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: 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 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 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 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.

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 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: An apparatus comprising at least one processor and at least one memory including computer program code. The at least one memory and the computer program code is configured to, with the at least one processor, cause the apparatus at least to determine a loudness estimate of a first audio signal, generate a parameter dependent on the loudness estimate; and control the first audio signal dependent on the parameter.

Abstract: A method for controlling altitude information-based user functions and a mobile device adapted thereto are provided. The mobile device includes a storage unit for storing audio data that will be output according to the activation of a user function, and auxiliary data that will be applied to the audio data if an audio data altering function is set when the user function is activated, a sensing unit for collecting altitude information to set auxiliary data that will be applied to the audio data according to the activation of the user function, a controller for creating altered audio data by applying predefined auxiliary data to audio data that will be output when the user function is executed, according to the altitude information, and an audio processing unit for outputting the altered audio data.

Abstract: According to one embodiment, there is provided a sound quality control apparatus, including: a characteristic parameter extractor; a speech score calculator; a music score calculator; a power value acquisition module; a first storage configured to store speech scores and music scores; a second storage configured to store power values; a power-based score corrector configured to correct a current music score or a current speech score based on a first comparison result between a current power value and past power values, a second comparison result between the current music score and past music scores and a third comparison result between the current speech score and past speech scores; and a sound quality controller configured to perform a sound quality control by using at least one of the speech score and the music score corrected by the power-based score corrector.

Abstract: According to at least one embodiment, a volume controller includes an audio processor configured to generate an output signal by variably controlling an amplitude of an input signal; and a volume controller configured to set a sound volume for the variable control based on the input signal.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The apparatus comprises an information receiving unit receiving a downmix signal including and a plurality of preset informations; an external preset information receiving unit receiving a plurality of external preset informations being inputted from external; an external preset applying-determining unit determining whether the plurality of the external preset informations applies to the downmix signal; an external preset information selecting unit selecting one external preset information, if the plurality of external preset informations is selected; and a rendering unit controlling the object by applying the external preset information to the all data regions. Accordingly, an audio signal can be efficiently reconstructed by individually selecting and applying external preset information by a data region unit or by selecting and applying the same external preset information to a whole downmix signal.

Abstract: A dynamic surround channel volume control applies a variable gain control to the surround channels of a multi-channel system. Thus the levels of the surround channels are related to the main volume control of the system. This maintains a predictable relationship between the surround channels and the main or front channels.

Abstract: System and method for detecting a glitch is disclosed. An embodiment comprises increasing a bias voltage of a first capacitor, sampling an input signal of a first plate of the first capacitor with a time period, mixing the input signal with the sampled input signal, and comparing the mixed signal with a reference signal.

Abstract: A method and system is presented for sampling analog signals in a manner that avoids the effects of signal clipping due to a limited dynamic range. A method and device for sampling an analog input using multiple gains, or gain mask, is described. By using different gains during different time quanta, a subset of the sampled points may effectively be attenuated before being sampled and converted to a digital representation. If clipping occurs during the sampling process, the true values of the clipped samples may be interpolated using the amplitudes of the non-clipped samples, which may not have been attenuated. Such interpolation may include constructing and/or solving a constraint optimization problem using linear programming. In one embodiment, such a problem may be constructed and/or solved by using sign information from the clipped samples and/or by imposing a sparsity assumption on the signals during the reconstruction process.

Abstract: An electricity-to-sound converter reproduces a sound according to a first audio signal. A sound collector outputs a second audio signal. A first frequency band limit filter filters the second audio signal. A sound leakage evaluating block evaluates a leakage of a sound reproduced by the electricity-to-sound converter on the basis of the first audio signal and the filtered second audio signal. A multiplier multiples the first audio signal by a coefficient approximating a transmission characteristic between the electricity-to-sound converter and the sound collector when the sound reproduced by the electricity-to-sound converter leaks outside a housing to be picked up by the sound collector. A second frequency band limit filter filters the multiplied first audio signal and supplies the filtered first audio signal to the sound leakage evaluating block. The first frequency band limit filter and the second frequency band limit filter have passing bands of 1 kHz to 3 kHz.

Abstract: Disclosed is a decoding device which can efficiently encode/decode spectral data in a high pass section of a broadband signal. In the disclosed device: a sample group extraction unit (372) partially selects spectral components by means of an ease of selection importance which is the extent that the spectral components come close to the spectral component having the maximum amplitude value, in the spectrum of a high pass estimated by means of first parameters contained in second encoded information and bands most approximated to each of the spectrums of a plurality of sub-bands calculated from the spectrum of a second decode signal; a logarithmic gain application unit (373) applies second parameters to the partially selected spectral components; and an interpolation processing unit (374) applies third parameters which are adaptively set according to the value of the second parameters, to the spectral components which were not partially selected.

