Abstract: An amplifying device is provided that reduces power consumption and quickly starts amplification of an audio signal of a channel to be used. A control section 17 inputs a control signal for instructing operation or standby into post-amplifying sections 11b and 12b, and inputs a control signal for instructing standby to post-amplifying sections 13b and 14b. In the post-amplifying section 11b, when the control signal for instructing standby or operation is input from the control section 17, a modulation circuit 52 modulates the analog audio signal into a switching signal. An output stage circuit 54 amplifies the output signal. When the control signal for instructing the operation is input from the control section 17, a driving circuit 53 drives the output stage circuit 54 in response to the switching signal, and stops the driving of the output stage circuit 54 when the control signal for instructing standby is input.

Abstract: A system for performing loudness normalization based on user feedback is described herein. The system includes a content delivery system and a plurality of audio playback devices. The audio playback devices may communicate volume settings used during playback of pieces of sound program content to the content delivery system. Based on these collected data points, a statistical analysis may be performed to generate loudness adjustment values for pieces of sound program content. The loudness adjustment values may be communicated to the audio playback devices through metadata in associated pieces of sound program content or as separate communications from the content delivery system. An offline version is also described that supports individual loudness normalization adjustments based on a single user's preferences. Under either system, loudness normalization may be achieved based on real world volume settings for individual pieces of sound program content played using various playback configurations.

Abstract: The fitting of a hearing device and in particular of a hearing aid using frequency compression is to be simplified. A method for determining a knee point of a frequency compression characteristic for a hearing device is therefore proposed wherein a maximum audible frequency of a hearing device user is first determined and the knee point is then determined using a predefined rule in dependence on a maximum audible frequency. An end point of a compression characteristic can also be automatically determined.

Abstract: An exemplary system includes a detection facility configured to detect an input sound level of an audio signal presented to an auditory prosthesis patient; and an adaptive gain control (AGC) facility configured to 1) determine whether the detected input sound level is in a quiet region, an intermediate region, or a loud region, and 2) apply a gain to the audio signal in accordance with an AGC gain function that specifies the gain to be substantially equal to or less than a first gain threshold if the detected input sound level is in the quiet region, substantially equal to or less than a second gain threshold if the detected input sound level is in the loud region, and greater than the first and second gain thresholds if the detected input sound level is in the intermediate region. Corresponding systems and methods are also disclosed.

Abstract: A hearing protection earpiece having a shell shaped according to conform to the user's concha and ear canal and having an open hollow lateral portion to be inserted into the user's concha and a medial portion to be inserted into at least an outer portion of the user's ear canal, wherein the medial portion of the shell is axially adjacent to the lateral portion of the shell, wherein the lateral portion of the shell forms part of an open portion of the earpiece and the medial portion of the shell forms part of a closed portion of the earpiece, wherein the earpiece provides for a mechanical sound attenuation of at least 10 dB averaged over the audible frequency range, and wherein the lateral portion of the shell has a wall thickness of not more than 3 mm and is shaped to cover at least half of the concha area.

Abstract: Various embodiments relate to a system for adapting a signal gain, which comprises a peak detector configured to receive an audio input signal containing consecutive signal blocks and to dynamically establish a current input signal level of the audio input signal, and a loudness determination unit configured to dynamically determine a perceived loudness of a current signal block of the audio input signal based on a psycho-acoustic model of a human hearing. Furthermore the system comprises a time constant generation unit configured to determine a time constant based on the perceived loudness of the current signal block and the perceived loudness of a preceding signal block, wherein the time constant describing a change of the signal gain. A gain determination unit is configured to determine the signal gain for the current signal block based on the time constant and the current input signal level.

Abstract: When the instruction for increasing the volume level is input by the user's operation and the signal representing the set value or more is output from a detector, a controller does not increase the volume levels of the first volume controller and the second volume controller. When the signal representing the set value or more is not output from the detector and the second volume level has a predetermined value, the controller increases the first volume level. When the signal representing the set value or more is not output from the detector and the second volume level is less than the predetermined value, the controller increases the second volume level.

