Abstract: A voice band extender includes a component separator for receiving a band-limited speech signal over a telecommunications line to separate the speech signal into a noise-suppressed signal and an extracted-noise signal, a noise-suppressed signal component extender for adding a signal having its frequency band higher than that of the noise-suppressed signal to thereby produce an extended noise-suppressed signal, an extracted-noise signal component extender for producing an extended extracted-noise signal, a signal intensity adjuster for adjusting signal intensity of either or both of the extended noise-suppressed signal and the extended extracted-noise signal, and a synthesizer for combining the extended noise-suppressed signal and the extended extracted-noise signal obtained by the intensity adjustment. The respective frequency bands of the extracted-noise signal and the noise-suppressed signal are thus extended.

Abstract: A communication system having a common ground mechanism is provided. The communication system comprises a local host, a local telephone device, a sound input path and a sound output path. The local telephone device is connected to a remote telephone device through a telecom facility. The sound input path comprises a first transformer module and is connected between the local telephone device and the local host such that the local telephone device may transfer a sound input to the local host. The sound output path comprises a second transformer module and is connected between the local telephone device and the local host such that the local telephone device may receive a sound output from the local host. The first and the second transformer modules couple a first ground of the telecom facility and a second ground of the local host together such that the first and second grounds are common-grounded.

Abstract: A telephone with a speaker and a microphone uses the speaker both as an electric-to-acoustic transducer, to reproduce the far end party's voice, and an acoustic-to-electric transducer, to capture external sound in the vicinity of the near end party. From the captured sound signal and the signal picked up by the microphone, a decision unit decides whether or not the external sound is being blocked. When the external sound is blocked, a volume control unit increases the volume of the far end party's voice as reproduced by the speaker. The near end party can accordingly turn up the speaker volume by moving the part of the telephone including the speaker close to the user's ear to block external sound, without having to manipulate any manual controls.

Abstract: A system and method for enhancing communications through a phone. A voice communication is received from a user of the phone. A secondary signal is received from an environment in proximity to the phone. The secondary signal is processed to determine an inverse signal in response to receiving the secondary signal. The inverse signal is combined with the voice communication and the secondary signal to destructively interfere with the secondary signal for allowing a receiving party to more effectively communicate with the user.

Abstract: A method is disclosed that enables the managing of the overall sound level in an enterprise environment where telephones are used. A data-processing system such as a private branch exchange monitors whether one or more telephones are in use. Based on detecting when a first endpoint is in use and, therefore, when the endpoint's user is present, the private branch exchange controls one or more characteristics of the loudspeaker volume at a second endpoint. By accounting for other considerations such as the spatial closeness between the endpoints, which can be determined from office dimensions stored in a database, the private branch exchange of the illustrative embodiment is able to determine the degree of sound that is coupling over from one endpoint location to another. On a larger scale, the exchange is able to control the loudspeaker volumes of all of the endpoints in the workplace area. In doing so, the exchange manages the overall acoustic noise present.

Abstract: Constant phone-call quality is kept irrespective of setting of a tone adjusting function of a cellular phone under connection. There is provided a sound collector for collecting voices and converting the voices to a voice signal, a communication unit for communicating with a cellular phone terminal according to a predetermined communication protocol, and a reproducing unit for reproducing the voice signal received in the communication unit.

Abstract: A system and method for providing microphonic isolation on a transmission line. The transmission line has a first part and a second part. The first part of transmission line carries a data signal and a microphonic signal. The microphonic signal has frequencies that include those in a range of substantially 20 Hz to substantially 20 kHz. The system includes an isolation apparatus. The isolation apparatus has an input in electrical communication with a first part of the transmission line, an output in electrical communication with the second part of the transmission line, and a filter in electrical communication with the input and the output. The filter is arranged to substantially remove the microphonic signal received at the input from first part of transmission line and pass the data signal to the output.

Abstract: Methods and systems for secure conferencing in a network have been provided. Relevant endpoints from the provided endpoints are selected on the basis of the endpoint selection parameters received in an SRTP extension. Data streams generated from the selected endpoints are decrypted and mixed, to obtain all the possible data stream mixes. A source identifier is inserted in all the possible data stream mixes. On the basis of source identifiers, endpoints can selectively receive any of the generated data streams.

