Abstract: The present disclosure relates to the inclusion of amplitude compression inside a hearing aid remote microphone or audio streaming device. Compressor design is improved by using one local and one remote compressor operating in parallel. The subject matter will reference remote microphones as the primary use case. In one embodiment, hearing aid microphone audio and remote microphone audio are treated as two separate streams within the hearing aid, assigning each to a compressor and mixing the audio streams afterward. In another embodiment, a compressor is developed for the remote microphone to offload that portion of the signal processing.

Abstract: A system for attenuating vehicle audio for mobile application audible statements, the system comprising a vehicle computing system having one or more processors for playing audio in the vehicle from a plurality of different audio sources. The system also includes a wireless transceiver for communicating signals from a mobile computing device for processing by the vehicle computing system. The vehicle computing system is configured to receive a wireless signal from the mobile computing device representative of an audible statement to be played in the vehicle. If the vehicle computing system is playing audio in the vehicle at the time the wireless signal is received from the mobile computing device, determine an amount to attenuate the audio based on the source of the audio. Attenuate the audio and play the audible statement based on the determination.

Abstract: A terminal for transmitting a voice signal, comprising: a transmitter having a first transmission module arranged to transmit signals over a packet-switched data network and a second transmission module arranged to transmit signals over a circuit-switched telephony network; a microphone for generating a voice signal; and signal processing apparatus configured to generate a first signal and a second signal from the voice signal, each representing information from the voice signal over a same portion of time. The signal processing apparatus is further configured to supply the first signal to the first transmission module for transmission to a receiver via the packet-switched data network, and to supply the second signal to the second transmission module for transmission to the receiver via the circuit-switched telephony network. There is also provided a terminal for receiving such signals and reconstructing the voice signal, and corresponding methods and program products.

Abstract: A hearing aid function provision method and apparatus is provides a hearing-impaired person with the audio enhanced based on the hearing-impaired information shared through Social Network Service (SNS). A user device operates in an environment-adaptive hearing-aid mode. The hearing aid function method includes receiving, at the device, audio input by a device owner; enhancing the audio based on hearing impairment information of a hearing impaired person; and outputting the enhanced audio. The audio may be enhanced a number of ways that include amplification, shifting, spreading, and division that can be based on an audiogram performed by the device or such information provided to the device.

Abstract: A system and methods for controlling a mobile electronic device is disclosed. A notification in response to a prescribed event is outputted. A change in a position of the mobile electronic device is detected. A strength of the notification is reduced, when a change in the position of the mobile electronic device is detected while the notification sound is being outputted.

Abstract: Systems, devices and methods are provided for audio volume adjustment. For example, a noise volume level of current ambient noise and a first volume level of a first current audio signal of a first application are acquired using a sound collecting device. The first volume level and the noise volume level are compared using one or more data processors to obtain a first comparison result. The first volume level is adjusted based on at least information associated with the first comparison result and a reference volume level.

Abstract: A loudspeaker includes a plurality of electro-acoustic drivers that are configured to be positioned so that an axis running through an acoustic center of the drivers is one of a substantially straight line, J-shape, reverse J-shape, and C-shape. The loudspeaker also includes at least one sensor to detect a configuration of the plurality of electro-acoustic drivers, and a processing device configured to automatically adjust one or more of an equalization and a gain applied to an audio signal output through the electro-acoustic drivers based on the detected configuration.

Abstract: Aspects of the present disclose involve a method, a device, and a system comprising a processor and a machine-readable storage medium storing a set of instructions for providing audio level safeguards for output audio signals. In example embodiments, the method includes accessing a most recent audio level of an audio output device and determining whether the most recent audio level transgresses a threshold value. Based on determining the most recent audio level transgresses the threshold value, a current audio level output by the audio output device is set to the threshold value. Thereafter, the current audio level is gradually increased, at repeated time intervals, until it reaches the most recent audio level.

Abstract: A data transmitting apparatus is provided. The data transmitting apparatus includes a packet generator configured to generate a packet including a plurality of sub packets and a transmitter configured to transmit the generated packet to a data receiving apparatus. Each of the plurality of sub packets includes audio data corresponding to content among a plurality of contents.

Abstract: This audio signal output device includes an audio signal output portion, a volume setting information acquisition portion, and a control portion acquiring maximum volume information of volume information of an audio signal of content. The control portion is configured to calculate the maximum output volume of an external device on the basis of the maximum volume information of the content and volume setting information of the external device and perform control of lowering the output volume of the external device when the maximum output volume is higher than a prescribed volume threshold.