Abstract: The present invention relates a method and a mobile communication device for generating an adaptive ring signal level in the mobile communication device in response to the surrounding environmental condition of said mobile communication device and also in response to the way the mobile communication device is stored away. By comparing a transfer that is determined based on the relation between an emitted signal from the loudspeaker and a response recorded by the microphone and an ideal transfer function a calculated response function is obtained. The calculated response function is compared with predefined response functions stored in a look up table to find a match that is then selected. The ring signal is thereafter adjusted to a level that corresponds to the values associated with and stored together with the selected predefined response function.

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: An audio signal processing device capable of separating audio signal components which enhance the sence of presense from two channels of audio signals and increasing or decreasing pressure for the components is provided. The audio signal processing device (illustrated as an audio signal processing portion 20) is provided with a signal extraction portion (illustrated as a signal separation and extraction portion 23) for extracting, from input audio signals of two channels, uncorrelated signals derived from the signals; an uncorrelated signal increase/decrease portion (illustrated as gain adjustment portions 25a and 25b) for increasing or decreasing signals extracted at the signal extraction portion; and an audio signal summing portion for summing the signals that have been increased or decreased at the uncorrelated signal increase/decrease portion and the input audio signal.

Abstract: A modified limiter of the present invention produces a consistent loudness level across one or more input tracks. The modified limiter has a target threshold, wherein signal levels below the target threshold are adjusted. The signal levels below the target threshold are adjusted by using an upward expander, which is not a time-based parameter but, instead, is based on signal levels only. A release target of the modified limiter is a highest peak within a look-ahead delay. The modified limiter is highly optimized as compared to other look-ahead limiters. As such, its processing footprint on an application such as audio management software like iTunes™, or a consumer music playback device like an iPod™, is minimal. The modified limiter is also able to be used in other technologies, such as but not limited to cell phones like an iPhone™ and set-top boxes like an Apple TV™.

Abstract: A method for controlling consumer exposure to audio dosage for a composite audio signal, the method comprising: sampling the composite audio signal; transforming the sampled audio signal, using a fast fourier transform algorithm, to produce a signal representative of the amplitude of component frequencies of the audio signal; comparing the transformed audio signal with a predefined impulse control threshold profile, representing the target maximum amplitude for each frequency component for the audio signal threshold profile to produce a configuring signal representative of the difference between the broadcast signal and the profile; using the configuring signal to automatically configure in real time a Finite Impulse Response (FIR) filter so that it attenuates the amplitude of the transformed audio signal in frequency bands centered on the frequencies at which the target threshold is exceeded; and outputting the attenuated audio signal for consumer exposure.

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

Abstract: An acoustic device, which reproduces an acoustic signal of one acoustic source selected from a plurality of acoustic sources. A first averaging unit averages signal levels of the acoustic signal for a first time period to obtain a first average signal. A second averaging unit averages the signal levels for a second time period shorter than the first time period to obtain a second average signal. A selecting unit selects either the first average signal or the second average signal as one signal having a signal level higher than the other. A gain decreasing unit decreases a gain of the acoustic signal according to the signal level of the one signal if the signal level of the one signal is higher than a threshold value determined according to the one acoustic source. A correcting unit corrects the signal levels of the acoustic signal according to the gain.

Abstract: An apparatus for processing an input sound signal, the apparatus including: gain circuitry configured to control a gain based on a plurality of respective sub-signals of the input sound signal; and an amplification apparatus configured to adjust the amplification of all the plurality of amplitudes based on the gain.

Abstract: Various dynamic audio ducking techniques are provided that may be applied where multiple audio streams, such as a primary audio stream and a secondary audio stream, are being played back simultaneously. For example, a secondary audio stream may include a voice announcement of one or more pieces of information pertaining to the primary audio stream, such as the name of the track or the name of the artist. In one embodiment, the primary audio data and the voice feedback data are initially analyzed to determine a loudness value. Based on their respective loudness values, the primary audio stream may be ducked during the period of simultaneous playback such that a relative loudness difference is generally maintained with respect to the loudness of the primary and secondary audio streams. Accordingly, the amount of ducking applied may be customized for each piece of audio data depending on its loudness characteristics.