Abstract: Acoustic echoes in communications systems are distracting and undesirable. Acoustic echoes occur in communications systems where sound produced by a speaker is picked up by a microphone in a communications system. In a stereo playback environment, echo cancellation techniques become more complicated. Echo cancellation can be performed by performing echo cancellation on a center signal, which is the sum of a left channel signal and the right channel signal, or left signal and a difference signal, which is the difference of the right channel signal and the left channel signal. The adaptation rates of the two echo cancellers meet certain constraints to prevent degeneracies in the echo cancellation system.

Abstract: The present disclosure generally pertains to systems and methods for identifying subscriber lines coupled to transceivers of interest. In one exemplary embodiment, an end of a subscriber line is coupled to and terminated by a transceiver of interest. The other end of the subscriber line is coupled to a speaker. The transceiver is then selectively activated and deactivated causing it to transmit its normal frequency spectrum in a predefined pattern. The frequency spectrum includes frequencies audible to humans. Thus, the speaker produces sound capable of being heard by a human, and the sound exhibits periods of noise followed by periods of silence. Indeed, the periods of noise and silence are in a pattern according to the activation/deactivation pattern of the transceiver. Thus, a user can identify the subscriber line that is coupled to the transceiver by identifying which of the lines is coupled to the speaker when the speaker emits noise and silence according to the predefined pattern.

Abstract: Active noise cancellation (ANC) circuitry is coupled to the input of an earpiece speaker in a portable audio device, to control the ambient acoustic noise outside of the device and that may be heard by a user of the device. A microphone is to pickup sound emitted from the earpiece speaker, as well as the ambient acoustic noise. Control circuitry deactivates the ANC in response to determining that an estimate of how much sound emitted from the earpiece speaker has been corrupted by noise indicates insufficient corruption by noise. In another embodiment, the ANC decision is in response to determining that an estimate of the ambient acoustic noise level is greater than an estimate of the anti-noise produced by the ANC. Other embodiments are also described and claimed.

Abstract: The present invention concerns a method and device for cancelling interference between a signal carried by a carrier current line of a system distributing digital services at the home of a subscriber and a signal carried by a telephone line of said system that is situated close to the carrier current line. The method is characterised in that it consists of obtaining the signal carried by the carrier current line, filtering it in order to obtain an estimate of the signal that disturbs the signal carried by the telephone line and subtracting this estimate from the signal carried by the telephone line.

Abstract: Apparatus and Method for canceling or reducing voice sounds produced during a conversation by the user of a cellular telephone or other communication device such that other persons close to the user cannot hear the users voice and is not disturbed by the conversation.

Abstract: In a preferred embodiment, the invention is a mobile communication device having a digital signal processor (DSP), a speaker output node, a local audio source, and an analog front-end (AFE), wherein: (1) the DSP receives a first audio signal corresponding to sound captured by a microphone near a user of the device, (2) if the device is operating in a call mode, the DSP derives a background noise signal from the first audio signal, for subtraction from the first audio signal before transmission to the AFE, and (3) if the device is operating in a non-call mode, then the DSP (i) generates a speaker output signal which substantially corresponds to the first audio signal subtracted from a local audio signal provided by the local audio source and (ii) provides the speaker output signal to a speaker via the speaker output node.

Abstract: Methods and systems are disclosed for setting a volume level of ringback media presented to a calling party. The method may include receiving at a ringback server an indication of ambient noise at a client device, using the received indication as a basis to set a volume level of the ringback media, and transmitting the ringback media at the volume level, for receipt by the client device.

Abstract: A telephone to be connected to a wireline having reduced sensitivity to RF interference over a predetermined frequency range from an adjacent mobile wireless communications device may include a transmit amplifier, a receive amplifier, an audio input transducer connected to the transmit amplifier, and an audio output transducer connected to the receive amplifier. The telephone may further include an RF shield(s) surrounding the audio input transducer and the audio output transducer for reducing RF interference over the predetermined frequency range from the adjacent mobile wireless communications device. Moreover, the telephone may also include at least one RF audio input filter element connected to the audio input transducer and at least one RF audio output filter element connected to the audio output transducer, both of which are also for reducing RF interference over the predetermined frequency range from the adjacent mobile wireless communications device.

