Abstract: This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor ? of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.

Abstract: The present invention provides methods and systems for digitally processing audio signals in broadcasting and/or transmission applications. In particular, the present invention includes a pre-transmission processing module which is structured and configured to generate a partially processed signal. A transmitter is then structured to transmit or broadcast the partially processed signal to a receiver, where the signal is then fed to a post-transmission processing module. The post-transmission processing module is structured and configured to further processes the signal based upon, for example, the listening environment, profile(s), etc. and generate a final output signal.

Abstract: A device for audio mixing includes a cascade input port and a first cascade input component that includes a first cascade input level detector component that detects an audio level of an upstream sum audio signal. The first cascade input port also includes an attenuator component that attenuates the first upstream mix audio signal by a gain corresponding to difference between the upstream sum audio signal value and a detected audio level of an input sum audio signal. The device also includes a summer component where the signals summed include at least the first upstream sum audio signal. The device also includes an input sum level detector component that detects the audio level of the input sum audio signal. The device also includes a mixer component that is configured to provide a mix output signal by summing, where the signals summed include at least the first attenuated upstream mix signal.

Abstract: A system can be provided for increasing loudness of an audio signal to present a perceived loudness to a listener that is greater than a loudness provided natively by a loudspeaker. The system can include one or more of the following: a frequency suppressor, a loudness adjuster, an equalizer, and a distortion control module. The frequency suppressor can increase headroom in the audio signal by filtering out low and/or high frequencies. The loudness adjuster can calculate a loudness of the audio signal and apply a gain to the audio signal to increase the loudness. The equalizer can further increase headroom by attenuating portions of a passband of the loudspeaker's frequency response. The distortion control module can induce partial harmonic distortion in the audio signal to further increase loudness.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; receiving user gain input; generating a linear gain factor and a non-linear gain factor using the user gain input; modifying the non-linear gain factor using absolute threshold of hearing and power of the input signal to generate a modified non-linear gain factor; and, applying the linear gain factor and the modified non-linear gain factor to the audio signal.

Abstract: An audio volume control circuit includes a signal intensity calculating circuit for generating a first signal intensity value corresponding to a signal intensity corresponding to an audio channel data; a low-pass filter for filtering the first signal intensity to generate a second signal intensity value; an averaging unit for averaging the second signal intensity value and previous M?1 second signal intensity values to obtain a third signal intensity value, with M being a natural number greater than 1; a gain calculating circuit for obtaining an original gain value according to the third signal intensity value with reference to the adjustment condition; a buffer for temporarily storing the audio channel data; and an audio volume adjusting circuit for generating an adjustment gain value according to the original gain value to adjust the audio channel data stored in the buffer.

Abstract: An exemplary audio processing system includes a gain control unit, a sampling unit, and a triggering unit. The gain control unit is configured for amplifying an audio signal. The sampling unit is configured for sampling the audio signal. The triggering unit is configured for generating a gain reduction unit if the amplitude exceeds a predetermined value over a predetermined time period. The first predetermined value is set so that if the amplitude of the audio signal exceeds the predetermined value and the gain of the gain control unit is not reduced, the amplitude of the amplified audio signal exceeds a predetermined acceptable range. The gain control unit is also configured for reducing the gain of the gain control unit responding to the gain reduction signal to limit the amplitude of the amplified audio signal within the predetermined acceptable range.

Abstract: In the specification and drawing a control volume system is described and shown for synchronizing a sound volume of an electronic device and another sound volume of the peripheral device according to a volume specification of the electronic device and another volume specification of the peripheral device. Moreover, a control volume method and the electronic device are also disclosed in the specification and drawing.

Abstract: A volume control apparatus that is connectable to a controller and controls a volume value of a sound signal based on an instruction from the controller includes a receiving section for, when a controller-side volume setting value settable in the controller is changed by a user's operation, receiving the changed controller-side volume setting value from the controller. The apparatus also includes a first difference value calculating section for calculating a first difference value between the changed controller-side volume setting value and the controller-side volume setting value before change, a second difference value calculating section for calculating a second difference value between a maximum volume value controllable by the controller and a current volume value, and an increase value calculating section for multiplying the second difference value by a ratio of the first difference value so as to calculate an increase value of the volume value.

Abstract: A gain control circuit includes a comparator that compares an input gain value with a count value to generate a comparison result signal, a counter that counts up or counts down the count value in accordance with the comparison result signal, and a gain modulator circuit that modulates the count value to generate a gain control signal which changes in a time-divided manner. The gain modulator circuit modulates the count value so that a gain obtained by time-averaging a gain corresponding to the gain control signal matches a gain based on the count value.