Abstract: The present invention provides a single wire programmable Micro Electromechanical System (MEMS) microphone and a programming method and system thereof. The single wire programmable MEMS microphone includes an MEMS sensor and an Application Specific Integrated Circuit (ASIC) chip connected to each other; the MEMS sensor is used for implementing acoustic-electric conversion; the ASIC chip includes an OUT interface, so that an upper computer judges, according to an output signal of the OUT interface, whether the ASIC chip is in a normal start mode or a programming mode, where if the output signal of the OUT interface is at a high level, the ASIC chip is in the normal start mode, and otherwise, the ASIC chip enters the programming mode.

Abstract: A method and system for automatically pausing a remote voicemail recording to answer an incoming call and to notify an incoming caller of a user voicemail interaction. A command to pause the voicemail recording can be determined by the user prior to establishing the voicemail recording. Upon receiving an incoming call the user can be notified of the incoming call prior to terminating the voicemail recording. The command for pausing voicemail recording can be sent to a voicemail system to pause the voicemail recording if the user opts to receive the incoming call. When the call is concluded, the user can be returned to the voicemail system with options such as resume recording, playback, delete, or cancel the message.

Abstract: A method of operating an audio system that provides audio radiation to a plurality of listening positions includes providing at least one source of audio signals. At each listening position, at least one array of speaker elements is provided. A filter is provided between the at least one source and at least one of the speaker elements at a first listening position. The filter is optimized so that the filter reduces acoustic energy radiated from the first array to at least one other listening position of the plurality of listening positions, compared to acoustic energy radiated from the first array to the first listening position.

Abstract: An information processing apparatus including an audio input unit, an amplifier unit, and a controller. The audio input unit is capable of converting audio into an electric audio signal. The amplifier unit is configured to amplify the audio signal obtained by the audio input unit and capable of performing auto gain control. The controller is configured to turn off the auto gain control of the amplifier unit when an application that uses an audio recognition is executed with respect to the amplified audio signal.

Abstract: Method and arrangement in an audio handling entity, for damping of dominant frequencies in a time segment of an audio signal. A time segment of an audio signal is obtained, and an estimate of the spectral density or “spectrum” of the time segment is derived. An approximation of the estimate is derived by smoothing the estimate, and a frequency mask is derived by inverting the approximation. An emphasized damping is assigned to the frequency mask in a predefined frequency range, as compared to the damping outside the predefined frequency range. Frequencies comprised in the audio time segment are then damped based on the frequency mask. The method and arrangement involves no multi-band filtering or selection of attack and release times.

Abstract: Method and arrangement in an audio handling entity, for damping of dominant frequencies in a time segment of an audio signal. A time segment of an audio signal is obtained, and an estimate of the spectral density or “spectrum” of the time segment is derived. An approximation of the estimate is derived by smoothing the estimate, and a frequency mask is derived by inverting the approximation. Frequencies comprised in the audio time segment are then damped based on the frequency mask. The method and arrangement involves no multi-band filtering or selection of attack and release times.

Abstract: A first device receives a sound suppression message from a second device via a communication network. The sound suppression message indicates that the second device will produce an audible sound. The first device determines a sound profile associated with the second device. The first device then modifies (e.g., suppresses or reduces) a volume of sound that that the first device (or a third device) produces in response to receiving the sound suppression request message and based, at least in part, on the sound profile associated with the second device. The sound profile can include a device identifier associated with classes, models, and makes of devices.

Abstract: A method for improving the performance of a noise cancellation device, the method includes determining whether one or more noise making objects (NMO) are near an audible range of the noise cancellation device and receiving a signal from the one or more NMOs indicative of a kind of noise the one or more NMOs is generating. The method also includes selecting a specific noise cancellation model to reduce an expected noise in response to the received kind of noise the one or more NMOs is generating.

Abstract: A system for performing loudness normalization based on user feedback is described herein. The system includes a content delivery system and a plurality of audio playback devices. The audio playback devices may communicate volume settings used during playback of pieces of sound program content to the content delivery system. Based on these collected data points, a statistical analysis may be performed to generate loudness adjustment values for pieces of sound program content. The loudness adjustment values may be communicated to the audio playback devices through metadata in associated pieces of sound program content or as separate communications from the content delivery system. An offline version is also described that supports individual loudness normalization adjustments based on a single user's preferences. Under either system, loudness normalization may be achieved based on real world volume settings for individual pieces of sound program content played using various playback configurations.

