Abstract: Embodiments of the invention provide for methods for automatically controlling loudness of an audio signal. Embodiments of the invention also provide for a device for automatically controlling loudness of an audio signal. In particular, the invention relates to controlling loudness of an audio signal to be replayed within a vehicle, such as a car.

Abstract: Disclosed are a method, an apparatus and an electronic device for controlling audio playback of multiple loudspeakers, wherein the method comprises: determining the location information of each speaker and voice signals issued by each speaker; determining the area where each speaker is located according to the location information of each speaker; determining voice instruction corresponding to each voice signal; and controlling the multiple loudspeakers to play the audio indicated by the corresponding voice instruction respectively for the area where the speaker of each voice instruction is located. According to the method and apparatus and/or the electronic device in an embodiment of the present disclosure, different audios can be played for different areas in a preset space.

Abstract: Described are techniques for sharing data among a group of mobile devices that are registered to the same user. In some instances, the shared data is sensor data captured by a mobile device or media content being processed by the mobile device. The mobile devices in the group are configured to wirelessly communicate with each other and to share status information. Using the status information, the mobile devices can keep each other updated as to the operational status of each individual mobile device and, in some aspects, the status of a mobile device with respect to the user. The sensor data or media content can be forwarded to a different mobile device in response to a determination made based on the status information. For instance, forwarding can be based on a rule that specifies one or more conditions relating to the status of a mobile device and/or user status.

Abstract: Method for operating a vehicle-mounted acoustic signal-generating device (8), which is configured to generate an acoustic compensation signal (2) for compensating acoustic interference signals (3), which result from a stream of vehicles (5) coming towards a vehicle (4) in question, comprising the steps: —detection of a stream of vehicles (5) coming towards a vehicle (4) in question, and generation of detection information describing at least one parameter of the stream of vehicles, —determination of an acoustic interference signal (3) resulting from the stream of vehicles (5) coming towards the vehicle (4) in question on the basis of the detection information, —generation of an acoustic compensation signal (2) for compensating the acoustic interference signal (3) resulting from the stream of vehicles (5) coming towards the vehicle (4) in question.

Abstract: Aspects of the present disclosure are directed to a measuring device for determining a measured variable of a measuring gas by means of a photoacoustic method. In some embodiments, the measuring device includes a flow channel for the measuring gas having at least one feed line, a photoacoustic measuring cell and a discharge line. in such an embodiment, the measuring device further includes at least one acoustic filter member tuned to a useful frequency of the measuring cell, and at least one cavity resonator arranged on the flow channel.

Abstract: A vehicle proximity notification device that notifies proximity of a vehicle by causing a speaker disposed outside a vehicle interior of the vehicle to output a warning sound. The vehicle proximity notification device includes: a controller that, when a sound pressure of a sound collected by a microphone disposed outside the vehicle interior exceeds a first threshold, sets a sound pressure of the warning sound output by the speaker to a level lower than the sound pressure of the warning sound output by the speaker when the sound pressure of the sound collected by the microphone is lower than or equal to the first threshold.

Abstract: A method includes receiving a voice command, determining, by a processing device, a first loudness parameter associated with the voice command, generating a voice response to the voice command, and determining a target loudness parameter associated with the voice response in view of the first loudness parameter associated with the voice command.

Abstract: A control device includes a first communication circuit, a microphone, and a first control circuit.

Abstract: A networked speaker system automatically adjusts certain audio speaker system settings based on whether the system is inside a building or has been moved outside. Also, techniques are described for adjusting speaker driver direction based on walls or other barriers in a room or based on the location of a listener.

Abstract: Workplace safety is a principal concern in many environments. Protecting user ears from damage due to extended exposure to unacceptably high sound volume serves as an important component to workplace safety. Monitoring a device, such as a phone, utilized by a user often provides an incomplete picture of the sound level presented to the user. As provided herein, monitoring a user's sound exposure on one device may cause the sound level presented to the user from a second device to become limited. Additionally, over time the sound level limits may be adjusted based on the cumulative historic sound exposure. As a result, the user may avoid exposure to unacceptably high sound levels originating from more than one source and/or over an extended period of time.

