Auto Setup Program (“ASP”) may be implemented in the sound system of a vehicle to accomplish the goal of a dramatically improved audio quality achievable in any vehicle, regardless of the speaker components and head-unit utilized. The ASP performs automated time alignment measurements and equalizer processes using an external microphone. The ASP eliminates the need for expensive and time consuming professional tuning operations. The on-board ASP can be run multiple times as desired. For example, the ASP can be executed after changes to the system, such as upgrading speakers.

Abstract: An audio device includes a speaker array and a controller for beam-steering audio output by the speaker array to localize sound in different locations in a local area around the audio device. The audio device also includes a microphone array or a set of cameras configured to detect an object, such as a human, in the local area around the audio device. From data captured by the microphone array or the set of cameras, the audio device determines a location of the object in the local area and steers the audio output by the speaker array towards the determined location of the object. As the object moves within the local area, the audio device dynamically steers the audio output to move along with the object.

Abstract: A system manages temperature and excursion effects on a loudspeaker. Data corresponding to a temperature of the loudspeaker and data corresponding to an excursion (i.e., displacement) of a membrane of the loudspeaker are determined. The dynamic ranges of the temperature and excursion data are compressed, combined, and smoothed. This smoothed data is multiplied with a delayed version of audio data; peak values of the result may be limited. This operation may be performed on multiple frequency bands of audio data. The final output data is combined to form a full-band signal and sent to a loudspeaker.

Abstract: The present disclosure provides methods and systems for homogenizing a low frequency listening experience for users in a plurality of locations, such as a plurality of seat positions in a vehicle. An example audio system includes a plurality of woofers configured to output low frequency sound into a listening environment to yield a homogeneous low frequency sound field. Homogeneity of the low frequency sound field in this case is for example spectrally uniform throughout an extended listening space.

Abstract: A system is provided for limiting distortion of an audio speaker. The system includes a first lowpass filter circuitry that is configured to receive a system input signal and generate a first lowpass filtered output signal. The system also includes a first limiter circuitry that is configured to limit the first lowpass filtered output signal. Limiting the first low pass filtered output signal includes reducing an amplitude of the first lowpass filtered output signal below a first predetermined threshold value, thereby to generate a first limited output signal. A second lowpass filter circuitry is configured to receive the first limited output signal and to generate a second lowpass filtered output signal.

Abstract: Devices and techniques are generally described for automatic volume leveling of an audio signal. An audio signal may be received and a first portion of the audio signal may be stored in a buffer. A root mean square (RMS) estimate of a level of the first portion of the audio signal may be determined based at least in part on a second portion of the audio signal previous to the first portion. A modified audio signal may be generated by multiplying the first portion of the audio signal in the buffer by a variable gain. The variable gain may be based at least in part on the RMS estimate. The modified audio signal may be sent to a loudspeaker.

Abstract: The present disclosure relates to an integrated circuit output driver, e.g. operating in class-D, for driving an audio transducer. The integrated circuit output driver comprises a first half-bridge driver comprising a first PMOS transistor and a first NMOS transistor connected in series between positive and negative supply voltage rails. A first body terminal is connected to a body of the first PMOS transistor for receipt of a first back bias voltage and a second body terminal connected to a body of the first NMOS transistor for receipt of a second back bias voltage. The integrated circuit output driver comprises a bias voltage generator configured to adjust at least one of the first back bias voltage and the second back bias voltage to control on-resistance of the first PMOS transistor and/or the first NMOS transistor. The integrated circuit output driver is well-suited for hearing aids, headsets and other audio devices.

Abstract: The examples provided herein involve multiple groups in a playback system. An example method involves facilitating storage of a configuration of multiple zone groups as a multi-zone scene of a media playback system, the media playback system including at least a first zone group including at least two zone players and a second zone group, a user interface of a controller presenting at least an indication of (1) the first zone group, (2) the second zone group, and (3) a multi-zone group scene creation command.