Abstract: An audio interruption and buffering playback system includes a primary audio source for reproducing primary audio content and a secondary audio source for reproducing secondary audio content. A processing device detects an interruption event that includes the secondary audio source reproducing secondary audio content having priority over the primary audio source. The processor mutes the output of the primary audio content in response to the interruption event. A buffer buffers the audio content from the primary audio source during the interruption event. The buffered audio content is reproduced by the processor to the user at an accelerated playback speed following the interruption event.

Abstract: According to one embodiment, an audio controlling apparatus includes a first receiver configured to receive audio signal, a second receiver configured to receive environmental sound, a temporary gain calculator configured to calculate temporary gain based on environmental sound received by second receiver, a sound type determination module configured to determine sound type of main component of audio signal received by first receiver, and a gain controller configured to stabilize temporary gain that is calculated by temporary gain calculator and set gain, when it is determined that sound type of main component of audio signal received by first receiver is music.

Abstract: A computer manages methods for managing a volume of an audible transmission by, receiving an audible transmission with an associated location and orientation tag, wherein the associated location and orientation tag contains a geographical origin of the audible transmission and a direction of the audible transmission. Determining a location and orientation of a recipient of the audible transmission. Determining a volume level to be associated with the audible transmission based on the determined location and orientation of the recipient with respect to the associated location and orientation tag of the audible transmission.

Abstract: A computer manages methods for managing a volume of an audible transmission by, receiving an audible transmission with an associated location and orientation tag, wherein the associated location and orientation tag contains a geographical origin of the audible transmission and a direction of the audible transmission. Determining a location and orientation of a recipient of the audible transmission. Determining a volume level to be associated with the audible transmission based on the determined location and orientation of the recipient with respect to the associated location and orientation tag of the audible transmission.

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: A system for controlling a volume of a broadcast advertisement including an advertisement agency server, a broadcaster server, a monitoring apparatus, and an advertisement company server, which are connected through a wire or wireless communication network is provided. The advertisement agency server includes an advertisement material storage unit, an advertisement material conversion unit, an advertisement watermark generation unit, a format conversion unit, and a cost calculation unit.

Abstract: A receiver can extract digital data from wireless radio frequency signal. The digital data contains at least two sets of compressed digital audio data each associated with a different set of audio program. One set of compressed digital audio data is delivered in a first frequency band and the other set in a second frequency band. One of the sets is stored in a memory system and the remaining set is not stored in the memory system. A user can select one of the sets of compressed digital audio data for decompression and then conversion to analog audio signal.

Abstract: A selector selects an analog audio signal input to one input port from among analog audio signals input to multiple input ports according to an instruction from the user. An analog gain control circuit amplifies the analog audio signal received from the selector, with a corresponding one of the gains set for the respective input ports. An analog gain control circuit is configured to gradually change its gain when the gain is switched. An A/D converter converts an output signal of the analog gain control circuit into a digital audio signal. A first audio signal processing circuit is monolithically integrated on a signal semiconductor substrate.

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: 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 volume amplitude limiting device includes a volume adjustment unit, an amplitude limiting unit, and a control unit. The volume adjustment unit adjusts the amplitude of an input sound signal to a larger degree as the adjustment value increases and outputs the resulting sound signal. The amplitude limiting unit limits the amplitude of the sound signal output from the volume adjustment unit to an output limitation value and outputs the resulting signal to a speaker. The control unit sets an adjustment value for the volume adjustment unit and also sets an output limitation value for the amplitude limiting unit according to an adjustment value.

Abstract: An electronic device receives audio from an audio source and outputs the audio via speakers of the device. While outputting the audio via the speakers, the device senses its surrounding environment, and adjusts its operation, based on the sensed environment, to alert a listener wearing the device. The adjustment may comprise generation of one or more audio, visual, and/or vibration notifications to the listener using the device. A volume of the audio output via the speakers may be adjusted based on the sensed surrounding environment. The device may detect whether a first condition is present in the surrounding environment, decrease the volume when the first condition is detected in the surrounding environment, and increase the volume when the first condition is not detected in the surrounding environment.