Abstract: Systems and techniques are generally described for audio signal processing for motion detection. In some examples, a first energy level of a first audio signal detected by a first microphone is determined over a first period of time. In some further examples, a first change in the first energy level from a first energy level value to a second energy level value is determined during the first period of time. In some other examples, a determination is made that the first change in the first energy level exceeds a threshold change value associated with the first period of time. In at least some other examples, a bias of an audio amplifier of the first device is adjusted.

Abstract: A gain function controller may be configured to: receive a first plurality of sub-band-signals; determine a frequency-domain-gain-function for a second plurality of sub-band-signals, based on: the first plurality of sub-band-signals; a power of a first full-band signal; and a predetermined compression curve; and apply the frequency-domain-gain-function to the second plurality of sub-band-signals to provide a frequency-domain-output-signal.

Abstract: The technology described in this document can be embodied in a method that includes receiving a plurality of representations of the signal corresponding to samples of the signal within a frame of predetermined time duration, and estimating a power spectral density (PSD) for each of a plurality of frequency bins. The PSD for a particular frequency bin is estimated based on a smoothing parameter calculated from a noise estimate for the particular frequency bin as obtained from samples corresponding to a preceding frame. The method also includes generating, based on the PSD for each of the plurality of frequency bins, an estimate of the steady-state noise floor, and computing a measure of spectral flatness associated with the samples within the frame. The method also includes determining that the measure of spectral flatness satisfies a threshold condition, and in response, computing an updated estimate of the steady-state noise floor.

Abstract: An example system for playing media content with a media playback device in a vehicle can be programmed to obtain a sound measurement indicative of a sound level associated with playback of the media content by the media playback device in the vehicle. The example system also can determine a deviation in an expected sound level based upon the sound measurement. Finally, the system can modify playback of the media content by the media playback device (110) based upon the deviation.

Abstract: Examples of techniques for controlling a volume of an audio device or system based on a user profile are disclosed. In one example implementation according to aspects of the present disclosure, a computer-implemented method includes generating, by a processing device, a user volume profile. The method further includes receiving, by the processing device, a volume command to change a volume level of audio. The method further includes determining, by the processing device, a volume change factor based at least in part on the user volume profile. The method further includes changing, by the processing system, the volume level of the audio based at least in part on the volume change factor.

Abstract: This disclosure describes, in part, voice-controlled light switches that act as voice-controlled endpoints at which users may provide voice commands. These light switches may physically couple to a power source, as well as to one or more appliances. The light switches may include physical switches to selectively provide power to the appliances, either in a binary fashion (on or off) or along of spectrum of positions. In either instance, the appliances coupled to the light switches may include lights, electrical outlets, home appliances (e.g., dishwashers, washing machines, etc.), factory machinery, or any other device that receives electrical power.

Abstract: In one aspect, a device includes a processor and storage accessible to the processor. The storage bears instructions executable by the processor to receive, at a microphone, voice input, and to determine an amplitude of the voice input. The instructions are executable to automatically adjust a volume of a speaker based on the amplitude of the voice input.

Abstract: An electronic device is provided. The electronic device may output a sound output event through a speaker at an output volume, generate a predicted input waveform based on the output sound event, compare a level of an input sound received via a microphone with a predicted level of the predicted input waveform, determine when a level of the input sound is lower than a threshold value for a predetermined time, and control the output volume of the sound output event when the level of the input sound is lower than the threshold value for the predetermined time.

Abstract: A distributed automatic level control function is provided, in which information relating to a common automatic level control parameter is transmitted to each of a plurality of microphone devices, wherein the information transmitted to at least one microphone device is derived from an audio sample of at least one different microphone device. Each microphone device produces the common automatic level control parameter based on the information received by the microphone device and applies the common automatic level control parameter produced by the microphone device to a distributed automatic level controller of the microphone device.