Abstract: A portable radio (100) having a control knob (102) can further enable additional radio control functions through a rapid back and forth bi-directional rotation (202, 204, 206) of the knob (102). No additional buttons or menu features are required.

Abstract: A method and apparatus trigger audio reduction using an ultrasonic signal. Device audio can be output at a first device audio setting from a device through a headphone coupled to the device. An ultrasonic signal can be received. Device audio volume can be reduced through the headphone to output the device audio at a second device audio setting through the headphone in response to receiving the ultrasonic signal. An external audio signal can be received. The external audio signal can be sent through the headphone in response to receiving the ultrasonic signal.

Abstract: Volume limiting systems and methods are operable to limit volume output from media presentation devices. An exemplary embodiment detects a sound using a microphone, wherein the sound corresponds to an audio output of at least one controlled media presentation device, and wherein the microphone is remotely located from the at least one controlled media presentation device; compares a level of the detected sound with a predefined maximum volume limit; generates a volume output limit command in response to the detected sound exceeding the predefined maximum volume limit; and communicates the volume output limit command to the media presentation device. The media presentation device then reduces a volume level of its audio output. In some instances, volume may be limited during user specified periods.

Abstract: The disclosure recites a feedback system and method for adjusting output sounds generated by a communication device. The feedback system comprises: a transducer; a microphone; and an enclosure for the communication device defining a first location for the placement of the transducer, a second location for the placement of the microphone adjacent to the transducer, a first port connecting the first location to a cavity located on a top side of the enclosure, a second port connecting the second location to the cavity. For the system, the first port connects to the cavity beside the second port.

Abstract: The application discusses a computer implemented method and apparatus for performing audio equalization in an audio receiver device, such as an integrated receiver/decoder or set top box, or integrated TV, connected to one or more audio playback devices, such as a television unit, computer screen and speakers, amplifier or home theater equipment. The method and apparatus use an equalization process which compares audio signals received in different audio formats (e.g. MPEG-1 Layer II, AC-3 2.0, AC-3 5.1 and HE-AAC) with one another, allowing a correction gain factor to be determined for equalizing the perceived loudness of the signals when played-back at a connected playback device. The correction gain factor is then applied in the audio receiver device before output.

Abstract: A microphone circuit has a clip detection circuit (30) which detects when an analog to digital converter (ADC, 12) output has reached a threshold. A variable capacitance (34a, 34b, 34c, 34d), which functions as a variable input load associated with the microphone (11), is controlled based on the clip detection circuit output, the feedback is thus based on the ADC output level, and the processing of this signal can be implemented without requiring baseband processing of the signal—it can simply be based on a state of the ADC output.

Abstract: A transcoding tool for transcoding an audio stream to be played at a playback device is provided. The transcoding tool comprises a receiving section adapted to receive at least one bit stream comprising an audio stream and metadata associated with the audio stream. The transcoding tool further comprises a processing section connected to the receiving section and adapted to create a processed audio stream based on the audio stream and metadata, and a transmitting section connected to the processing section and adapted to transmit the created processed audio stream to the playback device.

Abstract: In a method for adjusting a volume setting of an electronic device, a current location and a current time are obtained. The method determines whether a volume state file corresponding to the executed application is stored in a storage system. When no volume state file of the executed application is stored, the volume state file is created and a time period corresponding to the current time, the current location and a volume level corresponding to a preset volume setting are recorded as a volume state package into the volume state file. When the volume state file of the executed application is stored in the storage system, the volume setting of the electronic device is adjusted according to a volume level of a matching volume state package matching the current time and the current location in the volume state file.

Abstract: A wireless device is provided that makes use of other nearby audio transducer devices to generate a surround sound effect for a targeted user. To do this, the wireless device first ascertains whether there are any nearby external microphones and/or loudspeaker devices. An internal microphone for the wireless device and any other nearby external microphones may be used to ascertain a location of the desired/targeted user as well as the nearby loudspeaker devices. This information is then used to generate a surround sound effect for the desired/targeted user by having the wireless device steer audio signals to its internal loudspeakers and/or the nearby external loudspeaker devices.

Abstract: In accordance with embodiments of the present disclosure, an apparatus for providing an output signal to an audio transducer may include an analog signal path portion, a digital-to-analog converter (DAC), and a control circuit. The analog signal path portion may have an audio input for receiving an analog input signal, an audio output for providing the output signal, and a selectable analog attenuation, and may be configured to generate the output signal based on the analog input signal and in conformity with the selectable analog attenuation. The DAC may have a selectable digital gain and may be configured to convert a digital audio input signal into the analog input signal in conformity with the selectable digital gain. The control circuit may be configured to select the selectable analog attenuation and select the selectable digital gain based on a magnitude of a signal indicative of the output signal.

