Abstract: The invention provides an apparatus capable of automatic volume adjustment. In one embodiment, the apparatus comprises a speaker, an array microphone located in the vicinity of the speaker, a beamforming module, a signal-to-noise ratio calculation module, and a volume adjustment module. The speaker first broadcasts a first audio signal. The array microphone converts a sound into a plurality of second audio signals. The beamforming module derives a speaker sound signal and an ambient noise signal from the second audio signals, wherein the speaker sound signal mainly comprises speaker sound components generated by the speaker and the ambient noise signal mainly comprises noises other than the speaker sound components. The signal-to-noise ratio calculation module calculates a ratio of a first power of the speaker sound signal to a second power of the ambient noise signal to obtain a signal-to-noise ratio.

Abstract: A method, apparatus and computer program product for performing audio ducking is presented. A plurality of tracks are displayed in a user interface, each track of the plurality of tracks including at least one audio clip. A first clip of the plurality of tracks is linked to at least one other clip of the plurality of tracks. At least one audio clip of the first track is preprocessed to obtain characteristic data of the at least one audio clip of the first track. The characteristic data of the at least one audio clip is utilized to determine level change parameters for the at least one clip of the plurality of tracks linked to the first clip.

Abstract: A disclosed system and method dynamically controls the perceived volume of a stereo audio program including left and right channel signals. The system comprises: a dynamic volume control configured and arranged so as to maintain a perceived constant volume level of the stereo audio program; and an excessive spatial processing protection processor configured and arranged for controlling the level of a difference signal, created as a function of the right channel signal subtracted from the left channel signal (L?R), relative to the level of a sum signal, created as a function of the right channel signal plus the left channel signal; wherein the excessive spatial processing protection processor processes the audio signals so as to control the difference (L?R) signal relative to the sum (L+R) signal.

Abstract: A circuit for preventing clipping in an Automatic Level Control (ALC) or Limiter, where the amplitude of the signal above the clipping point is estimated, then the signal level is automatically reduced over a defined period substantially equal to the feedforward delay in the ALC/Limiter. By adaptively controlling, based on the excess amplitude and the delay time available, an attack rate used in the ALC/Limiter to reduce the gain applied to an input signal, it can be ensured that the output amplitude is brought within the clipping level sufficiently quickly to prevent audible clipping.

Abstract: The level of an audio signal can be compressed by receiving an input audio signal, determining a measure of loudness of the signal using a perceptually derived filter, determining a target gain amount, determining a current gain amount using piecewise linear attack/release logic, and producing an output audio signal by adjusting the input audio signal by the current gain amount. An audio compression filter for compressing the level of audio can comprise a loudness measuring module configured to determine a measure of loudness using a perceptually derived filter, and a compression module configured to determine a target gain amount, determine a current gain amount using piecewise linear attack/release logic, and adjust an input audio signal by the current gain amount. Audio level compression can be implemented using integer calculations.

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 information processing apparatus and method include dividing an audio signal, determining a time period having a power change ratio of an audio signal larger than a first threshold value as an attack candidate, searching the time period of the attack candidate and a time period immediately before the time period of the attack candidate for an attack starting point, correcting a power of an audio signal included in the time period, and determining whether a power change ratio of the audio signal included in the time period is larger than a second threshold value for attack detection which is larger than the first threshold value.

Abstract: A method of compensating for spatial audio frequency characteristics that varies in accordance with a mounting condition of a down firing speaker of an audio/video (AV) apparatus includes calculating a listening distance between the AV apparatus and a listener, calculating a distance between a speaker mounted on the AV apparatus and a neighboring reflective surface, setting a spatial frequency compensation filter value and a speaker frequency characteristic compensation filter value based on the calculated distances, and compensating for frequency characteristics of an audio signal by combining the spatial frequency compensation filter value and the speaker frequency characteristic compensation filter value.