Abstract: The performance of an echo canceller is assessed using a) a test signal launched from originating test equipment and b) a simulated echo of the test signal launched from terminating test equipment. The launch of the simulated echo signal is timed in such a way that it arrives at the tandem echo canceller(s) at a particular point in time relative to the arrival of the test signal, at the tandem echo canceller(s), when the tandem echo canceller(s) is (are) not able to cancel the simulated echo signal. The latter thus arrives uncanceled at the target echo canceller. The launch of the simulated echo signal is further timed in such a way that it arrives at the target echo canceller at a point in time relative to the arrival of the test signal, at the target echo canceller, when the target echo canceller is able to cancel the simulated echo signal.

Abstract: An audio mixing console comprises a first amplifier, an analogue to digital converter, a second amplifier and a gain range control unit. The first amplifier may receive an analogue audio input signal and amplify the received audio input signal with a first adjustable gain. The analogue to digital converter may receive the amplified analogue audio input signal and generate a digital audio signal by an analogue to digital conversion of the analogue audio signal. The second amplifier may receive the digital audio signal and amplify the received digital audio signal with a second adjustable gain. The gain range control unit may identify a signal level overload at the analogue to digital converter and control the gain of the first and second amplifiers, respectively.

Abstract: A system and method for dynamically mixing audio signals may calculate a signal amplitude for each of two or more audio signals. The signal amplitude may be the absolute value of the audio signal. A signal sum may be calculated using each of the two or more signal amplitudes. Each of the two or more signal amplitudes may be smoothed. The signal sum may be smoothed. The smoothing may be a filter or a leaky integrator. A respective mixing gain may be calculated for each of the two or more audio signals using a respective ratio of each of the two or more smoothed signal amplitudes and the smoothed signal sum. Each of the two or more audio signals may be gain adjusted responsive to the respective mixing gain. Each of the two or more gain adjusted audio signals may be mixed to create an output signal.

Abstract: According to one embodiment, audio controlling apparatus includes first receiver configured to receive audio signal, second receiver configured to receive environmental sound, masking gain calculator configured to calculate masking gain for each frequency band, based on audio signal and environmental sound, and gain controller configured to smooth masking gain of frequency band that is less than preset threshold frequency in first interval, smooth masking gain of frequency band that is equal to or higher than threshold frequency in second interval that is longer than first interval, and thereby set gain.

Abstract: In a method and system for audio level detection and control, an amplitude of an audio signal may be compared to a threshold and an attenuation applied to the audio signal may be adjusted based on the comparison. In instances that the amplitude of the audio signal is greater than or equal to the threshold the adjustment may comprise increasing a first attenuation factor until the amplitude of the audio signal is less than the threshold. The first attenuation factor may be subsequently decreased until the amplitude of the audio signal is greater than or equal to the threshold or until the first attenuation factor is equal to zero. The attenuation of the audio signal may be controlled via a digital gain circuit within the hardware audio CODEC, wherein an overall attenuation factor of the digital gain circuit is a sum of the first attenuation factor and a second attenuation factor.

Abstract: Embodiments are described herein that provide numerous volume enhancements to a media playback system, especially for use by a system that contains two or more zone players that are capable of being grouped to form a synchronous audio playback group.

Abstract: Methods and systems are provided for, volume interactions for connected playback device. In one example, a plurality of playback devices may be rendering audio in synchrony, when a volume adjustment is made for the plurality of playback devices as a group. In one case, a first playback device in the plurality of playback devices may have a volume limit, as indicated by a state variable associated with the first playback device. In this case, the resulting volume adjustment for the first playback device may be made according to both the group volume adjustment and the volume limit of the first playback device.

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: 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.

Abstract: A method of noise-robust speech classification is disclosed. Classification parameters are input to a speech classifier from external components. Internal classification parameters are generated in the speech classifier from at least one of the input parameters. A Normalized Auto-correlation Coefficient Function threshold is set. A parameter analyzer is selected according to a signal environment. A speech mode classification is determined based on a noise estimate of multiple frames of input speech.

