Abstract: A signal processor (105) for a headset (101) configured with a microphone terminal (106) for receiving a microphone signal, a loudspeaker terminal (107) for outputting a loudspeaker signal, and a far-end terminal (108) for communicating an inbound signal and an outbound signal with a far-end; comprising: a side-tone path (110) configured to generate a side-tone signal from the microphone signal via a controllable side-tone filter; wherein a side-tone filter controller (114) receives the microphone signal and computes a first noise estimate with a signal-to-noise level of the microphone signal at respective frequency bands and based thereon controls the side-tone filter (111) to improve or optimize a signal-to-noise ratio.

Abstract: A voice detection method which makes it possible to detect the presence of voice signals in an noisy acoustic signal x(t) from a microphone, including the following consecutive steps: calculating a detection function FD(?) based on calculating a difference function D(?) varying in accordance with the shift ? on an integration window with length W starting at the time t0, with: a step of adapting the threshold in said current interval, in accordance with values calculated from the acoustic signal x(t) established in said current interval; searching for the minimum of the detection function FD(?) and comparing the minimum with a threshold, for (?) varying in a predetermined time interval referred to as current interval so as to detect the possible presence of a fundamental frequency F0 that is characteristic of a voice signal in said current interval.

Abstract: A method of reducing noise in an acoustic system, the method comprising at a first user terminal: receiving an audio signal from at least one further user terminal over a network; executing a communication client on a processing unit, the client configured so as when executed to: supply the audio signal to an audio signal processing module of the first user terminal, wherein the processing module processes the audio signal, whereby a level of gain is applied to the audio signal, and outputs a processed audio signal to a speaker; estimate a noise level of the audio signal and the processed audio signal and estimate the gain applied by the processing module taking into account both the noise level estimates; selectively apply a system gain reduction step to at least one of the audio signal and an audio signal received via a microphone, based on at least the estimated gain.

Abstract: Judgment result deriving means 74 makes a judgment between active voice and non-active voice every unit time for a time series of voice data in which the number of active voice segments and the number of non-active voice segments are already known as a number of the labeled active voice segment and a number of the labeled non-active voice segment and shapes active voice segments and non-active voice segments as the result of the judgment by comparing the length of each segment during which the voice data is consecutively judged to correspond to active voice by the judgment or the length of each segment during which the voice data is consecutively judged to correspond to non-active voice by the judgment with a duration threshold. Segments number calculating means 75 calculates the number of active voice segments and the number of non-active voice segments.

Abstract: A telephone includes: a housing; a voice output device that is placed inside the housing and produces voice; a contact detection unit that detects a position of an object that contacts the housing; a position offset calculation unit that calculates a distance between the contact position of the object detected by the contact detection unit and the voice output device; and a voice adjustment unit that adjusts the voice produced from the voice output device, depending on the distance.

Abstract: Embodiments of the present invention include methods and apparatuses for adjusting audio content when more multiple audio objects are directed toward a single audio output device. The amplitude, white noise content, and frequencies can be adjusted to enhance overall sound quality or make content of certain audio objects more intelligible. Audio objects are classified by a class category, by which they are can be assigned class specific processing. Audio objects classes can also have a rank. The rank of an audio objects class is used to give priority to or apply specific processing to audio objects sin the presence of other audio objects of different classes.

Abstract: Apparatus, methods and articles of manufacture to predict vectored digital subscriber line (DSL) performance gains are disclosed. A disclosed example method includes determining a model coefficient of a noise-to-margin ratio (NMR) model from performance data measured for a DSL subscriber loop prior to provisioning of vectoring for the DSL subscriber loop, computing, using the model coefficient, a first NMR value with disturbers enabled and a second NMR value with disturbers disabled, and estimating an expected vectoring performance gain for the DSL subscriber loop based on the first and second NMR values.

Abstract: A method for removing a noise signal from an input signal that is received. In the method, the amount of energy of the input signal is detected, a noise signal included in the input signal is estimated, an intermediate output signal is generated by removing the estimated noise signal from the input signal, and a final output signal is generated by amplifying the amount of energy of the intermediate output signal based on a difference between the amount of energy of the intermediate output signal and the amount of energy of the input signal.