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: A method for processing audio signals optimizes the listening experience for hearing impaired listeners, as well as non-hearing impaired listeners, without forcing hearing impaired individuals to feel stigmatized by requiring them to employ special hearing-impaired equipment. A user actuated controller controls a mixture of a preferred audio signal and a remaining audio signal across a range sufficiently wide enough to encompass all individuals. The preferred audio is recorded and maintained separate from all remaining audio and delivered to the listener in a manner that maintains the separateness of the preferred audio and the remaining audio. The user actuated controller includes the capability of automatically maintaining the listener established ratio in the face of changes in the audio signal.

Abstract: An apparatus for controlling a volume of an audio signal and method thereof are disclosed, by which a volume of an input signal can be controlled by using a non-linear gain curve and a target volume of the input signal. The present invention includes storing a non-linear gain curve indicating relations between volumes of an input signal and an output signal; receiving an input signal and target volume information; determining a first gain by using the volume of the input signal and the target volume information; determining a second gain from one of a) the volume of the input signal and the target volume information or b) the first gain, by using the non-linear gain curve; and adjusting the volume of the input signal by applying the second gain to the input signal, wherein the non-linear gain curve is generated by using non-linearity information.

Abstract: The disclosed BTSC encoder includes a left high pass filter means for receiving a digital left channel audio signal and for digitally high pass filtering the digital left channel audio signal and thereby generating a digital left filtered signal; a right high pass filter means for receiving a digital right channel audio signal and for digitally high pass filtering the digital right channel audio signal and thereby generating a digital right filtered signal; a matrix means for receiving the digital left and digital right filtered signals, and including means for summing the digital left and digital right filtered signals and thereby generating a digital sum signal, and including means for subtracting one of the digital left and digital right filtered signals from the other of the digital left and digital right filtered signals and thereby generating a digital difference signal; a difference channel processing means for digitally processing the digital difference signal; and a sum channel processing means for d

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

Abstract: A method (400, 600, 700) and apparatus (220) for enhancing the intelligibility of speech emitted into a noisy environment. After filtering (408) ambient noise with a filter (304) that simulates the physical blocking of noise by a at least a part of a voice communication device (102) a frequency dependent SNR of received voice audio relative to ambient noise is computed (424) on a perceptual (e.g. Bark) frequency scale. Formants are identified (426, 600, 700) and the SNR in bands including certain formants are modified (508, 510) with formant enhancement gain factors in order to improve intelligibility. A set of high pass filter gains (338) is combined (516) with the formant enhancement gains factors yielding combined gains which are clipped (518), scaled (520) according to a total SNR, normalized (526), smoothed across time (530) and frequency (532) and used to reconstruct (532, 534) an audio signal.

Abstract: Systems and methods for amplitude compressing a digital signal. An input signal is divided into frames having a first and second sets of samples. The samples in the second set are also in a subsequent frame. Peak values are determined for the first and second sets. One or more slopes are calculated based on the peak values. The slopes are used to define a scale factor which is applied to the first set to produce the output signal. For example, if the first peak value exceeds an amplitude threshold, first and last samples in the first set to exceed the amplitude threshold are found. Slopes are calculated for each of three regions of the first set demarcated by the first and last samples. In each region a slope is selected. These slopes along with an initial scale factor are used to calculate the scale factor.

Abstract: Audio signal processing apparatus is disclosed. The audio signal processing apparatus includes a first audio signal extracting section, a second audio signal extracting section, a sense-of-depth controlling section, a sense-of-sound-expansion controlling section, a control signal generating section, and a mixing section. The first audio signal extracting section extracts a main audio signal. The second audio signal extracting section extracts a sub audio signal. The sense-of-depth controlling section processes the extracted main audio signal to control a sense of depth. The sense-of-sound-expansion controlling section processes the extracted sub audio signal to vary a sense of sound expansion. The control signal generating section generates a first control signal with which the sense-of-depth controlling section is controlled and a second control signal with which the sense-of-sound-expansion controlling section is controlled.