Abstract: Information useful for modifying the dynamics of an audio signal is derived from one or more devices or processes operating at one or more respective nodes of each of a plurality of hierarchy levels, each hierarchical level having one or more nodes, in which the one or more devices or processes operating at each hierarchical level takes a measure of one or more characteristics of the audio signal such that the one or more devices or processes operating at each successively lower hierarchical level takes a measure of one or more characteristics of progressively smaller subdivisions of the audio signal.

Abstract: An apparatus to perform sound processing in a virtual reality system includes a sound processing unit to process and produce sound output in sound areas divided into a focus area within a predetermined visual field and a non-focus area out of the predetermined visual field in a virtual reality space for sound sources; and a control unit to divide the sound areas into the focus area and the non-focus area, and to control the sound output of the sound processing unit such that a volume of sound in a first space is gradually decreased while a volume of sound in a second space is gradually increased when a space shift from the first space to the second space in the virtual reality space is detected.

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: A system includes a plurality of inputs each configured to receive a filtered version of a source signal. The system extracts the energy information from each input signal and compares the energy information of a plurality of input signals. Alternatively, the system extracts energy information from a signal that is the difference of two input signals. Based on the energy information, the system determines at least one parameter that may be changed in at least one circuit in a plurality of circuits to minimize the differences in energy of the input signals or to minimize the energy of the difference signal. Parameters may include for example amplification, delay, and corner frequency values. The set of circuits may include microphone interface circuits. Merely by way of example, a system with microphone interface circuits may be included in a hearing enhancement device or in a hands-free earpiece.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; estimating indicator function using a signal power of the input signal; obtaining an adapted filter using the indicator function and an equalization filter; and, generating an output signal by applying the adapted filter to the input signal.

Abstract: A peripheral apparatus, for connecting to a first or a second transmission port of an electronic device, includes a signal transmission interface, a triggering unit, an amplifying unit and an adjusting unit. The signal transmission interface, coupled to the first or the second transmission port, includes a first and a second group of pins. The triggering unit is coupled to the signal transmission interface. When the signal transmission interface is coupled to the first or the second transmission port, the triggering unit outputs a first or a second voltage level signal, respectively. The amplifying unit is coupled to the signal transmission interface and a functional unit. The adjusting unit, coupled to the signal transmission interface, the triggering unit, the amplifying unit and the functional unit, changes an output power transmitted from the amplifying unit to the functional unit according to the first or the second voltage level signal.

Abstract: A method includes comparing a magnitude value of an audio signal in a fixed unit section with a target volume level, measuring audio gain in each fixed unit section using specific functions which are each determined to be different according to a ratio of the target volume level to the audio signal magnitude, and providing the measured audio gain to the audio signal in the fixed unit section.

Abstract: Information on a use situation of a portable electronic apparatus possessed by a user is acquired from an illuminance sensor, a contact sensor, and a GPS portion as use situation-acquiring portions, and a volume of a sound outputted either to a speaker provided as an internal sound-outputting unit in the portable electronic apparatus or to an external sound-outputting unit such as a headphone is controlled in accordance with the acquired information. Thus, even when a plug of the headphone falls off from the portable electronic apparatus despite an intention of the user, an unpleasant sound is prevented from being outputted from the speaker to a circumference.

Abstract: A circuit configured to adjust the level of an audio signal includes filters each of which is configured to receive an audio signal, and to pass a band set for the filter. Variable gain amplifiers are severally provided to the respective filters, and each variable gain amplifier amplifies the output signal of the corresponding filter. An adder sums the input audio signal and the output signals of the variable gain amplifiers. A control unit controls the level of the output signal of each of the variable gain amplifiers, and controls the gain of each variable gain amplifier according to the level thus monitored.

Abstract: Two or more acoustic transducers share the same acoustic port in a device. The acoustic properties—such as acoustic impedance and frequency response—of the shared acoustic port are matched to each of the two or more acoustic transducers. To accomplish acoustic impedance matching, a separate back volume is provided for each of the acoustic transducers, matched to that transducer. Frequency response matching can be accomplished by the design of the transducer itself, but also by providing an adjacent element in the acoustic system of the transducer. One transducer may serve as an element in the acoustic system of another transducer. Frequency response adjustment of an individual element may also affect acoustic impedance of the entire port-transducer system.