Abstract: One exemplary problem addressed by the techniques disclosed herein is that the A-to-D converter of the broad-band audio codec that is being used more frequently in IP telephony, G.722, samples the analog audio source 16,000 times per second, rather than 8,000. Since all G.722-capable telephones must continue to be G.711-capable, one problem is that a microphone that provides appropriate bandpass filtering for G.722 encoding fails to provide adequate filtering for G.711. One exemplary aspect is therefore directed to telephones that must be able to switch back and forth between narrow-band digital audio encoding in which the A-to-D converter samples the audio stream 8,000 times per second, and a wide-band audio encoding in which the A-to-D converter samples the audio stream 16,000 times per second. This is accomplished using one or more of a plurality of switched microphones, a filter and a modification of the resonant frequencies of a handset.

Abstract: Methods and communication systems are presented, in which impulse noise is monitored on a communication channel, and impulse noise protection parameters are adjusted according to the monitored impulse noise without interrupting communication service.

Abstract: Disclosed herein are teleconferencing pods that include a microphone and a loudspeaker for open-air conversations, connectible to a separate telephonic device that generates an intermittent sidetone that is active at certain times not in the control or monitoring of the pod. In those pods is a sidetone suppressor that suppresses a distracting open-air echo produced from the intermittent sidetone and a compensator that detects sidetone transitions to avoid generating an echo signal in the pod and to avoid adapting suppression in the absence of an externally-generated sidetone. Sidetone suppression may be by several means, and may include variables controlled by adaption, including delay factors, scalers and coefficients. Detailed information on various example embodiments of the inventions are provided in the Detailed Description below, and the inventions are defined by the appended claims.

Abstract: Systems and methods for noise cancellation are provided. In particular, information regarding events associated with a computer providing a telephony function or associated with a hardware telephone providing a telephony function is provided to a noise cancellation process. The provided information can be used to identify specific noises associated with events, so that filters specifically adapted for removing an identified noise can be applied. Alternatively or in addition, information regarding the existence and timing of an event can be provided in order to facilitate the application of a noise cancellation filter. Noise cancellation as described can also be applied in connection with recording speech or other audio signals.

Abstract: Embodiments of the invention relate to a feedback mechanism that informs a user of a communication device to adjust the volume, pitch, tone or other characteristic of his voice so as to compensate for noise in the surrounding environment. The feedback mechanism includes feedback circuitry that analyzes audio signals from the microphone and preferably from one or more additional environmental noise sensors. From the analysis, the feedback circuitry determines characteristics of the user's voice and characteristics of the environmental noise, and provides an analysis of how the user might modify his voice to best compensate for environmental noise. This analysis results in an indication to the user, such as through a vibration, a sound, or graphical indication on the device, which tells the user whether and to what extent the user should adjust a characteristic of his voice to best overcome such environmental noise.

Abstract: A portable electronic device and personal hands-free accessory are provided. The personal hands free accessory includes at least one electro-acoustic speaker, and at least one microphone that receives ambient sound and converts the ambient sound to an ambient sound signal.

Abstract: A noise-canceling device of a voice communication terminal that removes noise elements included in received voice signals. The device comprises: a digital filter array that exhibits filter qualities in response to a coefficient setting signal showing each supplied arrays of filter coefficients, and includes a first-stage filter that receives the received voice signals as well as multiple later-stage filters connected thereto in a straight line; a filter qualities designator that generates input designation that designates each qualities of the multiple digital filters forming the digital filters array; and a filter coefficient setter that retains multiple arrays of filter coefficients, extracts a filter coefficient array corresponding to the designation input from among the multiple filter coefficient arrays, and supplies to each multiple digital filters.

Abstract: Embodiments disclosed herein describe a network device including a class AB common mode suppression (CMS) circuit coupled in parallel between a line voltage source and a physical layer (PHY) device that provides active EMI suppression and Phy device termination. A network connector is coupled to provide the line voltage source to the class AB CMS circuit. The class AB CMS circuit provides current to the PHY device, terminates open-drain transmit drivers of the PHY device and suppresses common mode noise thereby minimizing electromagnetic interference. In other embodiments, the class AB CMS circuit is coupled in parallel between the network connector and a physical layer (PHY) device. The class AB CMS circuit suppresses common mode noise, and terminates open-drain transmit drivers of the PHY device, thereby minimizing electromagnetic interference.