Abstract: An apparatus configured as a compandor to achieve a defined dynamic range for an output signal in response to an input signal. In particular, the apparatus comprises a first circuit adapted to generate a first signal from the input signal, wherein the first signal includes a first dynamic range (e.g., a first sensitivity and first compression point); and a second circuit adapted to generate a second signal from the input signal, wherein the second signal includes a second dynamic range (e.g., a second sensitivity and second compression point) that is different from the first dynamic range of the first signal. The apparatus may further include a third circuit adapted to generate an output signal related to a sum of the first and second signals. By adjusting the first and second dynamic ranges, an overall dynamic range for the output signal of the companding apparatus may be achieved.

Abstract: An audio output control device includes: display means for displaying two screens; speaker output control means for recognizing whether a main screen or a sub screen is being selected on the display means that is displaying the two screens and performing a control process to output a main or sub sound of the selected main or sub screen through a speaker; and headphone output control means for, before a headphone is connected, performing a control process to supply to a headphone terminal of the headphone the sub or main sound of the sub or main screen that is not being selected.

Abstract: An amplifier device includes a plurality of input terminals for receiving input signals (e.g. audio signals), a main switch, which switches over the input terminals so as to select one input terminal, an analyzer, which analyzes the input signals at respective timings so as to determine whether or not each of the input signals corresponds to a playback signal, a storage for storing a plurality of determination results in connection with the input terminals, a controller for controlling the main switch to select the input terminal whose input signal is newly determined as the playback signal, and an amplifier for amplifying the input signal of the selected input terminal. The analyzer performs frequency analysis on the input signals so as to determine the input signal including a specific frequency component having a high intensity as noise not forming the playback signal.

Abstract: A method for clipping and post-clipping processing an audio signal, includes clipping an audio signal to provide a clipped audio signal; filtering, by a first filter, the audio signal to provide a filtered unclipped audio signal; and filtering, by a second filter, the clipped audio signal to provide a filtered clipped audio signal. The method further includes differentially combining the filtered clipped audio signal and the clipped audio signal to provide a differentially combined audio signal; and combining the filtered unclipped audio signal and the differentially combined audio signal to provide an output signal.

Abstract: Volume control circuits for use in electronic devices are provided. Some embodiments of the present invention provide volume control circuits for use in electronic devices. The volume control circuits include a gain circuit, a limiter circuit, a power measurement circuit and a gain adjustment circuit. The gain circuit is configured to receive an input audio signal and amplify the input audio signal based on a gain. The limiter circuit is configured to limit the amplified input audio signal based on a peak safety limit and output an output audio signal having a peak value based on the peak safety limit. The power measurement circuit is configured to measure a power level of the output audio signal. The power level has an associated power safety limit. The gain adjustment circuit is configured to adjust the gain of the gain circuit responsive to the measured power. Related methods and electronic devices are also provided herein.

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

Abstract: A method and an apparatus for switching speech or audio signals, wherein the method for switching speech or audio signals includes when switching of a speech or audio, weighting a first high frequency band signal of a current frame of speech or audio signal and a second high frequency band signal of the previous M frame of speech or audio signals to obtain a processed first high frequency band signal, where M is greater than or equal to 1, and synthesizing the processed first high frequency band signal and a first low frequency band signal of the current frame of speech or audio signal into a wide frequency band signal. In this way, speech or audio signals with different bandwidths can be smoothly switched, thus improving the quality of audio signals received by a user.

Abstract: The hearing device to be worn by a user of the hearing device comprises an input transducer for transducing incoming acoustical sound into audio signals; a signal processor for processing audio signals; an output transducer for transducing audio signals into outgoing acoustical sound. Said signal processor is designed such that there exists an input SPL range of at least a portion of said incoming acoustical sound, in which an increase in input SPL of said at least one portion of said incoming acoustical sound results in a decrease in output SPL of at least a portion of said outgoing acoustical sound, wherein SPL stands for sound pressure level. Through this, at high input SPL the hearing device's power consumption can be reduced by making use of direct sound, which propagate as sound waves from outside the user's ear canal to the user's ear drum.