Abstract: A spectral tilt of an audio signal is used to determine whether a speaker will introduce perceptible distortion during playback of the audio signal. The spectral tilt may be indicated by determining a ratio between energy in a distortion-producing frequency band and energy in a distortion-masking frequency band. Based on the determined spectral tilt, the distortion-producing frequency band may be attenuated to reduce the distortion introduced by the speaker. Additionally, the distortion-masking frequency band may be amplified to reduce perceptibility of the distortion produced by the speaker.

Abstract: The present disclosure provides systems and methods to dynamically extend loudspeaker capabilities. In particular, a system comprising a loudspeaker can receive an electronic audio signal. Responsive to a change in available headroom, one or more parameters of an equalizer being applied to at least a portion of the signal in order to extend a physical low-frequency response of the loudspeaker can be modified based on the change in available headroom. Additionally or alternatively, responsive to the change in available headroom, a bandwidth of low-frequency content of the signal that is being synthesized by the system to extend low-frequency capability of the system beyond physical capabilities of the system can be adjusted based on the change in available headroom.

Abstract: A circuit and a method for maintaining the desired tone, harmonic relationship and dynamic response of a musical instrument amplifier throughout the complete master volume power attenuation range.

Abstract: An image processing apparatus includes: an embed-data generating unit configured to generate embed-data, which is to be embedded in input-image-data, the input-image-data representing an image including a first color, the embed-data representing an image including a second color; an inverting unit configured to invert the image represented by the embed-data, in response to meeting a predetermined condition relating to the first color and the second color; and an output-image-data generating unit configured to embed, in the input-image-data, embed-data representing the image that is inverted by the inverting unit, in order to generate output-image-data in a case where the predetermined condition is met, and configured to embed, in the input-image-data, embed-data representing the image that is not inverted by the inverting unit, in order to generate output-image-data in a case where the predetermined condition is not met.

Abstract: An audio system includes an input node for receiving an audio input signal, a signal processor coupled to the input node to receive the audio input signal, and an output node for providing an audio output signal to a speaker. The audio system includes a first fuzzy logic controller configured to receive sampled signals related to the audio input signal and a gain of the signal processor, the first fuzzy logic controller configured to determine a risk level of the audio output signal. The audio system also includes a second fuzzy logic controller configured to receive feedback signals related to the audio output signal and to determine a correction factor. Further, the audio system is configured to determine a control signal based on the risk level and the correction factor, and to adjust the audio output signal based on the control signal.

Abstract: An apparatus for receiving audio data from a nomadic device is provided. The apparatus comprises a first portable speaker for being operably coupled to the nomadic device. The nomadic device receives first information indicative of a first crossover frequency for the first portable speaker and determines the first crossover frequency based on the first information. The nomadic device transmits the audio data to the first portable speaker in accordance to the first crossover frequency. The first portable speaker is configured to playback the audio data at the first crossover frequency.

Abstract: The present invention relates to a speaker, and a method and surround sound system for processing multi-channel audio signals in each of a plurality of audio output sources for generation of surround sound in a listening area. In particular, the system comprises a transmitter for transmitting a left channel (L) signal and a right channel (R) signal to a speaker. The speaker comprises a processing unit configured to (a) receive an audio signal having a left channel (L) signal and a right channel (R) signal; (b) process separately and independently the L and R audio signals to produce processed signals; and (c) mix the processed signals to produce the surround sound signal.

Abstract: A speaker control device in one embodiment includes an oscillator connected in parallel with a drive circuit for driving a speaker, the oscillator changing an oscillation frequency according to a voltage, and a control circuit detecting a variation in the oscillation frequency of the oscillator, and adjusting an amount of current supplied to the speaker by the drive circuit in the case where a variation in the voltage exceeds an allowable value.

Abstract: A system is provided for limiting distortion of an audio speaker. The system includes a first lowpass filter circuitry that is configured to receive a system input signal and generate a first lowpass filtered output signal. The system also includes a first limiter circuitry that is configured to limit the first lowpass filtered output signal. Limiting the first low pass filtered output signal includes reducing an amplitude of the first lowpass filtered output signal below a first predetermined threshold value, thereby to generate a first limited output signal. A second lowpass filter circuitry is configured to receive the first limited output signal and to generate a second lowpass filtered output signal.