Abstract: A method and an electronic device for communication quality improvement based on ambient noise sensing are provided for improving communication quality. The method includes: during a call, dynamically setting a noise reduction mode in response to the varying of an ambient noise amount of at least one audio signal received by the electronic device. The setting step includes: during the call, dynamically determining whether the ambient noise amount indicates the needs of the application of noise reduction. If the ambient noise amount indicates the needs of the application of noise reduction, then one of a plurality of noise reduction levels is selected according to the ambient noise amount, and the noise reduction mode is set according to the selected noise reduction level, wherein the noise reduction levels include at least two levels corresponding to two different amounts of noise reduction.

Abstract: A method and a computer program product for performing the method are disclosed for detecting a noise level on a conference bridge telephone call, the method including but not limited to establishing via a first conference server, a plurality of two-way telephonic communication end user device connections between a plurality of end user devices and a plurality of conference ports, wherein each of the plurality of conference ports are in data communication with the first conference server; measuring a noise level on each one of the end user device connections; detecting a noisy end user device connection having a noise level above a threshold level on at least one of the plurality of end user device connections; identifying the noisy end user device connection; and taking predetermined mitigation steps to reduce the noise from the noisy end user device connection.

Abstract: A voice band extender includes a component separator for receiving a band-limited speech signal over a telecommunications line to separate the speech signal into a noise-suppressed signal and an extracted-noise signal, a noise-suppressed signal component extender for adding a signal having its frequency band higher than that of the noise-suppressed signal to thereby produce an extended noise-suppressed signal, an extracted-noise signal component extender for producing an extended extracted-noise signal, a signal intensity adjuster for adjusting signal intensity of either or both of the extended noise-suppressed signal and the extended extracted-noise signal, and a synthesizer for combining the extended noise-suppressed signal and the extended extracted-noise signal obtained by the intensity adjustment. The respective frequency bands of the extracted-noise signal and the noise-suppressed signal are thus extended.

Abstract: A technique is directed to controlling speaker volume of a speaker of a portable communications device (e.g., a cellular telephone, a cordless hand phone, a combination of a cellular phone and a hands-free ear piece, etc.). The technique involves receiving an environmental signal from an environmental microphone of the portable communications device. The technique further involves identifying a level of environmental noise based on the environmental signal, and adjusting the speaker volume of the speaker of the portable communications device (e.g., ringer volume, voice output volume, etc.) based on the identified level of environmental noise. Such operation enables a user to accommodate a variety of settings without burdening the user with ongoing manual volume control of the portable communications device.

Abstract: Disclosed is an echo suppresser or cancellation circuit including a speech signal extrapolation unit. The speech extrapolation unit may provide extrapolated signal segment approximations to an echo segment replacement unit. The echo segment replacement unit may replace a segment of a first speech signal suspected of being corrupted with an echo component of a second signal using the extrapolated signal segment approximated to represent the corrupted signal segment.

Abstract: A telephone terminal (UE) comprises telephone signal receiver means (MER) and processing means (MT) adapted in the absence of voice activity to determine a level of comfort noise to be generated as a function of noise data and in the presence of received telephone signals representing voice signals to determine noise signals as a function of noise data and then to subtract at least a portion of said noise signals from the received telephone signals.

Abstract: A method for adjusting the volume and frequency response for the audio output in a mobile communication device comprises estimating the noise level of the environment surrounding the mobile communication device and then adjusting the volume and frequency response based on the estimated noise level.

Abstract: A bandwidth extension apparatus can generate from an inputted speech signal a bandwidth-extended signal whose bandwidth is automatically adjusted according to the surrounding hearing situation by means of a parameter. The apparatus has a surrounding characteristic predictor for predicting the characteristic amount of the surrounding sound of the phone terminal on which the apparatus is installed; an adjustment amount predictor for comparing the characteristic amount with the parameter to predict the adjustment amount for the parameter; a speech characteristic predictor for predicting the characteristic amount of the inputted speech signal; and an adjustment amount determiner for determining an adjustment amount for updating the parameter based on the characteristic amount of the surrounding sound and the inputted speech signal.

Abstract: An approach is provided for compensating audio signals during a communication session by receiving audio signals from parties to the communication session, sampling the audio signals to determine signal level and noise floor, and compensating the audio signals to bring each signal to approximately an equal level based on the determination.