Abstract: Embodiments of the present invention include methods and apparatuses for adjusting audio content when more multiple audio objects are directed toward a single audio output device. The amplitude, white noise content, and frequencies can be adjusted to enhance overall sound quality or make content of certain audio objects more intelligible. Audio objects are classified by a class category, by which they are can be assigned class specific processing. Audio objects classes can also have a rank. The rank of an audio objects class is used to give priority to or apply specific processing to audio objects sin the presence of other audio objects of different classes.

Abstract: A signal processing system enhances communication. When an audio signal is detected, an echo component of the detected audio signal may be estimated. A near party communication device may receive an audio signal from a remote party communication device. A characteristic of the received audio signal may be adjusted based on the echo component of the detected audio signal.

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: Apparatus, methods and articles of manufacture to predict vectored digital subscriber line (DSL) performance gains are disclosed. A disclosed example method includes determining a model coefficient of a noise-to-margin ratio (NMR) model from performance data measured for a DSL subscriber loop prior to provisioning of vectoring for the DSL subscriber loop, computing, using the model coefficient, a first NMR value with disturbers enabled and a second NMR value with disturbers disabled, and estimating an expected vectoring performance gain for the DSL subscriber loop based on the first and second NMR values.

Abstract: Systems and methods for operating a telecommunications device in whisper mode are presented. The method generally includes adjusting a transmit audio signal gain responsive to a transmit audio signal voice activity status, transmit audio signal speech level, transmit audio signal signal-to-noise ratio, and receive audio signal voice activity status. A sidetone feedback signal gain is adjusted in conjunction with adjusting the transmit audio signal gain.

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: An echo canceller apparatus comprises a receive side attenuator coupled in a receive side signal path that is configured to couple from a conference call bridge to a caller; a convolution processor coupled to the receive side signal path at a convolution processor pick-off point; a double-talk detector coupled to the receive side signal path and to a sending side signal path that is configured to couple from the caller to the conference call bridge; and logic coupled to the receive side attenuator which when executed is responsive to a double-talk condition detected by the double-talk detector and operable to determine a level of echo canceled by the convolution processor, to determine an additional amount of attenuation to introduce, and to activate the receive side attenuator to introduce the additional attenuation.

Abstract: In one embodiment, the present invention includes an apparatus having an automatic gain control (AGC) stage to receive an input signal from a communication channel physical medium, a first local gain stage coupled to an output of the AGC stage, an equalizer coupled to an output of the first local gain stage, an echo canceller to receive local data to be transmitted along the communication channel physical medium, and a second local gain stage coupled to an output of the echo canceller. Other embodiments are described and claimed.

Abstract: A method and system for managing a communication session is provided. The communication session is associated with multiple communication devices. The method includes learning (304) a set of derived acoustic features of an audio communication signal that is associated substantially only with one user of a communication device. The method also includes receiving (306) a communication session signal. The communication session signal is an audio signal that includes a combination of audio communication signals. Each audio communication signal of the audio communication signals is associated with a user of a communication device of the multiple communication devices. The method includes modifying (308) the communication session signal based on the set of derived acoustic features.

Abstract: Methods and arrangements for controlling modes of audio devices according to user selectable features are disclosed. The system can receive a user selection for an alternate mode for audio device. A monitor can monitor audio levels in a physical area and the system can automatically switch an audio device from a current mode to the alternate mode when the monitored audio level exceeds a predetermined threshold. The alternate mode can provide a higher audio level or volume than the current mode. The alternate mode can also be a closed caption mode, a record mode, and a transcript mode to name a few. Thus, when the ambient sound level reaches a predetermined level many audio devices can change modes based on user configured instructions.

Abstract: According to one embodiment, a voice mail apparatus is connectable to a telephone exchange, records a voice message sent from a caller telephone terminal in a mail box corresponding to each telephone terminals, and reproduces the voice message in response to a reproduction instruction. The voice mail apparatus stores in a memory an audio file having approximately the same signal level as the voice message. The voice mail apparatus inputs the audio file in a voice processing device when a request for processing the voice message is given from a request source telephone terminal. The voice mail apparatus controls the processing gain such that the signal level output from the voice processing device becomes a prescribed level, while performing muting for the request source telephone terminal. The voice mail apparatus inputs the voice message to the voice processing device after completing controlling of the processing gain.