Abstract: The method comprises the steps of: filtering the audio signal by means of a lowpass filter (101) with a cutoff frequency substantially equal to said cutoff frequency (F0) of the sound playback device; determining a fundamental frequency for reconstituting from the lowpass filtered audio signal; and generating a harmonic signal (Sharm) associated with said fundamental frequency to be reconstituted. It also comprises the steps of: detecting a time envelope (env(t)) of the lowpass filtered audio signal; adapting the dynamic range of said time envelope (env(t)) as a function of the frequency band under consideration; and reinjecting said harmonic signal in phase into said audio signal by addition after multiplying said harmonic signal (Sharm) with the adapted time envelope (envadapt(t)).

Abstract: Systems and methods for amplitude compressing a digital signal. An input signal is divided into frames having a first and second sets of samples. The samples in the second set are also in a subsequent frame. Peak values are determined for the first and second sets. One or more slopes are calculated based on the peak values. The slopes are used to define a scale factor which is applied to the first set to produce the output signal. For example, if the first peak value exceeds an amplitude threshold, first and last samples in the first set to exceed the amplitude threshold are found. Slopes are calculated for each of three regions of the first set demarcated by the first and last samples. In each region a slope is selected. These slopes along with an initial scale factor are used to calculate the scale factor.

Abstract: An array microphone system includes a first omni-directional microphone, a second omni-directional microphone, a gain control, and a beam former. The first omni-directional microphone faces a first direction. The second omni-directional microphone faces a second direction opposing the first direction. When receiving sound, the first omni-directional microphone and the second omni-directional microphone respectively generate a first signal and a second signal. The gain control amplifies the second signal to transform into a third signal, wherein strength of the third signal is equal to that of the first signal when the sound comes from the first direction. The beam former separates an in-beam sound signal and an out-beam sound signal from the first signal and the third signal.

Abstract: An audio signal in which an audio signal is received as a stream of digital samples, each being a numerical value representing a sampled signal level. A first zero crossing point is identified and the received audio samples are stored until a second zero crossing point is identified, thereby storing a first half-wave of samples. The highest intensity sample is identified from the stored samples and this is compared against a predetermined threshold. All stored samples are scaled by an initial scaling factor so that the intensity of the highest intensity sample is not above this threshold. A second half-wave of samples is stored in which all samples of the second half-wave are below the threshold. All stored samples of the second half-wave are also scaled but by a modified scaling factor derived from a combination of the initial scaling factor and a decay factor.

Abstract: Audio signal processing relating to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal 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. In various embodiments, modification parameters are derived for modifying the audio signal in order to reduce the difference between its specific loudness and a target specific loudness.

Abstract: An audio level compressor for compressing the dynamic range of an input audio signal. The audio level compressor can be part of a microphone, or a separate component. The audio level compressor includes input terminals for receiving the input audio signal, a lamp having a filament resistance that is a function of the input audio signal; a load resistance; a signal gain controller that includes a means for increasing dynamic range compression of the input audio signal, and output terminals for providing an output audio signal that is a function of the input audio signal and the signal gain. The signal gain controller provides a signal gain that is a function of the filament resistance and the load resistance. The audio level compressor can also include an expander circuit or a power conditioning circuit. Power can be provided by an internal source or a phantom power supply.

Abstract: A method and audio device are presented that preserve mono energy during downmixing of a hybrid coding process of an audio signal. The method includes calculating a stereo scaling factor in a group level that is definable within a stereo band. The method may also include updating the stereo scaling factor using an update rate and synchronizing the update rate of a spatial parameter during a fast changing transient portion of the signal. A number of groups in a first stereo band may be greater than a number of groups in a second stereo band, and the first stereo band may be a lower frequency band than the second band or may be perceptually more important than the second band.

Abstract: A system and method control a volume based on location. The system comprises a switch and an asset. The switch is for a network including an operating area for a region. The region is divided into a plurality of zones. The asset is disposed in one of the zones. The asset includes an audio output device. The audio output device is set to a predetermined volume level as a function of one of the plurality of the zones in which the asset is disposed.