Abstract: A method of comparison between pieces of information characterizing reference values and pieces of information characterizing current values of sound-reproducing systems of a system of (n) microphones mi and (p) speakers hpj for the control of said sound-reproducing systems characterized in that: A: for each speaker hpj, at least one sound signal S is sent on the speaker hpj, for each microphone mi, a piece of information hpjmi is retrieved, this piece of information characterizing the sound-reproducing system comprising the speaker hpj and the microphone mi, B: a reference matrix Qr is saved, this reference matrix being constituted by all the pieces of reference information hpjmi obtained following the sending of the sound signal S, C: as soon as a comparison is to be made, the step A is run with a sound signal S? to obtain current information on a matrix Q, D: the matrices Q and Qr are compared.

Abstract: A noise reduction device including: an input signal spectrum obtaining unit that obtains an input signal spectrum by a subband unit based on a current frame of an input signal; an averaged spectrum obtaining unit that obtains an averaged spectrum of the input signal by averaging the input signal spectrum; an estimated noise spectrum obtaining unit that obtains an estimated noise spectrum estimated based on a past frame of the input signal by the subband unit; and an SN ratio obtaining unit that obtains an SN ratio by the subband unit, based on the averaged spectrum of the input signal obtained by the averaged spectrum obtaining unit, the estimated noise spectrum obtained by the estimated noise spectrum obtaining unit, and a function of the averaged spectrum of the input signal obtained by the averaged spectrum obtaining unit and the estimated noise spectrum obtained by the estimated noise spectrum obtaining unit.

Abstract: In an on-board hands-free communication apparatus, a gain of an amplifier of a speaker for outputting an incoming voice signal from a telephone device varies with a volume set value of a speaker volume switch adjusted by an user. When the gain of the amplifier increases equal to or greater than a threshold value, a feature extraction amplification circuit performs a feature extraction amplification processing to apparently amplify output power of the incoming voice signal, and a control circuit fixes the gain of the amplifier to the threshold value regardless of the volume set value.

Abstract: A novel mechanism for sharing filter taps across a plurality of interference cancellers. Each interference canceller may be directed to impairment, such as Ethernet impairments, including Ethernet 1000Base-T impairments. Various interference impairments include echo cancellation, NEXT cancellation and/or other interference detection or cancellation, etc. The hardware requirements of the interference impairment cancellers are reduced by sharing filter taps among the cancellers. In a first embodiment, the taps from a unified filter tap bank are shared across all the interference impairment cancellers for all four channels and over all ports. In a second embodiment, a portion of the taps of each filter are shared wherein each canceller comprises a fixed filter tap portion and a shared filter tap portion. A tap allocation algorithm assigns taps to those cancellers that need them the most. A canceller configuration is selected that yields maximal interference mitigation and the taps are allocated accordingly.

Abstract: In a communication apparatus has a line control unit that controls connection with a communication line network, a communication unit transmits/receives data through the communication line network. A detection circuit detects one of a line voltage value and a line current value when the communication apparatus is connected to the line network. A memory unit stores multiple voltage-current characteristic curves defining relationships of line voltage values with respect to line current values satisfying a predetermined standard. An adjusting unit adjusts the line voltage value and the line current value so as to satisfy the predetermined standard based on one of the voltage-current characteristic curves stored in the memory unit. A selecting unit selects another voltage-current characteristic curve different from the one of the voltage-current characteristic curves from among the multiple voltage-current characteristic curves stored in the memory unit.

Abstract: There is provided a call voice processing apparatus including an input correction unit that corrects characteristics of a first input sound input from a first input apparatus to characteristics of a second input sound input from a second input apparatus, a sound separation unit that separates the second input sound into a plurality of sounds, a sound type estimation unit that estimates sound types of the plurality of sounds separated by the sound separation unit, a mixing ratio calculation unit that calculates a mixing ratio of each sound in accordance with the sound type estimated by the sound type estimation unit, a sound mixing unit that mixes the plurality of sounds separated by the sound separation unit in the mixing ratio calculated by the mixing ratio calculation unit, and an extraction unit that extracts a specific sound from the first input sound corrected by the input correction unit.

Abstract: In a noise estimation apparatus, a microphone converts sound into an electric signal and outputs the electric signal as a sound signal. A noise estimator performs estimation for estimating a magnitude of a noise component contained in the sound signal so as to generate an estimated noise signal. The noise estimator limits a minimum value of a noise level of the noise component contained in the sound signal during the estimation to a predetermined default value, and integrates the noise level having the limited minimum value to generate the estimated noise signal.