Abstract: Described is a unit for amplifying and processing audio signals for driving an electro-acoustic transducer (D), comprising: an input for audio signals; a processor for audio signals (107); an output for a signal for driving said electro-acoustic transducer; and an input for at least one operating quantity of the electro-acoustic transducer. The audio-signal processor is programmed for setting a series of parameters defining a transducer to be emulated, the parameters of which define a model of the transducer to be emulated. The input audio signal is processed on the basis of said at least one operating quantity of the electro-acoustic transducer to obtain a behavior of the electro-acoustic transducer that emulates the transducer defined by said series of parameters set.

Abstract: A mobile device includes at least one speaker and an audio subsystem for providing an audio signal to the at least one speaker for playback by the at least one speaker. The audio subsystem operates by detecting a distance to a user of the mobile device using a proximity sensor and controlling volume output of the at least one speaker based on the distance to the user of the mobile device and the operational status of the proximity sensor. If it is determined that the proximity sensor is functional, the audio subsystem applies a gain factor to the audio signal that varies based on the distance to the user or limits the maximum allowable speaker output based on the distance. A high or low gain setting may be applied for when the user is far away or in close proximity to the mobile device.

Abstract: An audio signal processing apparatus which includes an input analysis unit which analyses the characteristics of an input signal and generates an input sound feature value; an environment analysis unit which analyses the characteristics of the environmental sound and generates an environmental sound feature value; a mapping control information generation unit which generates mapping control information as control information of amplitude conversion processing to the input signal by application of the input sound feature value and the environmental sound feature value; and a mapping process unit which performs amplitude conversion on the input signal based on a linear or non-linear mapping function determined according to the mapping control information and generates an output signal.

Abstract: An audio information source interface that receives first audio information from an audio information source directly connected to the audio playback device; an ADC configured to sample the first audio information; a network interface configured to connect the audio playback device to a network and to receive packets from the network containing second audio information and playback timing information from a network audio information source; a DAC for generating an analog audio signal based on either the first or second audio information; an audio amplifier interconnected with the DAC and configured to amplify the audio signal; an audio reproduction device interface interconnected with the audio amplifier; and, a control module configured to receive commands via the network interface and to provide to the DAC one of the first or second audio information.

Abstract: Speaker control systems are implemented using a variety of systems and methods. In one such system, an audio signal is distributed to at least one speaker selected from a set of speakers. The system includes a set of audio-distribution circuits for distributing the audio signal to the set of speakers. Logic circuitry is used to automatically determine an impedance-based parameter for use in distributing the audio signal to the at least one selected speaker based at least partly on a load-tolerance parameter for an audio source and on an impedance of the at least one selected speaker. The logic circuitry controls distribution of the audio signal to the selected speakers based at least partly on the impedance-based parameter. One or more of the set of audio-distribution circuits are used by the logic circuitry to distribute the audio signal to the at least one selected speaker.

Abstract: An audio signal adjusting method is disclosed, which comprises steps of: determining an environment noise level; calculating a ratio between an environment noise level and an output audio signal level to form a measuring value; comparing the measuring value with a predetermined threshold as an audio comfort value; and determining whether or not to adjust an audio signal parameter according to the comparison result. If the comparison result is incorrect, then the audio signal parameter is adjusted, and the output audio signal should be outputted according to the adjusted audio signal parameter. The audio signal adjusting method can satisfy the different user needs on different environment types.

Abstract: A sound processing apparatus includes a sound determination portion operable to determine whether an input sound includes a first sound emitted from a particular source based on location information of the source, a sound separation portion operable to separate the input sound into the first sound and a second sound emitted from a source different from the particular source if the sound determination portion determines that the input sound includes the first sound, and a sound mixing portion operable to mix the first sound and the second sound separated by the sound separation portion at a prescribed volume ratio.

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: Provided is method for automatic audio gain modulation and a related apparatus. In order to prevent the unstable signals as continuously increasing or decreasing the gain when the signals hover around an upper threshold, it's featured that the method is to predict the signal intensity as pre-adding a predict gain in a hold time before outputting signals. By determining whether the prediction exceeds a predetermined threshold or not, the suitable and adjustable gains can be obtained without exceeding the predetermined threshold. In the automatic gain control mechanism, it's to decrease the gain gradually as automatically entering an attack time, or to increase the gain gradually as performing a conventional auto-gain control procedure including entering a release time. The claimed method can modulate the gain automatically by predicting the gain, and confine the signals under a threshold. In addition to avoid sawtooth phenomena, the method can stabilize the signals.

Abstract: In an embodiment, an oversampled data converter includes a lowpass filter having a filter stage comprising a dynamic limiter, where the dynamic limiter having a limit set by an signal level at an input to the oversampled data converter. The oversampled data converter also includes a quantizing block comprising an input coupled to an output of the lowpass filter and an output coupled to an input of the lowpass filter.