Abstract: Embodiments are provided for updating a playlist that has been added to a playback queue in response to changes to the playback queue. The playback queue may be associated with a zone of a network media system such that items in the playback queue are to be rendered by the zone. The playlist may include one or more items playable by the zone, and may be stored separately from where the playback queue is maintained. Embodiments are also provided for updating a playback queue in response to modifications to a playlist included in the playback queue. In some cases, a user modifying the playlist or playback queue may be prompted upon making the modifications whether to also apply the modification to the playback queue or playlist, respectively.

Abstract: An apparatus for reverberation time extension is provided. The apparatus includes a module for calculating wave field synthesis information, and a signal processor for generating a plurality of audio output signals for a plurality of loudspeakers based on the audio input signals that have been recorded by a plurality of microphones, and based on the wave field synthesis information. Further, the apparatus includes an operating unit for determining a virtual position of one or several virtual walls. The module for calculating wave field synthesis information is implemented to calculate the wave field synthesis information based on the virtual position of the one or several virtual walls. Further, the virtual position is adjustable by the operating unit for at least one of the virtual walls.

Abstract: A sound processing apparatus processes a first sound signal and a second sound signal. The sound processing apparatus includes a first adjustor which adjusts a volume of the first sound signal, a second adjustor which adjusts a volume of the second sound signal, and a control unit which causes the first adjustor and the second adjustor to adjust the first sound signal gradually changed over a first gradually-change time and the second sound signal gradually changed over a second gradually-change time, wherein the second gradually-change time is shorter than and included in the first gradually-change time.

Abstract: The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Abstract: To control loudness during a junction between different types of broadcast content, such as a junction between programme and commercial or promotional content, representative loudness values for content respectively before (P) and after (C) the junction are received from a playout automation system. A time-varying gain control is applied before and after the junction in order to smooth loudness around the junction. The audio gain is smoothly increased prior to the junction to a gain (P+C)/2P times higher than the original gain value. Then, the gain is reduced shortly before the junction to a value (P+C)/2C times lower than the original gain value. After the junction, the gain is returned smoothly to the original value.

Abstract: A microphone includes a first signal input, a second signal input and a control unit coupled to the first signal input and the second signal input. The control unit is configured to, upon receiving a support signal at one of the first signal input and second signal input, process a signal received at the other of the first signal input and second signal input as an incoming data signal.

Abstract: A novel audio ducking method that is aware of the loudness levels of the audio content is provided. The method specifies a minimum loudness separation between audio tracks that are designated as masters and audio tracks that are designated as slaves. The method attenuates the volume of the slave tracks in order to provide at least the minimum loudness separation between the slave tracks and the master tracks. The amount of attenuation for a slave is determined based on the loudness levels of the slave and of a master.

Abstract: Implementations described herein may include activating an audio source associated with a graphical user interface, the audio source having an initial location associated with the graphical user interface. Some implementations may further include sensing a displacement of the audio source with respect to the initial location and determining that the displacement of the audio source exceeds a predetermined threshold. In some implementations, in response to determining that the displacement of the audio source exceeds a predetermined threshold, the process may include adjusting a level of audio associated with the audio source.

Abstract: A signal manipulator for manipulating an audio signal having a transient event may have a transient remover, a signal processor and a signal inserter for inserting a time portion in a processed audio signal at a signal location where the transient event was removed before processing by the transient remover, so that a manipulated audio signal has a transient event not influenced by the processing, whereby the vertical coherence of the transient event is maintained instead of any processing performed in the signal processor, which would destroy the vertical coherence of a transient.

Abstract: A microphone includes a volume reduction filling member therein. The volume reduction filling member includes an elastic body having an insertion hole for inserting an electrode extraction member which extracts a generated current to a back end side by contacting a front end side thereof to a fixed electrode of a microphone unit, and reducing a volume inside a unit case. The elastic body is divided into a tip-side main body portion and a base-end-side flexible flange portion through a circular groove around the insertion hole from an outer peripheral face side of a base end portion thereof. When an outer peripheral edge portion of the base-end-side flexible flange portion receives an equal pressure toward a tip direction so as to bend, the elastic body concentrates a stress into a front portion side of the insertion hole through the tip-side main body portion.

Abstract: Presented herein are techniques for controlling the level of ultrasound pairing signals generated in a teleconferencing environment. The levels of ultrasound pairing signals transmitted in a meeting room are adjusted automatically based on the ultrasound signal levels received at one or more of the sound receiving devices that can communicate with a teleconferencing endpoint in the meeting room.