Abstract: Embodiments are directed toward user profile based audio adjustment techniques. The techniques are used to render various audio and/or audio/video content having different audio output parameter values in accordance with a user profile that characterizes a user's desired value and/or range of one or more of the output parameter levels.

Abstract: Systems and methods for leveling loudness variation in an audio signal are described. Embodiments use both a perceptual leveling algorithm and a standards-based loudness measure together to minimize audio process artifacts and ensure that the measured loudness of the processed audio is close to a required measure, according to a particular standard measurement of loudness. These systems and methods can be used either offline or in real-time.

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: Sound is outputted, without giving the sense of incongruity to the user, even if reproduction of animation is stopped by the user. An animation acquiring section 11 acquires animation data D1 representing an animation generated in advance based on a setting operation by the user, and sound data D2 representing a sound to be reproduced in association with the animation. A sound output control section 12 calculates stop time sound information representing a feature of the sound at a point of time at which reproduction of the animation is stopped, using sound attribute information D4, in response to input of a stop command by the user, and determines a predetermined output method of the sound that matches the animation, based on the calculated stop time sound information, to reproduce the sound by the determined output method.

Abstract: An audio signal processing apparatus that includes a band division section, an analysis section, a gain adjustment amount calculation section, and a gain adjustment section. The band division section is configured to generate a resonance in-band signal by band division on an audio input signal. The analysis section is configured to extract an amount of features from each of the resonance in-band signal and the input signal. The gain adjustment amount calculation section is configured to calculate a gain adjustment amount for a resonance frequency band in the input signal, the gain adjustment amount being calculated based on the amount of features of the resonance in-band signal and the amount of features of the input signal. The gain adjustment section is configured to perform a gain adjustment on the resonance frequency band in the input signal based on the gain adjustment amount.

Abstract: A headset includes a microphone for receiving a user's voice, a microphone for receiving ambient noise, a receiver for receiving a plurality of voice signals, a speaker for delivering sound to the user's ear and a processing device. The processing device is configured to identify a signal level of a first one of the plurality of voice signals and a second one of the plurality of voice signals, the signal level of the second voice signal being different than the signal level of the first voice signal. The processing device is also configured to measure the ambient noise level and adjust a gain applied to at least one of the first and second voice signals, taking into consideration the ambient noise level. The first and second voice signals are provided to the speaker.

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: 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: An exemplary personal listening device includes thermo-sensitive elements in its transducers. Each thermo-sensitive element senses temperature change of a corresponding transducer and changes a current flowing through the transducer according to temperature change of the transducer, to reflect the amount of time for which the personal listening device has been working.

Abstract: Systems and methods to automatically equalize coloration in speech recordings is provided. In example embodiments, a reference spectral shape based on a reference signal is determined. An estimated spectral shape for an input signal is derived. Using the estimated spectral shape and the reference spectral shape a comparison is performed to determine gain settings. The gain settings comprise a gain value for each filter of a filter system. Using gain values associated with the gain setting, automatic equalization is performed on the input signal.

Abstract: Systems and methods are provided for noise reduction. An input audio signal is received. A target gain corresponding to a target volume level is determined. One or more increments of gain change are determined to reach the target gain. A first non-zero amplitude in the input audio signal is detected. The first non-zero amplitude is not within a predetermined range of zero amplitude. Upon the detection of the first non-zero amplitude in the input audio signal, the one or more increments of gain change are applied at one or more zero-crossing points of the input audio signal. The input audio signal is within the predetermined range of zero amplitude at the one or more zero-crossing points. An output audio signal is generated.

Abstract: A motion picture soundtrack reproduction system has a center front soundtrack channel and a plurality of other soundtrack channels. A volume control adjusts the gain of all the channels. The volume control has a range of settings from a minimum to a maximum, the gain of the center front channel having substantially a first relationship to the volume control settings and the gain of the other channels having substantially a second relationship to the volume control settings, the relationships being such that for a range of volume control settings less than a first setting the gain of the center front channel remains substantially constant while the gain of the other channels decreases as the setting decreases or decreases more gradually than the gain of the other channels as the setting decreases.