Abstract: A sound reproducing device includes parametric speakers disposed to a display, a loudspeaker, drivers, a human-body sensor, and a controller. The loudspeaker has a wider directivity than the parametric speakers. The drivers change directions of the parametric speakers, respectively. The parametric speakers and the drivers constitute at least two parametric speaker units each composed of respective one of the parametric speakers and respective one of the drivers. The controller reproduces sound information based the number of persons in front of the display detected from an output from the human-body senor. The sound reproducing device reduces the possibility for a number of persons to hear the sound noisily, and allows drawing more attention to the content, e.g. an advertisement.

Abstract: An apparatus and a method for automatically controlling a gain using phase information are provided. The apparatus includes a frequency conversion unit converting each of input signals received from a plurality of acoustic input apparatuses to frequency input signals, a factor determination unit determining a scaling factor according to a gain difference between the input signals based on a phase difference of the frequency input signals, and a scaling performance unit performing scaling for gain compensation between the input signals based on the scaling factor.

Abstract: A 3-point positioning device senses a scene including an auxiliary positioning module by means of an image sensor, and accordingly generates a sensed image. The auxiliary positioning module comprises a first, a second, and a third auxiliary positioning units. By means of the sensed image, the 3-point positioning device recognizes a first, a second, and a third sensing units of the image sensor corresponding to the first, the second, and the third auxiliary positioning units. In this way, the 3-point positioning device can calculate out an absolute location of the auxiliary positioning module according to the locations or the received power of the first, the second, and the third sensing units.

Abstract: Disclosed is a gain control system in which speech model constituted from a sound pressure and a feature is stored in a speech model storage unit for each of a plurality of phonemes or for each of clusters into which a speech is divided. When an input signal is given, a feature conversion unit calculates a feature and a sound pressure of the input signal. A sound pressure comparison unit determines a sound pressure ratio between the input signal and each of speech models. A distance calculation unit calculates a distance between the feature of the input signal and the feature of each of the speech models. A gain calculation unit calculates a gain value from the sound pressure ratio and information on the distance. A sound pressure compensation unit thereby compensates for the sound pressure of the input signal.

Abstract: When any one of a plurality of amplifiers has been selected in response to amplifier selecting operation, an amplifier Detail View of the selected amplifier is displayed, where corresponding parameters are displayed via elements of the amplifier Detail View. When any one of a plurality of groups has been selected in response to group selecting operation, a group Detail View of the selected group is displayed, where maximum and minimum values detected from among values of a parameter of individual amplifiers of the selected group are displayed via a band-shaped element of a corresponding fader element. In another example, where a hierarchical group is selected in a tree display section on a control screen, respective state information, including parameter values or states of parameters, of a plurality of amplifiers belonging to the selected hierarchical group is displayed in a state information section.

Abstract: Provided is an information processing apparatus including a display control unit and an output unit. The display control unit displays image data of an operation unit including a rotary operation dial, and displays while rotating, according to a designated target volume value, the operation dial from a rotation angle position corresponding to a current volume value to a rotation angle position corresponding to the target volume value based on at least one of a predetermined speed and a predetermined acceleration speed. The output unit outputs a volume value corresponding to the rotation angle position of the operation dial displayed by the display control unit.

Abstract: An audio amplifier circuit and a method of the same are provided. The audio amplifier circuit includes a processor, a pre-processing module, and a post-processing module. The processor controls a pre-gain value of the pre-processing module and a post-gain value of the post-processing module in accordance with a volume adjustment signal so as to process an audio input signal and generate an audio output signal. The pre-gain value and the post-gain value are dynamically adjusted by the audio amplifier circuit and a method thereof by according to the volume adjustment signal so as to generate low distortion, low noise audio output signals and improve audio playback quality of an audio output device.

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: According to one embodiment, a digital signal generator includes an amplifying unit, a reference voltage generator and a modulator. The amplifying unit amplifies an analog input signal having a signal level linearly depending on a temperature. The reference voltage generator generates a reference voltage linearly depending on the temperature. The modulator converts the analog input signal amplified by the amplifying unit into a digital output signal based on the reference voltage.

Abstract: A home theater system, an audio output adjusting device of a home theater system and methods thereof allow a prescribed or three-dimension audio configuration state for at least one prescribed position even when the prescribed location and/or a configuration of the speakers in a home theater system are modified. The device can include a checker for analyzing at least one of volume and delay time of a test audio signal outputted from speakers and providing the analyzed information to a controller. The controller can control output level and timing of audio signals subsequently output from the speakers based on the analyzed information.

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.