Abstract: Signals in an automobile audio system having at least two near-field speakers located close to an intended position of a listener's head are adjusted such that in a first mode, audio signals are distributed to the near-field speakers according to a first filter that causes the listener to perceive a wide soundstage, and in a second mode, the audio signals are distributed to the near-field speakers according to a second filter that causes the listener to perceive a narrow soundstage. A user input of a variable value is received and, in response, distribution of the audio signals is transitioned from the first mode to the second mode, the extent of the transition being variable based on the value of the user input.

Abstract: There is described Audio driver circuit comprising a main, 1-bit digital-to-analog converter, adapted to receive a 1-bit oversampled digital audio signal. Thanks to an auxiliary digital-to-analog converter which converts into the analog domain a digital offset compensation value stored in e.g. an OTP memory register 34, and to a summing and filtering arrangement, it is possible to reduce the offset cancellation granularity in order to compensate for the offset of the main digital-to-analog converter in the output signal.

Abstract: An apparatus for providing a set of spatial cues associated with an upmix audio signal having more than two channels on the basis of a two-channel microphone signal has a signal analyzer and a spatial side information generator. The signal analyzer is configured to obtain a component energy information and a direction information on the basis of the two-channel microphone signal, such that the component energy information describes estimates of energies of a direct sound component of the two-channel microphone signal and of a diffuse sound component of the two-channel microphone signal, and such that the directional information describes an estimate of a direction from which the direct sound component of the two-channel microphone signal originates. The spatial side information generator is configured to map the component energy information and the direction information onto a spatial cue information describing the set of spatial cues associated with an upmix audio signal having more than two channels.

Abstract: In a volume control method applied to an audio processing system, firstly, a digital audio stream including a plurality of digital audio packets is received from a transmitter. It is determined whether an excitation signal is detected. The digital audio packets are sampled within a predetermined sampled period according to codecing rules, on condition that the excitation signal is detected. Output values corresponding to the digital audio packets are retrieved from the sampled digital audio packets according to the sampling operation. The digital audio packets are analyzed to generate sums of sampled frequency. An averaged volume level is calculated according to the sums of sampled frequency. An audio shift value is generated according to the averaged volume level and a predetermined standard volume level. Volume control is implemented to subsequently received digital audio packets from the digital audio stream according to the audio shift value.

Abstract: An equalizer (402) is shown, having a state variable filter in which a first MDAC (502) is coupled to a first integrator (503) and a second MDAC (504) is coupled to a second integrator (505) to allow control of the corner frequency of the state variable filter. The MDACs are each configured to provide variable attenuation in a plurality of discrete steps. A digital attenuation controller is coupled to and adjusts the attenuation of the MDACs independently, and effects a one-step increase in the corner frequency of the state variable filter by first determining the attenuation levels of the MDACs, and then controlling the attenuation levels of the MDACs in accordance with their existing attenuation levels.

Abstract: A system for attenuating vehicle audio for mobile application audible statements, the system comprising a vehicle computing system having one or more processors for playing audio in the vehicle from a plurality of different audio sources. The system also includes a wireless transceiver for communicating signals from a mobile computing device for processing by the vehicle computing system. The vehicle computing system is configured to receive a wireless signal from the mobile computing device representative of an audible statement to be played in the vehicle. If the vehicle computing system is playing audio in the vehicle at the time the wireless signal is received from the mobile computing device, determine an amount to attenuate the audio based on the source of the audio. Attenuate the audio and play the audible statement based on the determination.

Abstract: Presently described are systems and methods for identifying a mute/sound attribute of an audio sample set having first-channel samples and second-channel samples. An embodiment takes the form of a method that includes: (i) receiving the audio sample set; (ii) for each first-channel sample, determining a first max amplitude and a first min amplitude; (iii) calculating a first span based on a first function of the first max amplitude and the first min amplitude; (iv) for each second-channel sample, determining a second max amplitude and a second min amplitude; (v) calculating a second span based on a function of the second max amplitude and the second min amplitude; and (vi) identifying the audio sample set as having a sound attribute if both the first span and the second span are greater than a min-volume threshold.

Abstract: An electronic circuit (100) for use in a wireless receiver (1720). The circuit (100) includes an audio demodulator (15), an audio envelope portion (35) operable to deliver an audio envelope signal, a high pass filter (38) coupled between the audio demodulator (15) and the audio envelope portion (35), a volume control (25) fed by the audio demodulator (15) and operable to provide a controllable audio signal output, and a controller (45) responsive to the audio envelope portion (35) to provide an attenuation control to the volume control (25). Other circuits, receivers, processes of operation, processes of manufacture, and processes of testing are disclosed.