Abstract: In a sound processing apparatus, a storage unit stores, for example, a distance range in which sound uttered from a sound source is varied as information showing an area associated with the sound source. A determination unit determines the distance between the sound source and an attended point which is a position in which the sound uttered from the sound source is heard, and determines whether or not the attended point is in the distance range stored in the storage unit. A varying unit varies the sound when the attended point is in the distance range. A reproducing unit reproduces the sound varied by the varying unit.

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

Abstract: This specification describes technologies relating to reducing audio masking. In general, one aspect of the subject matter described in this specification can be embodied in methods that include the actions of receiving a primary audio signal and a secondary audio signal; for each audio signal, calculating an average perceived intensity over time for each of a plurality of frequency bands; comparing the average perceived intensity of the secondary audio signal with the average perceived intensity of the primary audio signal for each frequency band; and for each frequency band where the average perceived intensity of the secondary audio signal is greater than the average perceived intensity of the primary audio signal by a specified threshold amount, attenuating the secondary audio signal by a specified amount to form a modified secondary audio signal.

Abstract: A system and method for improved audio controls on a personal computer is provided. The system and method provide a unified architecture for audio controls across hardware and software interfaces of the personal computer. An intelligent facility may automatically change audio controls for users to simply interact with various communications and media applications. To this end, a configurable audio controller intelligently handles various aspects of the system's audio devices by following various rules that may be based at least in part on user-configurable settings and a current operating state. The present invention also provides audio controls so that a user may easily change audio settings such as the volume of an audio output device. There are many applications that may use the present invention for automatic control of audio devices based upon the user's context.

Abstract: The present invention relates to a process for adjusting the sound volume of a digital sound recording characterized in that it comprises: a step consisting of determining, in absolute values, for a recording, the maximum amplitude values for sound frequencies audible for the human ear, a step consisting of calculating the possible gain for a specified sound level setting, between the maximum amplitude value determined above and the maximum amplitude value for all frequencies combined, a step consisting of reproducing the recording with a sound card by automatically adjusting the amplification gain level making it possible to obtain a sound level for the recording of a specified value so that it corresponds to the gain calculated for this recording.

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

Abstract: When controlling the audio signal to the mute mode from an unmute mode, first, the muting circuit switches from the non-attenuating state to the attenuating state to cause the muting circuit to attenuate the audio signal by the predetermined amount of attenuation, and then, the amount of attenuation by the volume control section is set to the maximum amount of attenuation to cause the volume control section to attenuate the audio signal by the maximum amount of attenuation.

Abstract: A post phase-inverter master volume regulator for use in combination with a tube-type musical amplifier having a phase-inverter tube. The signals at both sides of the phase-inverter tube's DC voltage plate are extracted from the amplifier for processing by the master volume regulator. In one embodiment, such signals are extracted by inserting a male/female tube base between the amplifier's phase-inverter tube and its socket. The two voltage plate signals are then transmitted through a multiconductor umbilical to an exterior enclosure. Each signal is connected to ground through its respective user adjustable rheostat, permitting the user to simultaneously change the inverter-tube's two voltage signals (and vary the amplifier's volume). The regulator may also include circuitry adapted to provide a bass-boost, and/or a high frequency cutoff. Sometimes, first and second volume controls may be provided to permit a user to quickly change between two volume settings.

Abstract: An audio amplifier includes a plurality of sub-audio amplifier sets, a clock source, and a speaker. When the audio amplifier is under an audio amplify mode, the plurality of sub-audio amplifier sets is operated under a synchronic operating frequency with a same clock signal generated by the clock source. When the audio amplifier is under a self-oscillation mode, the clock signal is isolated from being input to the plurality of sub-audio amplifier sets. A higher distortion is prevented by repeatedly charging and discharging an oscillating capacitor of a sub-audio amplifier in a sub-audio amplifier set.