Abstract: Serializing circuitry is provided that can multiplex multiple device output signals and that can drive time-multiplexed data signals on the bus wires of a data path of an electronic system. Bus registers placed at the ends of the bus wires can register or buffer the data signals transmitted over the bus wires. The registered signals may be passed on to deserializing circuitry for demultiplexing the data signals to provide parallel device input signals. The bus registers and the serializing/deserializing circuitry can be provided along signal paths that require additional latency.

Abstract: A system, method, and apparatus are directed towards managing an audio message, such as a Voice over Internet Protocol (VOIP) message over a network. The invention employs a statistical mechanism to automatically optimize a gain control for setting a volume of an audio message being sent by a client device. An initial gain value is automatically adjusted based, in part, on a statistical sampling of energy levels in the audio message. Environmental factors, such as a sound card within the client device, background noise, and the like, may also be considered through a setting of a servo coefficient that may be used to map between volume levels and decibel levels. The servo coefficient may also be adjusted based, at least in part, on decibel (dB) feedback information from a destination device for which the audio message is intended.

Abstract: A hybrid serial/parallel bus interface has a data block demultiplexing device. The data block demultiplexing device has an input configured to receive a data block and demultiplexes the data block into a plurality of nibbles. For each nibble, a parallel to serial converter converts the nibble into serial data. A line transfers each nibble's serial data. A serial to parallel converter converts each nibble's serial data to recover that nibble. A data block reconstruction device combines the recovered nibbles into the data block. The data block is employed by a gain controller. Each nibble has at least two start bits whose states collectively represent both a function and/or destination.

Abstract: A signal processing device includes: a speaker amplifier for amplifying a reception voice; a reception voice attenuator for attenuating the reception voice inputted to the speaker amplifier during a voice transmission mode; a reception voice detection circuit for detecting the reception voice outputted from the speaker amplifier; a microphone amplifier for amplifying a transmission voice inputted to a microphone; a transmission voice attenuator for attenuating the transmission voice outputted by the microphone amplifier during a voice reception mode; a transmission voice detection circuit for detecting the transmission voice outputted by the microphone amplifier; and a discriminator for discriminating the mode to be operated among the voice transmission mode and the voice reception mode on the basis of detection outputs from the reception voice detection circuit and the transmission voice detection circuit.

Abstract: A modem includes a base unit for transmitting a data signal, and a communication card which receives the data signal from the base unit over a wireless medium and which performs echo canceling on the data signal. The base unit is in communication with a telephone line and receives an original signal from the telephone line. The base unit generates an RF modulated signal based on the original signal. The base unit includes a transmitter for transmitting the data signal. Circuitry in the base unit receives the original signal from the telephone line and generates a combined data signal from the original signal and echo signals and maintains a peak voltage of the combined signal which is within the linear amplification region of the transmitter.

Abstract: A method, apparatus, system, and signal-bearing medium that in an embodiment determine a degree of correlation between a speaker output signal and a microphone input signal and modulate an adaptive gain of an acoustic echo canceller based on the degree of correlation.

Abstract: A digital telephone answering device having speakerphone capability allows a recorded message to be played back and heard by the far-end party as well as over the local speakerphone by the near-end party during speakerphone operation as if it were a normal receive signal. Moreover, normal speakerphone conversation is possible during this conversational playback mode allowing the far-end party and/or near-end party to break-in over the played back pre-recorded message as desired and be heard by the other party. Both message and conversation signals are preferably (but not necessarily) at similar levels. Also, the message playback at the near end is preferably (but not necessarily) subject to the same speakerphone digital volume control as that received at the far end.

Abstract: An apparatus for controlling operation of a recording device with a telephone configured for orientation in a first state and a second state includes: (a) an information conveying section for conveying information from the telephone to a recording locus in the recording device; (b) an indicating section for receiving a monitor signal having a first value when the telephone is in the first state and having a second value when the telephone is in the second state; the indicating section employs a single fixed bias to generate a first indicating signal when the telephone is in the first state and to generate a second indicating signal when the telephone is in the second state; and (c) an engaging section; the engaging section responds to the first and second indicating signals to orient the recording device for recording the information or for not recording.