Abstract: An appliance for receiving and reading audio signals, includes a receiver of high-frequency electromagnetic waves designed to capture radio broadcast signals including a plurality of audio tracks; a wireless communication interface, separate from the receiver and designed to receive at least one regulating parameter to be applied to at least one audio track of the captured signal; and a digital signal processor designed to apply the regulating parameter to the audio track and produce a mixed signal.

Abstract: Provided is a method, non-transitory computer program product and system for an improved noise suppression technique for speech enhancement. It operates on speech signals from a single source such as either the output from a single microphone or the reconstructed speech signal at the receiving end of a communication application. The system performs background noise monitoring of an in-coming speech signal and determines its level, and performs a time domain gain calculation. The noise suppressed output signal is the gain shaped original speech signal.

Abstract: A camera system includes a first microphone, a second microphone, and a microphone controller. The first microphone and the second microphone are configured to capture audio over a time interval to produce a first captured audio signal and a second captured audio signal, respectively. The second captured audio signal is dampened relative to the first captured audio signal by a dampening factor. The microphone controller is configured to store the first captured audio signal in response to a determination that the first captured audio signal does not clip. In response to a determination that the first captured audio signal clips, the microphone controller is configured to identify a gain between the first captured audio signal and the second captured audio signal representative of the dampening factor, amplify the second captured audio signal based on the identified gain, and store the amplified second captured audio signal.

Abstract: This disclosure describes, in part, voice-controlled light switches that act as voice-controlled endpoints at which users may provide voice commands. These light switches may physically couple to a power source, as well as to one or more appliances. The light switches may include physical switches to selectively provide power to the appliances, either in a binary fashion (on or off) or along of spectrum of positions. In either instance, the appliances coupled to the light switches may include lights, electrical outlets, home appliances (e.g., dishwashers, washing machines, etc.), factory machinery, or any other device that receives electrical power.

Abstract: Method for audio volume control, the method comprising the steps of: receiving (301) a volume change request; changing (304) the volume according to a volume change step; the method being characterized in that the volume change step determination (302) is executed by applying one or more rules wherein each rule provides a definition of at least two ranges, with respect to a monitored condition, wherein the volume change step, associated with each such range, is different; storing a new volume change step in a volume step register (120).

Abstract: A method for audio matching may include sending a request to an external server to deliver a plurality of main media segments to the media device, receiving a first main media segment, and playing the received first main media segment. The method may further include determining audio parameters of the first main media segment, determining audio parameters of a first interstitial media segment to be served in proximity to the first main media segment, determining when the audio parameters of the first interstitial media segment are to be adjusted to match the audio parameters of the first main media segment, adjusting the audio parameters of the first interstitial media segment, and playing the first main video followed by the first interstitial media segment.

Abstract: An audio reproduction method and apparatus, in which an audio volume is automatically controlled based on audio energy and human auditory characteristics. The audio reproduction method includes splitting a reproduction audio signal into audio signals corresponding to a plurality of frequency bands, extracting audio energy for each of the frequency bands, and comparing the audio energy for each of the frequency bands with a predetermined threshold and controlling the volume of the audio signal corresponding to each of the frequency bands.

Abstract: A camera system includes a first microphone, a second microphone, and a microphone controller. The first microphone and the second microphone are configured to capture audio over a time interval to produce a first captured audio signal and a second captured audio signal, respectively. The second captured audio signal is dampened relative to the first captured audio signal by a dampening factor. The microphone controller is configured to store the first captured audio signal in response to a determination that the first captured audio signal does not clip. In response to a determination that the first captured audio signal clips, the microphone controller is configured to identify a gain between the first captured audio signal and the second captured audio signal representative of the dampening factor, amplify the second captured audio signal based on the identified gain, and store the amplified second captured audio signal.