Abstract: A noise reduction system including an audio-visual user interface combines visual features extracted from a digital video sequence with audio features extracted from an analog audio sequence. The digital video sequence may show the face of a speaker, and the analog audio sequence may include background noise in an environment of said speaker. Audio sequence detection means are used to detect said analog audio sequence, and audio feature extraction and analysis means are used to analyze said analog audio sequence and extract said audio features therefrom. Video sequence detection means are used to detect said video sequence, and visual feature extraction and analysis means are used to analyze the detected video sequence and extract said visual features therefrom. A noise reduction circuit is configured to separate the speaker's voice from said background noise based on a combination of derived speech characteristics and output a speech activity indication signal.

Abstract: A noise suppression device comprises subband SN ratio calculation means which receives a noise likeness signal, an input signal spectrum and a subband-based estimated noise spectrum, calculates the subband-based input signal average spectrum, calculates a subband-based mixture ratio of the subband-based estimated noise spectrum to the subband-based input signal average spectrum on the basis of the noise likeness signal, and calculates the subband-based SN ratio on the basis of the subband-based estimated noise spectrum, the subband-based input signal average spectrum and the mixture ratio.

Abstract: Methods and communication systems are presented, in which impulse noise is monitored on a communication channel, and impulse noise protection parameters are adjusted according to the monitored impulse noise without interrupting communication service.

Abstract: There is disclosed conductor and connector configurations for use in communications systems which take advantage of the orientation and positioning of adjacent balanced pairs of conductors to reduce cross talk. These conductor and connector configurations may be used to reduce cross talk between adjacent balanced pairs of a single transmission path, or to reduce alien cross talk between adjacent balanced pairs located in different transmission paths.

Abstract: A method for providing an embellishment representation of a noise information is discloses.

Abstract: A telephone to be connected to a wireline having reduced sensitivity to RE interference over a predetermined frequency range from an adjacent mobile wireless communications device may include a transmit amplifier, a receive amplifier, an audio input transducer connected to the transmit amplifier, and an audio output transducer connected to the receive amplifier. The telephone may further include an RE shield(s) surrounding the audio input transducer and the audio output transducer for reducing RE interference over the predetermined frequency range from the adjacent mobile wireless communications device. Moreover, the telephone may also include at least one RE audio input filter element connected to the audio input transducer and at least one RE audio output filter element connected to the audio output transducer, both of which are also for reducing RE interference over the predetermined frequency range from the adjacent mobile wireless communications device.

Abstract: A mobile device is provided that includes a digit input that can be manipulated by a user's fingers or thumb, an air conduction microphone and an alternative sensor that provides an alternative sensor signal indicative of speech. Under some embodiments, the mobile device also includes a proximity sensor that provides a proximity signal indicative of the distance from the mobile device to an object. Under some embodiments, the signal from the air conduction microphone, the alternative sensor signal, and the proximity signal are used to form an estimate of a clean speech value. In further embodiments, a sound is produced through a speaker in the mobile device based on the amount of noise in the clean speech value. In other embodiments, the sound produced through the speaker is based on the proximity sensor signal.

Abstract: In a noise suppresser, an input signal is converted to frequency domain by discrete Fourier analysis and divided into Bark bands. Noise is estimated for each band. The circuit for estimating noise includes a smoothing filter having a slower time constant for updating the noise estimate during noise than during speech. The noise suppresser further includes a circuit to adjust a noise suppression factor inversely proportional to the signal to noise ratio of each frame of the input signal. A noise estimate is subtracted from the signal in each band. A discrete inverse Fourier transform converts the signals back to the time domain and overlapping and combined windows eliminate artifacts that may have been produced during processing.

Abstract: A system for improving the attenuation of an undesired signal found in a differential signal path through the use of inductive coupling. The system includes a primary inductor, a secondary inductor, and a filter. The primary inductor and the secondary inductor operably couple an input differential signal pair to an output differential signal pair, and the filter attenuates an undesired signal in the output differential signal pair.

Abstract: An intelligent automatic volume control technique (100) for a communication device is provided that takes a measurement of the background audio environment at the time a user manually selects the volume (102, 104). A user preference is established for the volume setting relative to the background audio level being experienced (106). The informational relationship is stored and used for automatic volume control (108). When the background noise changes (110), the automatic volume control makes an intelligent decision on changing the volume (116). The current background noise level, the initial user volume setting and the background noise level when that volume was set are all used in the volume change decision making process.