Abstract: A volume control device and a method thereof are provided. The volume control device adapted to a mobile electronic device comprises a signal generating unit generating a plus pulse signal and a minus pulse signal according to user's rotation operation, a detection unit detecting a connecting state between the volume control device and an earphone to generate a detection signal, and a control unit selecting one of a first volume modulation mode and a second volume modulation mode according to the detection signal. Under the first volume modulation mode, the control unit generates a first control signal according to the plus pulse signal or the minus pulse signal. Under the second volume modulation mode, the control unit generates a second control signal according to the plus pulse signal or the minus pulse signal. The mobile electronic device controls volume according to the first or second control signal.

Abstract: A computing device comprising a touch input device and a processor. The touch input device is configured to detect multiple touch point contacts. The processor is coupled to the touch device and configured to execute software instructions that process the multiple touch point contacts and, based on the multiple touch point contacts, adjust at least one audio setting of a plurality of audio output devices.

Abstract: According to an aspect, a mobile electronic device includes a sound emitting unit, an input unit, and a processing unit. The sound emitting unit emits a sound based on a sound signal. The input unit receives a response with respect to the sound emitted by the sound emitting unit. The processing unit determines a compensation parameter for compensating the sound to be emitted by the sound emitting unit based on correctness of the response.

Abstract: The present invention relates to an apparatus that senses the intensity of sounds and the velocity of objects. Specifically, the present invention relates to an apparatus that may prevent an audio signal from reaching a user, make a warning noise, or otherwise alert the user when a object is noisy and/or approaching the user at a threatening rate. More specifically, the present invention takes data from the surrounding area and compares it to one or more thresholds to determine an object's threat level. The present invention relates to one or more speakers, and includes, generally, one or more microphones to receive and measure the intensity of surrounding noises and one or more range sensors to detect distance, velocity, and acceleration.

Abstract: An interactive presentation environment for eMeetings or the like that provides participants with more control over what they see and hear. The interactive presentation environment may comprise a meeting recorder adapted to create a recording of a live meeting and a navigation control for selecting a portion of the recording to view during the live meeting. The interactive presentation environment may further comprise a timeline control containing a first graphical indicator associated with a live position and a second graphical indicator associated with a current position, a bookmark control adapted to mark a portion of the recording for archiving, and a display operatively connected to the meeting recorder and the navigation control.

Abstract: A computerized method adjusts two presented audio signals using an audio player. A first audio signal is received and played through an audio playing unit. When a second audio signal that is inserted to be played is received in time during the playing of the first audio signal, a peak amplitude X of a volume of the first audio signal and a peak amplitude Y of a volume of the second audio signal are extracted. Each of the peak amplitude X and the peak amplitude Y are compared with a predetermined threshold value A. The first audio signal and/or the second audio signal are adjusted according to the comparison, and the first audio signal and/or the second audio signal is outputted to the audio playing unit and played through the audio playing unit according to the adjustment for the first audio signal and/or the second audio signal.

Abstract: The dynamic setting of an increment on an amplitude scale of an output signal includes receiving a user input requesting a change in an amplitude of the output signal. The amplitude of an external signal related to the output signal is determined, and a set of increments on the amplitude scale is calculated based upon the current amplitude of the output signal and the amplitude of the external signal. The existing set of increments are replaced with the newly calculated set of increments and the amplitude of the output signal is set to the next increment of the newly calculated set of increments.

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: When an effect parameter of an effector is changed, if only a switching operation is provided to bypass the original audio signals, which are input signals of the effector, during the period required for making that change, the output sound level may drastically change before and after the switching causing one to feel a sense of discomfort. In order to eliminate such a drawback, a gain adjustment circuit is provided in a bypass channel that outputs the original audio signals by bypassing the effector. The gain adjustment circuit is controlled in accordance with a rate of the output signal level of the effector and the original audio signal level, and the output signal level of the bypass channel is adjusted thereby. Consequently, a fluctuation in the output sound level before and after switching of the output signals can be eliminated.

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

Abstract: A sound volume control device includes a sound input section, a band dividing section, an excessive output notification section, a correction band decision section and a sound output control section. The sound input section converts input sound into a sound signal. The band dividing section carries out band-division with respect to the sound signal. The excessive output notification section outputs excessive output notification information when output sound from the present device is excessive. The correction band decision section calculates audible signal distribution information when the excessive output notification information is not output and excessive signal distribution information when the excessive output notification information is output, for each band in the sound signal, and decides a correction band according to the audible signal distribution information and excessive signal distribution information.