Abstract: There is provided an automatic gain control circuit for volume control receiving at least one audio signal input via either a wired connection or a wireless connection. The circuit includes an output signal regulator receiving a first portion of the at least one audio signal input, with the regulator being able to vary an amplitude of the first portion of the at least one audio signal input when a control signal from a circuit controller is received. The circuit controller may be powered by a DC generator; with the DC generator receiving a second portion of the at least one audio signal input. A signal from the output signal regulator may be passed through an acoustic generator. It is preferable that the first portion of the at least one audio signal input is substantially more than the second portion. It is also preferable that the output signal regulator varies the amplitude of the at least one audio signal input by reducing the amplitude. A corresponding method of volume control is also disclosed.

Abstract: A system and method for improved audio controls on a personal computer is provided. The system and method provide a unified architecture for audio controls across hardware and software interfaces of the personal computer. An intelligent facility may automatically change audio controls for users to simply interact with various communications and media applications. To this end, a configurable audio controller intelligently handles various aspects of the system's audio devices by following various rules that may be based at least in part on user-configurable settings and a current operating state. The present invention also provides audio controls so that a user may easily change audio settings such as the volume of an audio output device. There are many applications that may use the present invention for automatic control of audio devices based upon the user's context.

Abstract: A control method to automatically adjust the volume of a sound transmitter based on the measurement and the computation of the acoustic statistics of the room where the sound is emitted. The system comprises at least one sensor, a calculator determining statistics of the signal collected by the at least one sensor and a controller using these statistics provided by the calculator to adjust the volume of the transmitter in the room.

Abstract: A speaker device comprises a first speaker (SP1) for reproducing an audio signal and a second speaker (SP2) adapted for reproducing an audio signal and spaced from the first speaker horizontally by a predetermined distance. At least one of the first and second speakers phase has varying means (APF1, APF2, . . . ) for varying the phase of the audio signal by a predetermined quantity of phase according to (i) the frequency of the audio signal and (ii) the predetermined distance.

Abstract: An automatic gain control system maintains desired signal content level, such as voice, in an output signal. The system includes automatic gain control over an input signal, and compensates the output signal based on input signal content. When the input signal level exceeds an upper or lower processing threshold level, or is distorted (e.g., clipped), the system applies a gain to the input signal level. The system may compensate for the gain in the output signal when the input signal includes desired signal content.

Abstract: A multi-channel audio playback apparatus including a channel interface, a first switching amplifier and a second switching amplifier is provided. The channel interface is used to receive multi-channel digital data and generate first channel digital data and second channel digital data. The first switching amplifier is used to convert the first channel digital data into a first pulse width modulation (PWM) signal according to a first reference signal with a first frequency. The second switching amplifier is used to convert the second channel digital data into a second PWM signal according to a second reference signal with a second frequency. The second frequency is different from the first frequency.

Abstract: An electronic device includes an amplifier, an output unit, a processing unit, and a mute control circuit. The amplifier is for amplifying audio signals and generating amplified audio signals accordingly. The amplifier includes an output port to output the amplified audio signals. The output unit is for emitting sounds corresponding to the amplified audio signals. The processing unit is for generating a mute control signal. The mute control circuit includes a main circuit and an auxiliary circuit. The main circuit is for responding to the mute control signal, and for driving the amplifier to change between a mute state and a play state accordingly. The auxiliary circuit is for enabling the output port of the amplifier to be electrically grounded at the moment of supplying power to the electronic device.

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: An audio-signal time-axis expansion/compression method for subjecting an audio signal to time-axis expansion/compression at a time domain includes the steps of: cross-fade-signal generating wherein a first period and a second period which are similar within the audio signal are employed to generate the cross-fade signal of the first period signal and the second period signal; correction-signal generating wherein the difference signal between the first period signal and the second period signal is subjected to time-axis reversal, and is multiplied with a window function to generate a correction signal; and connection-waveform generating wherein the cross-fade signal and the correction signal are added to generate a connection waveform for subjecting the audio signal to time-axis expansion/compression at the time domain.