Abstract: The present invention is directed to a telecommunication device, usable with standard telecommunication equipment, that automatically resets one or more acoustic parameters to predetermined levels prior to the initiation of the next communication session.

Abstract: Methods and apparatus are provided for an echo cancellation system. The echo cancellation system comprises an automatic gain control (AGC), a first scalar, a second scalar, a signal summing stage, and an adaptive filter. The AGC is responsive to a first signal in a reference path and a second signal in a near end path. The signal summing stage is between the first and second scalar in the near end path and the first and second scalars are responsive to the AGC. The adaptive filter is responsive to the first signal and provides a third signal to the signal summing stage that corresponds to an echo signal and is subtracted from a signal in the near end path. The second scalar has a scale rate that is the inverse of the scale rate of the first scalar such that unity scaling occurs on the near end path.

Abstract: Duplex communication methods and systems with maximum gain limiting such as in telephone handsets and speaker phones are disclosed. The communications system such as a telephone set may generally include a digital signal processor (DSP) having a receive and transmit path in communication with a receiver and a transmitter, respectively, a volume amplifier for amplifying signals on the receive path, and a maximum gain limiter for determining a total return loss between the receive and transmit paths and for limiting a volume amplification level of the volume amplifier to a maximum depending on the total return loss. The maximum gain limiter determines the total return loss by maintaining a total loop gain for the telephone set at less than 0 dB to prevent unstable operation and howling.

Abstract: An apparatus for enhancing audibility of a far-end speech signal from a far-end user to a near-end user in a telephony system includes a near-end background noise signal level estimator (26) at the near-end user. A near-end speech signal level estimator (24) is also provided at the near-end user. A gain control logic (22) determines a gain (G) for amplification of the far-end speech signal based on both estimated speech and background noise signal levels.

Abstract: A push-button signal receiving device and a gain control method whereby error in validity determination of a received signal that accompanies the insertion of an AGC circuit can be prevented. An input bush-button signal is amplified by amplifying circuit and separated into signals of low- and high-frequency bands by first and second band-pass filters, respectively. Dialed number identifying circuit detects the frequencies of the separated signals to identify the dialed number, determines validity of the received push-button signal, and outputs the identified dialed number in accordance with the result of validity determination. Level detecting circuit detects the level of the output signal from the amplifying circuit. Gain control circuit controls the amplification gain of the amplifying circuit in accordance with the detected signal level, and holds the value of the amplification gain of the amplifying circuit while the received push-button signal is judged valid by the dialed number identifying circuit.

Abstract: A system and method for adjusting the volume level of a communications device in response to ambient noise. In one embodiment, ambient noise is sampled once and the volume of a speaker associated with the communications device is increased to, and maintained at, a level sufficient to overcome the ambient noise such that a user can easily carry on a conversation or hear what is being transmitted. In another embodiment, ambient noise is periodically sampled and the volume of the speaker is adjusted in response to the sampled ambient noise. The system and method provides enhanced user convenience and power saving advantages.

Abstract: The present invention relates to a loudspeaker volume control arrangement for a telephone having a loudspeaker and a microphone which controls the loudspeaker volume of the telephone based on the estimated distance between the microphone and the loudspeaker.

Abstract: A system and method for reducing or entirely canceling background or ambient noise from a voice transmission from a communications device. A communications device, such as a mobile telephone, is configured with an ambient noise compensation signal generator that is connected between a microphone and a mixer. The original output of the microphone and a compensation signal generated by the ambient noise compensation signal generator are mixed together prior to being passed to a transmitter. In one embodiment a buffer is provided between the microphone and the mixer to help synchronize the timing of the signals to be mixed. In another embodiment a second microphone is employed to detect ambient noise.