Abstract: A bass enhancement system can provide an enhanced bass effect for speakers, including relatively small speakers. The bass enhancement system can apply one or more bass enhancements to an input audio signal. For example, the bass enhancement system can exploit how the human ear processes overtones and harmonics of low-frequency sounds to create the perception that non-existent (or attenuated) low-frequency sounds are being emitted from a loudspeaker. The bass enhancement system can generate harmonics of at least some low-frequency fundamental frequencies in one embodiment. Playback of at least some harmonics of a low-frequency fundamental frequency can cause a listener to perceive the playback of the low-frequency fundamental frequency. Advantageously, in certain embodiments, the bass enhancement system can generate these harmonics without performing processing-intensive pitch-detection techniques or the like to identify the fundamental frequencies.

Abstract: A portable electronic device having a speaker module is provided. The speaker module includes: a speaker unit; a back chamber; an acoustic region coupled between the speaker unit and the back chamber; and a porous material at least partially filling a portion of the front chamber, the back chamber, or the channel, for improving and extending bass performance and relieving a acoustic effect caused by the acoustic region coupling the speaker unit and back chamber. Moreover, the portable electronic device is advantageously able to compensate a resonance degradation caused by the porous material.

Abstract: A signal processing device includes an excessive input estimating unit that estimates excessive input of a target signal, a controller that calculates frequency characteristics which will lessen the excessive input of the target signal from the excessive input information estimated by the excessive input estimating unit, and a frequency characteristic modification unit that modifies the frequency characteristics of the target signal in accordance with the frequency characteristics the controller calculates.

Abstract: An apparatus for receiving audio data from a nomadic device is provided. The apparatus comprises a first portable speaker for being operably coupled to the nomadic device. The nomadic device receives first information indicative of a first crossover frequency for the first portable speaker and determines the first crossover frequency based on the first information. The nomadic device transmits the audio data to the first portable speaker in accordance to the first crossover frequency. The first portable speaker is configured to playback the audio data at the first crossover frequency.

Abstract: A method and apparatus for handling data compression in a mobile communication network. A node located between a mobile terminal and a mobility anchor point determines compression information relating to data transferred between the node and the mobile terminal. The node then sends the compression information to a network repository node for storage and retrieval by a further node in an event that subsequent mobility of the mobile terminal leads to transferring data between the mobile terminal and the further node.

Abstract: The present invention provides for methods and systems for digitally processing an audio signal to reproduce high quality sounds on various materials. In various embodiments, a method comprises filtering the signal with a low shelf filter and/or high shelf filter, passing the signal through a first compressor that, filtering the signal again with a low shelf filter and/or high shelf filter, processing the signal with a graphic equalizer based on a selected material profile, passing the signal through a second compressor, and outputting the signal to a transducer.

Abstract: The present invention relates to a method for dynamically adjusting a gain of parametric equalizer according to an input signal, a dynamic parametric equalizer employing the same and a dynamic parametric equalizer system employing the same wherein a gain of parametric equalizer is dynamically adjusted according to a level of an input digital audio signal to prevent distortion of output signal.

Abstract: This specification describes technologies relating to modifying digital audio data using a meta-parameter. A method is provided that includes receiving digital audio data; receiving an input modifying a meta-parameter to a particular meta-parameter value associated with two or more parameters according to a particular mapping, each of the two or more parameters being associated with the digital audio data; modifying at least a first parameter of the two or more parameters based on the particular meta-parameter value and the particular mapping such that a first meta-parameter value causes a modification of the first parameter associated with a volume control of the digital audio data and a second meta-parameter value causes a modification of a second parameter of the two or more parameters associated with dynamic range control of the digital audio data; and generating modified digital audio data from the two or more modified parameters of the digital audio data.