Abstract: Embodiments relate to apparatuses for, and methods of, compensating for distance 206 and velocity present between a microphone and user's 202 mouth. Such devices and methods allow compensation for varying amplitudes of sound pressure received at a the due to a varying distance 206 between a microphone and an originating user's 202 mouth. Additionally, the devices and methods may also compensate for pitch shifts due to a Doppler effect caused by a non-zero velocity of a microphone relative to a user's 202 mouth.

Abstract: Embodiments relate to systems for, and methods of, compensating for movement of a loudspeaker relative to a user's head, where the loudspeaker is present in a mobile device. Example systems and methods produce (300) an electrical signal representative of audio, determine (302) a distance between the device and the user's head and automatically set (304) a gain of the electrical signal in accordance with the distance. Example systems and methods also output (306) audio corresponding to the electrical signal with the gain.

Abstract: Systems and method for adapting a gain of an audio signal containing at least two different tracks with different signal level ranges. An example system includes a loudness determination unit for determining a perceived loudness of the audio input signal based on a psychoacoustic model of a human hearing. The system further includes a loudness adaptation unit configured to determine a loudness adaptation indicator based on the perceived loudness and a predetermined loudness, and to introduce the loudness adaptation indicator into the audio signal to generate an adapted gain signal. A gain determination unit is configured to adapt the gain for outputting the audio signal containing the at least two tracks based on the loudness adaptation indicators in the adapted gain signal at about the predetermined loudness.

Abstract: An apparatus comprises a selected volume detector that detects a selected output volume; an analog output signal amplifier; a digital volume amplifier; a boost gain control element coupled to the selected volume detector; the analog output signal amplifier; and the digital volume amplifier; wherein the boost gain control element is configured to: keep a gain of a path of the digital volume amplifier and the analog output signal amplifier substantially constant, wherein the boost gain control element can adjust both: a) a gain of the digital volume control; and b) a gain of the analog output signal amplifier; to keep the gain of the path of the digital volume amplifier and the analog output signal substantially constant and equal to the selected output volume.

Abstract: An audio adjustment device including a plurality of filters respectively amplifying or attenuating signal components of a multiplicity of frequency bands among input audio components, an adjustment part that adjusts gain characteristics of the plurality of filters, and a characteristic display part that displays gain characteristics adjusted by the adjustment part. The adjustment part is capable of independently adjusting gain in signal components of central frequencies of signal components subject to amplification or attenuation by the filters, and Q value of gain characteristics of the filters. The characteristic display part displays the gain characteristics of the plurality of filters in a row along one frequency axis.

Abstract: Compensating for varying degrees of ambient noise in a mobile device. One possible noise compensation application has a plurality of selectable settings associated with a range of speeds of motion. This noise compensation application, when executed by the processor, configures the control circuit to: receive an input from the detector, detect an amplitude of the input, compare the amplitude with a selected one of the settings, and selectively adjust an amplification device when the detected amplitude is within a prescribed range. Another possible noise compensation application has plurality of selectable settings each associated with a range of audio frequencies. This noise compensation application, when executed by the processor, configures the control circuit to: receive an input from the detector, detect a frequency of the input, compare the frequency with a selected one of the settings, and selectively filter the ambient noise when the detected frequency is within a prescribed range.

Abstract: A receiver can extract digital data from wireless radio frequency signal. The digital data contains at least two compressed digital audio data each associated with a different audio program. One compressed digital audio data is delivered in a first frequency band and the other compressed digital audio data in a second frequency band. A user can select the audio program. The receiver also receives a command instructing the receiver to receive digital audio data from a communication channel in a frequency band.

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: A method and apparatus for limiting the absolute magnitude of an audio signal. The method may include firstly variable-gain reducing the gain of an audio signal, and then secondly variable-gain reducing the gain of the audio signal faster than the first variable-gain reduction, thereby limiting the absolute magnitude of the audio signal to a threshold. The first variable-gain reduction may include variable-gain reducing the gain of the audio signal in a first stage, and the second variable-gain reduction may include variable-gain reducing the gain of the audio signal in a second stage that reduces the gain faster than the first stage. The second variable-gain reduction may include delaying the audio signal, finding a peak among the delayed audio signal, calculating a fast gain from a found peak, and modifying the delayed audio signal with the calculated fast gain.

Abstract: A system and method for automatically limiting or otherwise controlling the volume level within audio devices is provided. A sound level controlled amplification system is configured to sample audio signals for an audio device, and if above a predetermined limit, de-amplify the volume level to a safer level of sound, and control and maintain such safe level until the sample audio signal is within the predetermined limit of safe sound.