Abstract: A method and system for automatic gain control is provided. The automatic gain control may be used, for example, with a microphone and voice codec of a speakerphone telephone. In an exemplary embodiment, an amplification gain lookup table stores the gain values. The power of the input signal is estimated and the appropriate gain value is selected as a function of the estimated power of the input signal. The gain value is applied to the input signal to provide an automatic gain controlled output signal. In another embodiment, the gain lookup table may be adapted to compensate for the non-linearity of the microphone and voice codec of the speakerphone. In this alternate embodiment, the output power of the automatic gain controlled signal is estimated and compared to a reference signal to generate an error signal. The error signal may be scaled and used to update and dynamically adapt the gain values in the gain lookup table.

Abstract: A transceiver system is disclosed for use in a telecommunication system. The transceiver system includes a transmission circuit including a transmitter input coupled to an input of a transmission amplifier, a receiver circuit including a receiver output coupled to an output of a receiver amplifier, and a transmission line interface circuit that is coupled to an output of the transmission amplifier and to an input of the receiver amplifier. The transmission line interface circuit includes a matching impedance that is directly coupled to a feedback path of the transmission amplifier and that terminates the transmission line of the transceiver system.

Abstract: A method for improving a downlink signal received by a listener on a phone is disclosed. The method includes calculating an environment noise level of the listener and filtering and adjusting gain of the downlink signal based on the environment noise level.

Abstract: A full duplex speaker-phone that performs adaptive filtering to provide increased loop stability between users of the speaker-phone. The invention employs a sliding filter that performs adaptive frequency dependent attenuation of the speech signals within a speaker-phone. By detecting the mode of operation of the speaker-phone in real time, the speaker-phone performs adaptive filtering to ensure a high perceptual quality for the users of the speaker-phone. The sliding filtering is performed using a sliding low pass filter (LPF) in certain embodiments of the invention. Perceptually, the speaker-phone does not surrender the appearance of a full duplex speaker-phone from the perspective of the users. In addition, by performing frequency dependent filtering of the speech signals, the speaker-phone offers improved stability in the speaker-phone loop and reduces residual echoes significantly over conventional technology employed in speaker-phones.

Abstract: A side-tone control unit for a telecommunication instrument is configured to include a side-tone amplifier having a transfer function selectively controlled in accordance with a set of amplifier parameters, and a side-tone controller, coupled to the side-tone amplifier, for selectively applying the set of amplifier parameters to the side-tone amplifier based on the detected energy of an uplink signal and the detected energy of a downlink signal. The side tone controller can select a particular set of amplifier parameters by indexing a table in accordance with quantized values of the detected uplink signal energy and the detected downlink signal energy.

Abstract: An apparatus for enhancing intelligibility of a voice signal in a noisy environment includes a first noise estimator which estimates a far-end noise component from the far-end signal, and a second noise estimator which estimates a near-end noise component from the near-end signal. A noise reduction calculator determines a noise reduction gain from the estimated far-end noise, and an echo gain calculator determines an echo control gain. A master gain calculator combines both the echo control gain and the noise control gain into a master gain which is applied to the far-end signal. A comfort noise generator applies the pre-set minimum threshold of the master gain and constantly matches the spectrum of the far-end noise to synthesize a background noise for selectively mixing onto the far-end signal when echo is determined. Echo is determined based on a comparison of the near-end signal and the far-end signal spectra and by compensating the total gain applied.

Abstract: The mobile telephone is provided with the capability for automatically adjusting the gain of a microphone of the telephone based upon the detected noise level in which the cellular telephone is operated. As the noise level increases, the gain of the microphone is automatically decreased, thereby compensating for the natural tendency of telephone users to speak more loudly in noisy environments. Also, by decreasing the microphone gain, any clipping that might otherwise occur as a result of the user speaking more loudly is avoided and the signal-to-noise ratio is not thereby decreased. Furthermore, because the microphone gain decreases, the volume level of the voice of the user as it is output from the other party's telephone is not unduly loud. Hence, the other party need not manually decrease the speaker gain of his or her telephone.

Abstract: The present invention relates to an apparatus and a method for volume control in digital telephones such as second generation mobile telephone terminals, third generation IMT 2000 image mobile telephone terminals, and ISDN (integrated service digital network) terminals. In particular, an apparatus and a method in accordance with an embodiment of the present invention measure background noise and adjust volume of received sound and/or ring signal accordingly. An embodiment of the present invention measures background noise before ringing ring signal and adjusts volume of ring signal. While communicating, the apparatus automatically adjusts volume of received sound in response to the measured background noise. A telephone terminal in accordance with an embodiment of the present invention includes voice/non-voice determination part, volume measurement part, and gain control part. The voice/non-voice determination part separates pure background noise from composite signal (voice+background noise).