Abstract: A method of adjusting a loudness of an audio signal in real time may include receiving an electronic audio signal and dividing the audio signal into a plurality of frames. Processing of the frames may be performed in real time. The processing may include measuring initial loudness values for blocks of samples in a frame to produce a plurality of initial loudness values, and computing a weighted average of at least some of the initial loudness values. The weights may be selected based on one or more of the recency of the initial loudness values, variation of the initial loudness values, and estimated information content of the audio signal. The processing may further include selectively discarding at least some of the loudness values that reach an adaptive loudness threshold. Weights can be adjusted based on the variation of the loudness values of the audio signal.

Abstract: A camera system includes a first microphone, a second microphone, and a microphone controller. The first microphone and the second microphone are configured to capture audio over a time interval to produce a first captured audio signal and a second captured audio signal, respectively. The second captured audio signal is dampened relative to the first captured audio signal by a dampening factor. The microphone controller is configured to store the first captured audio signal in response to a determination that the first captured audio signal does not clip. In response to a determination that the first captured audio signal clips, the microphone controller is configured to identify a gain between the first captured audio signal and the second captured audio signal representative of the dampening factor, amplify the second captured audio signal based on the identified gain, and store the amplified second captured audio signal.

Abstract: A method is provided for automatically activating an in-cabin camera system whenever the control system determines that the volume level of sounds detected within the car are in excess of a preset sound level, thereby allowing the driver under certain circumstances to monitor occupants of the rear portion of the passenger cabin without stopping the car or turning around in their seat. The system can be configured to analyze the detected sounds and compare those sounds to preset sound patterns, thus helping the system to distinguish between routine background noise and any sounds that indicate that activation of the in-cabin camera system is warranted.

Abstract: An acoustic signal's high frequency component extracted by an HPF 13 is multiplied in a multiplier 16 by the first coefficient to control the level of the high frequency component. The first coefficient is generated by a coefficient generation portion 21 in accordance with the full band level of an output signal. A low frequency component extracted by an LPF 15 is multiplied in a multiplier 17 by the second coefficient to control the level of the low frequency component. The second coefficient is generated by a coefficient generation portion 24 in accordance with the level of the output signal's low frequency component extracted by an LP filter 22. The level control of the low frequency component of the output signal and the level control of the high frequency component of the output signal are done separately, resulting in the loudness control which is suitable for the characteristic of the acoustic signal.

Abstract: A method and system for peak limiting of speech signals for delay sensitive voice communication is disclosed. In an embodiment, a position of a sample with highest magnitude within a current block of samples is determined. Further, a peak gain to be applied for the current block of samples to bring down the highest magnitude to a predetermined threshold value is determined. Furthermore, a gain delta by which an old gain is updated to the peak gain is computed. Then, a gain factor is computed for the current block of samples based on the position of the sample with highest magnitude and the gain delta. Subsequently, the gain factor is set to a predetermined minimum gain factor when the computed gain factor is less than the predetermined minimum gain factor. In addition, gain is applied to the current block of samples using the gain factor.

Abstract: System, method, and computer program product to provide targeted and variable information during a remote conference, by: responsive to a first person joining the remote conference, playing a first tone having a first acoustical characteristic indicating an arrival of the first person, responsive to a second person joining the remote conference, playing a second tone having a second acoustical characteristic indicating an arrival of the second person, the second acoustical characteristic different than the first, and responsive to one of the first person and the second person leaving the remote conference, playing a third tone having a third acoustical characteristic indicating a departure of one of the first person and the second person, the third acoustical characteristic different than the second, a type of the first, second, and third acoustical characteristics being the same, the first, second, and third tone audible to at least one person on the remote conference.