Abstract: An acoustic processing apparatus is provided. The apparatus includes a pre-processing component, a filter and a first signal processing component. The pre-processing component compensates a non-linearity of a reference signal to generate an input signal. The filter coupled to the pre-processing component, the filter executes filtering on the input signal to generate an output signal. The first signal processing component, coupled to the pre-processing component, the reference signal obtains a gain from the first signal processing component to generate a first signal, and the first signal processing component passes the gain to the pre-processing component.

Abstract: In general, techniques of controlling a plurality of multimedia players in groups are disclosed. According to one aspect of the present invention, a mechanism is provided to allow a user to group some of the players according to a theme or scene, where each of the players is located in a zone. When the scene is activated, the players in the scene react in a synchronized manner. For example, the players in the scene are all caused to play a multimedia source or music in a playlist, wherein the multimedia source may be located anywhere on a network.

Abstract: Systems and methods for managing the volume of multiple VoIP streams are disclosed. The system includes a VoIP server configured to receive an input audio stream from a first VoIP handset, create separate output audio streams from the input audio stream for transmission to second and third VoIP handsets, and to connect to a communications network. The system also includes a volume control table coupled to the VoIP server, the volume control table including records of volume adjustments made during prior conversations between the two or more VoIP handsets. The VoIP server is further configured create the separate output audio streams such that one or more of the output streams has a volume that is different than input audio stream based on the records.

Abstract: A control method for a voice control system applied to an electronic apparatus is provided. The method includes: launching a predetermined audio receiving period and lowering an output volume of the electronic apparatus when determining to receive an audio signal for controlling the electronic apparatus; receiving the audio signal and executing a corresponding command; and restoring the output volume of the electronic apparatus when the predetermined audio receiving period ends. The audio signal is for controlling the electronic apparatus.

Abstract: The invention relates to a sound system in a vehicle. The sound system comprises a first audio signal source providing a first audio signal and a second audio signal source providing a second audio signal. The sound system further comprises loudness determination unit is configured for determining a loudness of the first audio signal based on a psycho-acoustic model of a human hearing and a type information unit configured for establishing type information of the second audio signal. The sound system further comprises a control unit configured for individually selecting, for each of at least two playback zones, a mixing mode depending on the type of the second audio signal, wherein the mixing modes relate to a relation between signal output gains of the first and second audio signals.

Abstract: A program asset is retrieved. The program asset has audio encoded at a first loudness setting and includes metadata specifying the first loudness setting. Dialog of the audio is identified. The loudness of the dialog is determined. The determined loudness is compared to the first loudness setting. The program asset is re-recorded at a second loudness setting corresponding to the determined loudness, if the first loudness setting and the determined loudness are different by more than a predetermined amount.

Abstract: A case for a handheld computing device is disclosed. The case includes a housing sized and dimensioned to receive a handheld computing device securely therein. A data port is configured to electrically connect to the data connector on the handheld computing device. A microprocessor is electrically connected to the data port. The microprocessor is configured and arranged to authorize audio stored on the handheld computing device to play through the data connector and connected data port. Audio filters are electrically connected to the data port. An audio output electrically connector to the audio filters.

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: A volume adjustment circuit includes an amplification circuit, a control circuit and an adjustment circuit. The amplification circuit amplifies a first audio signal received and generates a second audio signal, and outputs the second audio signal to an audio output terminal. The control circuit is electrically coupled to the audio output terminal, and is configured to generate a control signal based on the second audio signal. The adjustment circuit is electrically coupled to the control circuit, and configured to adjust an amplitude of the first audio signal inputted to the amplification circuit based on the control signal.

Abstract: A sound output device of the present invention is one that receives a digital audio signal including a plurality of channels and performs digital signal processing on the received digital audio signal, and includes a detecting portion which detects peak values or effective values of the plurality of channels included in the digital audio signal. The sound output device also includes an electric power calculating portion which calculates necessary electric power to output sound of each of the channels according to the peak values or the effective values. The sound output device also includes a sound volume adjusting portion which adjusts sound volume of the digital audio signal according to a total of the necessary electric power calculated by the electric power calculating portion for each of the channels.

Abstract: A method (100) for matching characteristics of two or more transducer systems (202, 208). The method involving: receiving input signals from a set of said transducer systems; determining if the input signals contain a pre-defined portion of a common signal which is the same at all of said transducer systems; and balancing the characteristics of the transducer systems when it is determined that the input signals contain the pre-determined portion of the common signal.