Abstract: An audio processing module, in accordance with the invention, includes an input for receiving input signals. A side tone generator is included for receiving the input signals and for generating a side tone to be output. A noise suppressor is coupled to the input for suppressing noise of the input signal prior to a coding process, the noise suppressor providing feedback to the side tone generator to adjust a gain of the input signal to the output. A method for adjusting the gain of the side tone generator is also included.

Abstract: This invention unifies a set of statistical signal processing, neuromorphic systems, and microelectronic implementation techniques for blind separation and recovery of mixed signals. A set of architectures, frameworks, algorithms, and devices for separating, discriminating, and recovering original signal sources by processing a set of received mixtures and functions of said signals are described. The adaptation inherent in the referenced architectures, frameworks, algorithms, and devices is based on processing of the received, measured, recorded or otherwise stored signals or functions thereof. There are multiple criteria that can be used alone or in conjunction with other criteria for achieving the separation and recovery of the original signal content from the signal mixtures. The composition adopts both discrete-time and continuous-time formulations with a view towards implementations in the digital as well as the analog domains of microelectronic circuits.

Abstract: An adaptive transmit amplifier for automatically adjusting the transmit output characteristics of a headset adapter output stage includes a detector adapted to detect a value of a bias current in a host microphone transmit circuit. The adaptive transmit amplifier further includes a switching stage coupled to the detector and adapted to select a gain level and impedance level of the headset adapter output stage based upon the bias current value. The adaptive transmit amplifier may further include a logic/timing block coupled to the detector and to the switching stage. The logic/timing block is adapted to control a sequence of detection of the bias current and selection of the gain level and impedance level of the headset adapter output stage.

Abstract: The Least Mean Square (LMS) and Normalized LMS (NLMS) algorithms commonly employed in adaptive filters of echo cancelers are further optimized. Finite impulse response (FIR) filters are used to estimate a transfer function of an echo channel in a communications link. The LMS and NLMS algorithms are used to adapt the filter coefficients of the estimated transfer functions. By including the echo channel energy gain in the LMS or NLMS update equation, adaptation speed is increased by making adaptation responsive to the channel energy gain. An algorithm for estimating the echo channel energy gain adapts the estimate based on measured system parameters, such as a measured instantaneous channel gain and a near-end voice level. By considering in the NLMS algorithm, the average energy of either the microphone signal or the error signal, as well as the standard reference signal, a higher nominal update gain can be used. With a higher nominal update gain, the NLMS algorithm will converge more quickly.

Abstract: Adaptive gain control techniques provide correctly adjusted audio signal levels during the entirety of an Internet telephony conversation and are resilient to background noise and loudspeaker echo. Disclosed techniques can account for multiple near-end speakers, as well as changes in near-end environment. In an exemplary embodiment, an adaptive gain controller includes a gain control processor configured to adjust an analog gain for a microphone output signal based on measurements of the microphone output signal and on measurements of a loudspeaker input signal.

Abstract: An efficient method and circuitry for transferring various levels of power across a high voltage isolation barrier. In one embodiment of the invention, a data access arrangement (DAA) is provided with system side circuitry that provides power to line side circuitry via the high voltage isolation barrier. The line side circuitry includes voltage measurement circuitry, such as an analog-to-digital converter, that periodically measures the line side supply voltage. The resultant information is transmitted across the high voltage isolation barrier to the system side circuitry, where control circuitry or a software process compares the digital representation of the measured voltage level to a predetermined value. Based on the results of this comparison, the control circuitry may alter the amount of power transferred across the high voltage isolation barrier to the line side circuitry in order to optimize power transfer efficiency.

Abstract: A device receives a signal that includes human-interpretable audio information. The device automatically adjusts the volume of the audio information at the received end. The volume control is determined by an automatic volume control gain, which is calculated as a function of the automatic gain control gain, a weighted dynamic range compression gain, and a weighted constant gain.