Abstract: System, method, and computer program product to provide targeted and variable information during a remote conference, by: responsive to a first person joining the remote conference, playing a first tone having a first acoustical characteristic indicating an arrival of the first person, responsive to a second person joining the remote conference, playing a second tone having a second acoustical characteristic indicating an arrival of the second person, the second acoustical characteristic different than the first, and responsive to one of the first person and the second person leaving the remote conference, playing a third tone having a third acoustical characteristic indicating a departure of one of the first person and the second person, the third acoustical characteristic different than the second, a type of the first, second, and third acoustical characteristics being the same, the first, second, and third tone audible to at least one person on the remote conference.

Abstract: Volume adjustment based on listener position is disclosed. A position of one or more speakers is identified, and a position of a listener is tracked. For each of the one or more speakers, a changing distance between that speaker and the listener is assessed. A volume of that speaker is automatically adjusted in real-time based on a current distance between that speaker and the listener.

Abstract: A method of attenuating an input signal to obtain an output signal is described. The method comprises receiving the input signal, attenuating the input signal with a gain factor to obtain the output signal, applying a filter having a frequency response with a frequency-dependent filter gain to at least one of a copy of the input signal and a copy of the output signal to obtain a filtered signal, the frequency-dependent filter gain being arranged to emphasize frequencies within a number N of predetermined frequency ranges, N>1; wherein the filter comprises a sequence of N sub-filters, each one of the N sub-filters having a frequency response adapted to emphasize frequencies within a corresponding one of the N predetermined frequency ranges; determining a signal strength of the filtered signal, and determining the gain factor from at least the signal strength.

Abstract: For a media clip that includes audio content, a novel method for performing dynamic range compression of the audio content is presented. The method performs an analysis of the audio content. Based on the analysis of the audio content, the method generates a setting for an audio compressor that compresses the dynamic range of the audio content. The generated setting includes a set of audio compression parameters that include a noise gating threshold parameter (“noise gate”), a dynamic range compression threshold parameter (“threshold”), and a dynamic range compression ratio parameter (“ratio”).

Abstract: Ambient sound is masked by receiving an audio signal, monitoring an acoustic signal that includes components of both the ambient sound the audio signal, determining a sound level sufficient to at least partially mask the ambient sound, and reproducing the received audio signal to produce an output acoustic signal at the determined sound level.

Abstract: A pseudo noise superimposing unit superimposes a pseudo noise (M-sequence) to an audio signal picked up by a microphone and outputs the superimposed signal to an amplifying system. An calculating unit calculates a correlation value between the audio signal picked up by the microphone and the pseudo noise. The calculating unit estimates a gain of a closed loop based on the correlation value. A gain control unit suppresses a gain of the audio signal based on the estimated gain of the closed loop.

Abstract: One variation of a method for augmenting a listening experience of a user through an audio device includes: detecting a location of the audio device; selecting a set of audio output feedbacks, each audio output feedback in the set of audio output feedback entered by an individual and associated with a physical site proximal to the location; identifying a common feature across audio output feedbacks within the set of audio output feedbacks; transforming an audio signal into a processed audio signal according to a hearing profile of the user and the common feature; and outputting the processed audio signal through the audio device.

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 method for modifying a current sound-volume value assigned by an on-board system to an acoustic alert signal emitted on board an aircraft is provided. The method includes furnishing, on board the aircraft, the on-board system with at least one information item defining a modification of the current sound-volume value to a new sound-volume value in accordance with a predetermined rule.

Abstract: An audio system includes a level controller receiving an input audio signal from an audio medium. The level controller includes a gain adjust module producing an output audio signal based on the input audio signal. A filter arrangement is communicatively coupled to the level controller. The filter arrangement receives the output audio signal and divides the output signal into a plurality of filtered signals. Each of the filtered signals corresponds to a respective frequency band. A multiband dynamics controller is communicatively coupled to the filter arrangement. The multiband dynamics controller receives the filtered signals and applies a respective gain characteristic to each of the filtered signals to thereby produce a plurality of gain adjust signals. The gain adjust signals are combined into a multiband dynamics controller output signal.