Abstract: A communication system for communicating with at least one loudspeaker is described where the loudspeaker is connected to audio equipment over standard two-wire speaker wire operable to carry an audio signal. The communication system includes a master node in electrical communication with a signal path carrying the audio signal between the audio equipment and the loudspeaker, the master node also including an interface with the audio equipment, a data encoder operable to encode data signals, a data transceiver operable to place the data signals onto the audio signal at frequencies above audio frequencies. The communication system also includes at least one slave node in electrical communication with the audio signal and each loudspeaker, the slave node including a data transceiver operable to receive data signals from the master node, a data decoder, and an interface able to communicate with the loudspeaker.

Abstract: Peak reduction and power limitations are used to prevent distortion and protect components. In a cellular telephone, peak reduction can be based on battery power level to prevent electrical distortion from saturation. In addition peak reduction can be used to prevent mechanical distortion such as rub and buzz. Dynamic range compression can be used for peak reduction. In another application dynamic range compression can be used to control the power output to protect a speaker from damage. One example of a dynamic range compressor/peak limiter comprises a look-ahead buffer and an analysis engine. For example, the look-ahead buffer holds a window of samples of a signal. The analysis engine selects a gain envelope function on the basis of the samples, for example, by selecting the Pth sample in the buffer whenever that sample exceeds a given threshold.

Abstract: The loudness of an audio signal is adjusted while reducing changes in its perceived spectral balance, using a dynamically-controllable filter having a high-frequency response characteristic and a low-frequency response characteristic, controlled by dynamically-changing information on the desired gain in each of a plurality of frequency bands of the audio signal.

Abstract: Integrated processing of multimedia signals can eliminate unnecessary signal processors and converters without losing the functionality of typical home entertainment system components. The integrated multimedia system includes a main player that captures and processes signals digitally. The main player may adjust the audio signal to provide audio output of equal loudness across all frequencies by accounting for sensitivity of the human ear for sounds of varying frequencies. The main player can also account for perceived differences in loudness based on the angle of a listener to a speaker by detecting the position of a user and making an adjustment accordingly. The invention further provides a speaker that has embedded performance characteristics or an identifier that allows the system to provide an optimal speaker driving current for a particular system or determine how that speaker would be best implemented in the integrated system.

Abstract: A computer-implemented method and system are provided for profiling or analyzing audio data based on an audio parameter. The audio data is divided into audio data into segments. Each segment is characterized by the audio parameter in a defined range. Effects are selected according to the segments meeting criteria for the audio parameter of the effects. A media presentation is authored or generated using the selected effects.

Abstract: Audio circuits for a mobile terminal include a microphone and an audio signal processing circuit having an input coupled to the microphone. The audio signal processing circuit has a switchable portion and is configured to switch between a voice mode utilizing the switchable portion and a full fidelity mode not utilizing the switchable portion responsive to a mode signal coupled thereto.

Abstract: In summary, this application describes a psycho-acoustically motivated, parametric description of the spatial attributes of multichannel audio signals. This parametric description allows strong bitrate reductions in audio coders, since only one monaural signal has to be transmitted, combined with (quantized) parameters which describe the spatial properties of the signal. The decoder can form the original amount of audio channels by applying the spatial parameters. For near-CD-quality stereo audio, a bitrate associated with these spatial parameters of 10 kbit/s or less seems sufficient to reproduce the correct spatial impression at the receiving end.

Abstract: In summary, this application describes a psycho-acoustically motivated, parametric description of the spatial attributes of multichannel audio signals. This parametric description allows strong bitrate reductions in audio coders, since only one monaural signal has to be transmitted, combined with (quantized) parameters which describe the spatial properties of the signal. The decoder can form the original amount of audio channels by applying the spatial parameters. For near-CD-quality stereo audio, a bitrate associated with these spatial parameters of 10 kbit/s or less seems sufficient to reproduce the correct spatial impression at the receiving end.

Abstract: Disclosed herein are apparatus, method, and computer program product whereby a device receives an acoustic signal, where the acoustic signal has a variable sound pressure level. A device outputs an electrical signal from an input audio transducer, where the input audio transducer is connected to a supply voltage. The device determines a distortion level of the electrical signal; and increases or decreases the supply voltage based on the distortion level.