Abstract: Embodiments of the present invention provide systems, methods, and computer-readable media for presenting a mute indicator to a caller when the caller is attempting to provide input to a call that has been muted. In particular, a determination is made that a mute function is engaged during a call. An area surrounding a calling system, such as a phone, is then monitored by a monitoring system to determine whether a caller is attempting to provide input to the call. A determination is then made that a caller is attempting to provide input to the call. The determination may be based on a caller providing a threshold volume of voice input within a threshold area of proximity near the calling system. A mute indicator is then presented to the caller to notify the caller that the call has a mute function engaged.

Abstract: An audio compressor may regulate the level of an audio input signal depending on whether the level is above or below a threshold value. The audio compressor may control a pumping that is created when regulating a dynamic range of an audio signal with the threshold value. A feedback loop connecting the signal output from the audio compressor with a release filter may be used to modify an effective release time of the signal. A controller may be used that allows a filter coefficient of the release filter to be controlled to adjust the effective release time as a function of the signal output.

Abstract: Disclosed is an auto volume control method and system for mixing of sound sources. The control method and system mixes the sound sources by controlling signals of the sound sources automatically inputted by a simple control method, such that the signals from a plurality of sound sources are mixed to prevent an overflow phenomenon of the sound sources from occurring.

Abstract: A method and an apparatus for processing audio data that controls pressure of a played sound to improve a quality of audio are provided. The method for processing audio in a mobile terminal includes: receiving the audio from a microphone; calculating by a controller a sound pressure level of the received audio; determining a sound pressure state of the received audio based on the calculated sound pressure level; and controlling an input gain of the audio based on the determination result.

Abstract: A volume control system, that can easily check current volume values with respect to volume values settable in a plurality of audio output devices when a volume value representing the volume values of the audio output devices is displayed, is provided. A controller of the volume control system can be connected to the plurality of the audio output devices, and can control the audio output devices. The controller obtains the volume values of the audio output devices and predetermined upper limit volume values stored in the audio output devices, and displays percentages of the volume values with respect to the predetermined upper limit volume values as total volume values. Further, the controller displays values that are determined by multiplying the total volume values and percentages of the predetermined upper limit volume values with respect to prescribed values as the volume values of the audio output devices.

Abstract: A hearing aid (101) comprises a microphone (121) converting sound into an electrical signal, a processor (105) disposed inside a main body case (102), a first volume adjuster (104) having a variable resistor, and a receiver (123) converting an electrical signal into sound. The processor (105) has an amplifier (135) amplifying an electrical signal from the microphone (121), a second volume adjuster (105b) setting a degree of amplification by the amplifier (135), and a controller (130) controlling the second volume adjuster (105b) and the amplifier (135). The controller (130) detects the resistance value of the variable resistor, determines whether or not the resistance value has exceeded a specific threshold, acquires a specific value for the degree of amplification by the amplifier (135), and sets the degree of amplification by the amplifier (135) to this specific value when the controller determines that the resistance value has exceeded the specific threshold.

Abstract: Disclosed herein is a recording apparatus including: an audio signal correction block configured to execute correction for flattening the frequency characteristic of an audio signal supplied from a microphone and/or correction of the level of the audio signal; a correction control block configured to make the audio signal correction block adjust the level of the audio signal attenuate the reference value with time, and, if an audio signal with a level thereof exceeding the reference value is entered, use the absolute value of the level of the audio signal exceeding the reference value as a new reference value; and a recording block configured to record the audio signal to a recording media.

Abstract: A sound processor includes a conversion unit converts a reference sound signal corresponding to a base of sound to be output and an observation sound signal based on each of sound signals output by a plurality of sound receiving units into frequency components, an echo suppression unit estimates echo derived from sound based on a converted reference sound signal and suppressing the estimated echo in a converted observation sound signal, a noise suppression unit estimates noise based on an arrival direction of sound and suppressing the estimated noise in the converted observation sound signal and an integrating process unit suppresses, with respect to each frequency component, echo and noise in the converted sound signal based on a observation sound signal obtained after echo suppression and a observation sound signal obtained after noise suppression.

Abstract: A signal processing platform (300) presents (101) a signal to be processed and identifies (102) signal portions with specific characteristics that are used (103) to automatically determine at least one bounding frequency that can be used to facilitate bandwidth extension for the signal. Identifying these signal portions can comprise identifying signal portions that exhibit at least a predetermined level of energy. The step of determining the bounding frequency can comprise computing a magnitude spectrum for each of the identified signal portions that can be used to determine a corresponding measure of flatness within a pass band as pertains to a corresponding normalized signal portion to thereby provide corresponding vetted signal portions. Determining the bounding frequency can then comprise accumulating the magnitude spectrum for these vetted signal portions and using the resultant accumulation to estimate a corresponding signal envelope.