Abstract: The audio processing system disclosed in the invention comprises an audio processor and an audio amplifier. The audio processor receives a data signal to generate a processed signal, and comprises at least one gain control circuit and at least one operational amplifier. The gain control circuit generates a gain signal according to a volume control signal, a reference signal, and a feedback signal. The operational amplifier couples to the gain control circuit and amplifies the data signal by the gain signal to generate a processed signal. The audio amplifier couples to the audio processor to receive and amplify the processed signal, wherein an amplified signal is generated.

Abstract: An analog front circuit for a medical device includes an automatic gain control loop and a 2-order-3-bit-quantization Sigma-Delta analog-to-digital converter. The automatic gain control loop is configured to implement automatic control of loop gain and output an analog signal to the 2-order-3-bit-quantization Sigma-Delta analog-to-digital converter. The 2-order-3-bit-quantization Sigma-Delta analog-to-digital converter is configured to convert the analog signal output from the automatic gain control loop into a digital code and output the digital code to a DSP for processing.

Abstract: A signal processing apparatus includes: one or more detection means for detecting movement of a diaphragm of a speaker in correspondence with feedback methods that are different feedback methods; analog-to-digital conversion means for converting one or more detection signals acquired by the detection means into a digital form; feedback signal generating means for generating feedback signals corresponding to the feedback methods using the digital detection signals; synthesis means for combining an audio signal to be output as a driving signal of the speaker with the feedback signals; correction equalizer means for setting an equalizing characteristic to allow a sound reproduced by the speaker to have a target frequency characteristic by changing the digital audio signal; feedback operation setting means for setting feedback methods in which a feedback operation up to combining the audio signal with the feedback signal is performed and the feedback operation is not performed equalizing characteristic changing a

Abstract: Disclosed is a tuning sound (TS) feedback apparatus. The tuning sound (TS) feedback apparatus includes: a TS pick-up casing which is disposed and fixed inside or near the hearing organ of a sound engineer who manages tuning operations for sound output from a sound output terminal of a sound device having an equalizer; a sound/electric conversion element which is included in the TS pick-up casing, picks up the TS output from the sound output terminal of the sound device, and converts the picked-up TS into an electrical signal; and a TS output module which is included in the TS pick-up casing, converts the TS converted into the electrical signal by the sound/electric conversion element into wired or wireless communication data to be used for external communication, and transmits the data to the sound device.

Abstract: A power amplifying circuit includes: a signal input terminal, a common-mode voltage input terminal, a power tube control unit, a feedback loop, a driving unit, a first switching element, a second switching element, a low-pass filter, a speaker, a source voltage terminal, and a ground terminal, wherein the power tube control unit includes: an operational amplifier, a first switch, a second switch, a third switch, a fourth switch, a fifth switch, a sixth switch, a seventh switch, a eighth switch, a ninth switch, a third switching element connected with the third switch and the fourth switch, a fourth switching element connected with the fifth switch and the sixth switch, and a comparator connected with the third switching element and the fourth switching element. The present invention is able to realize functions of CLASS AB amplifier and CLASS D amplifier.

Abstract: A method for controlling a maximum signal level output to headphones of a wireless device is provided. The method includes: determining an impedance of the headphones; determining a carrier specific maximum signal level for headphones having the impedance; and, adjusting an audio amplifier of the wireless device coupled to the headphones to restrict the maximum signal level output to the headphones to the carrier specific maximum signal level.

Abstract: There is provided a method for use in a computer for redirecting an audio stream from a first audio endpoint to a second audio endpoint in the computer having a controller and a computer operating system for execution by the controller. The method includes directing the audio stream from a client application running on the computer operating system through a first audio resource stack of the computer to the first audio endpoint. The method further includes creating an audio endpoint bridge using the computer operating system, where the audio endpoint bridge provides a path for the audio stream to be directed from the first audio resource stack to pass through a second audio resource stack to the second audio endpoint. The method may also enhance the audio stream to reduce noise and redirect the audio stream to the second audio endpoint using the audio endpoint bridge.