Abstract: The present invention provides methods and systems for digitally processing audio signals. Some embodiments receive an audio signal and converting it to a digital signal. The gain of the digital signal may be adjusted a first time, using a digital processing device located between a receiver and a driver circuit. The adjusted signal can be filtered with a first low shelf filter. The systems and methods may compress the filtered signal with a first compressor, process the signal with a graphic equalizer, and compress the processed signal with a second compressor. The gain of the compressed signal can be adjusted a second time. These may be done using the digital processing device. The signal may then be output through an amplifier and driver circuit to drive a personal audio listening device. In some embodiments, the systems and methods described herein may be part of the personal audio listening device.

Abstract: The present invention provides for methods and systems for digitally processing audio signals by adjusting the gain of a signal a first time, using a digital processing device located between a radio head unit and a speaker. The methods and systems may filter the adjusted signal with a first low shelf filter, compress the filtered signal with a first compressor, process the signal with a graphic equalizer, compress the processed signal with a second compressor and adjust the gain of the compressed signal a second time. The methods and systems may also filter the signal received from the first low shelf filter with a first high shelf filter prior to compressing the filtered signal with the first compressor. The signal may be filtered with a second low shelf filter prior to processing the signal with the graphic equalizer. Some embodiments filter the signal with a second high shelf filter after the signal is filtered with the second low shelf filter.

Abstract: The loudness of an audio signal is adjusted while reducing changes in its perceived spectral balance, using a dynamically-controllable filter having a high-frequency response characteristic and a low-frequency response characteristic, controlled by dynamically-changing information on the desired gain in each of a plurality of frequency bands of the audio signal.

Abstract: Disclosed are a method of generating a watermark, a method of generating a broadcast content including the same, and a watermarking system. The method of generating a watermark according to an exemplary embodiment of the present disclosure includes: determining a size of a target image to which a broadcast content is provided; determining the number of watermarks to be inserted into one frame of the target image; generating a watermark insert pattern by using the size of the target image and the number of watermarks; and scaling up the watermark insert pattern to a predetermined pattern for inserting the watermark insert pattern into an original image of the broadcast content.

Abstract: The present invention provides for methods and systems for digitally processing an audio signal. Specifically, the present invention provides for a headliner speaker system that is configured to digitally process an audio signal in a manner such that studio-quality sound that can be reproduced.

Abstract: A system for generating a security mark includes a data reception component that receives information. A security mark generation component in communication with the data reception component generates at least one security mark configuration based at least in part upon the received information. The at least one security mark configuration includes at least one simulation mark which resembles a natural feature. An application component applies one configuration of the at least one security mark configurations to a recipient. The applied security mark configuration obeys at least one rule whereby the security mark is distinguishable from the natural feature which it resembles by a system for detection of security marks.

Abstract: Provided are a method and apparatus for controlling a sound field through an array speaker. The method includes calculating a coefficient of a filter that controls sound pressure of an input signal, based on a sound pressure ratio of a suppression area that suppresses sound emitted from an array speaker and an emphasis area that emphasizes the sound, and sound pressure efficiency in the emphasis area, generating a plurality of output signals that focuses the sound to the emphasis area by filtering the input signal based on the calculated coefficient of the filter, and outputting a sound field controlled sound based on the generated plurality of output signals. Accordingly, a listener in a predetermined direction and distance from the array speaker can clearly hear sound, without wearing an earphone or a headset so as to focus the sound only to the listener.

Abstract: A system and method provide at least a single stage optimization process which maximizes the flatness of the net subwoofer and satellite speaker response in and around a cross-over region. A first stage determines an optimal cross-over frequency by minimizing an objective function in a region around the cross-over frequency. Such objective function measures the variation of the magnitude response in the cross-over region. An optional second stage applies all-pass filtering to reduce incoherent addition of signals from different speakers in the cross-over region. The all-pass filters are preferably included in signal processing for the satellite speakers, and provide a frequency dependent phase adjustment to reduce incoherency between the center and left and right speakers and the subwoofer. The all-pass filters are derived using a recursive adaptive algorithm.