Abstract: A multi-microphone system performs location-selective processing of an acoustic signal, wherein source location is indicated by directions of arrival relative to microphone pairs at opposite sides of a midsagittal plane of a user's head.

Abstract: One embodiment of the present invention provides a system that uses a sound sensor to adjust the audio output of a device. During operation, the system uses the sound sensor to determine an ambient sound level for the environment in the proximity of the device. The system then adjusts a volume setting for the device adaptively based on the determined ambient sound level. Adaptively adjusting the volume setting allows the device to adapt to its audio environment and ameliorates potentially-disruptive audio outputs.

Abstract: A controller for a headphone arrangement (101) comprises a drive circuit (203) which generates a signal for an earphone (105) from an audio signal. The drive signal is fed to the earphone (105) causing this to reproduce the audio signal. A first circuit (217) determines a signal level for the audio signal and a second circuit (209) determines an ambient sound level from a microphone signal from a microphone (109). A third circuit (211) determines an attenuated ambient sound level for the user from the microphone signal and an ambient sound attenuation of the earphone (105). A gain controller (205) controls the gain of the audio drive circuit (203) for the audio signal in response to the ambient sound level, the attenuated ambient sound level and the signal level. The dynamic and automated gain control may be used to reduce the risk of hearing damage e.g. by automatically restricting the sound level experienced by the user to the ambient sound level.

Abstract: The present invention concerns an audio amplifier circuit designed to provide an output signal to an audio transducer, said audio amplifier circuit comprising an audio power amplifier designed to receive an audio signal and designed to generate said output signal, a sensor designed to detect an audible sound having at least one noise component, to generate a detected signal. The audio amplifier circuit also includes a processing block configured to receive said detected signal at its input and to generate an off signal at its output, the latter being located at the input of said audio power amplifier. The processing block processes the detected signal according to said input signal to identify said noise component of said detected signal to generate a reference signal. The processing block generates the off signal when the value of said input signal is lower than the value of said reference signal.

Abstract: A microphone system has an output and at least a first transducer with a first dynamic range, a second transducer with a second dynamic range different than the first dynamic range, and coupling system to selectively couple the output of one of the first transducer or the second transducer to the system output, depending on the magnitude of the input sound signal, to produce a system with a dynamic range greater than the dynamic range of either individual transducer. A method of operating a microphone system includes detecting whether a transducer output crosses a threshold, and if so then selectively coupling another transducer's output to the system output. Some embodiments combine the outputs of more than one transducer in a weighted sum during transition from one transducer output to another, as a function of time or as a function of the amplitude of the incident audio signal.

Abstract: According to one embodiment, audio controlling apparatus includes first receiver configured to receive audio signal, second receiver configured to receive environmental sound, masking gain calculator configured to calculate masking gain for each frequency band, based on audio signal and environmental sound, and gain controller configured to smooth masking gain of frequency band that is less than preset threshold frequency in first interval, smooth masking gain of frequency band that is equal to or higher than threshold frequency in second interval that is longer than first interval, and thereby set gain.

Abstract: Systems and methods are disclosed that may be implemented to locally and remotely record, control, monitor and analyze sounds generated in an information handling system environment, e.g., such as in a server environment. Among other things, remote monitoring and observation of sound in a data center environment may be implemented to assist administrators with data center management and forensics analysis related to physical chassis events (e.g., such as chassis access and/or chassis intrusion), as well as providing live streaming, and local technician recorded sound bite capabilities. A sound policy may also be implemented involving server sound optimizations that works in conjunction with system thermal and power algorithms.

Abstract: A method and device are presented for identifying machines, such as vehicles, based on acoustic machine signatures. An acoustic sensor generates machine-acoustic data about the machine. A machine-signature identifier receives the machine-acoustic data, determines a machine signature from the machine-acoustic data, and identifies the machine based on the machine signature. A machine-signature database, configured to store multiple machine signatures and/or machine-signature templates, may receive and process queries about machine signatures for machine identification. The machine-signature identifier may generate and send an instruction based on the identified machine. The instruction may instruct application of electronic countermeasures or may permit a vehicle to enter a secured area. An acoustic modulator may generate the machine-acoustic data to act as a machine signature. The acoustic modulator may generate the machine signature using an ultrasonic carrier.

Abstract: Ambient noise sampled by a mobile device from a local environment is used to automatically trigger actions associated with content currently playing on the mobile device. In some implementations, subtitles or closed captions associated with the currently playing content are automatically invoked and displayed on a user interface based on a level of ambient noise. In some implementations, audio associated with the currently playing content is adjusted or muted. Actions can be automatically triggered based on a comparison of the sampled ambient noise, or an audio fingerprint of the sampled ambient noise, with reference data, such as a reference volume level or a reference audio fingerprint. In some implementations, a reference volume level can be learned on the mobile device based on ambient noise samples.

Abstract: The invention relates to a method for automated tuning of a sound system, the sound system comprising delay lines, equalizing filters, and at least two loudspeakers, the method comprising the steps of reproducing a useful sound signal through the loudspeakers, measuring sound pressure values at least one location, providing a target transfer function for tuning the delay lines and the equalizing filters of the sound system, the target transfer function representing a desired transfer characteristics of the sound system, adjusting the delay of the delay lines, and adjusting amplitude responses of the equalizing filters such, that the actual transfer characteristics of the sound system approximates the target function.

Abstract: Ambient sound is masked by receiving an audio signal, monitoring an acoustic signal that includes components of both the ambient sound the audio signal, determining a sound level sufficient to at least partially mask the ambient sound, and reproducing the received audio signal to produce an output acoustic signal at the determined sound level.

Abstract: The present invention features systems for dynamically adjusting audio signals by applying a gain to the signal in a spectrally varying manner to compensate for ambient noise, such that the sound is perceived to be unchanged in volume and spectral composition by the listener. The system obtains a threshold elevation for each frequency component by analyzing the spectral composition of the ambient noise. This threshold elevation is then used by a psychoacoustic model of hearing to determine an appropriate gain adjustment for the corresponding frequency component of the source signal which will make that source signal perceived by the human ear to be just as loud as if the noise were not present.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit uses the reference and/or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether one of the reference or error microphones is obstructed by comparing their received signal content and takes action to avoid generation of erroneous anti-noise.

Abstract: An electronic device including an audio module, a monitoring module and a fan is provided. The audio module generates an audio signal and an audio parameter according to audio data, and transmits the audio signal to a play device. Furthermore, the monitoring module determines a speed of fan rotation base on the type of the play device, the audio parameter and a sensed temperature. Then, a rotation speed of the fan is adjusted according to the determined speed of fan rotation.

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: Methods and apparatuses for comparing a level of a signal carrying ambient acoustic noise with a threshold level; and based on results of the comparison, ignoring time intervals identified as noise burst in estimating ambient noise levels.

Abstract: According to one embodiment, a signal processor includes a band divider, a first correction module, a determiner, an adjuster, and a second correction module. The band divider calculates an amplitude indicating sound pressure from an acoustic signal for each frequency range. The first correction module corrects the amplitude to change the dynamic range of the amplitude to a target range. The determiner determines whether the maximum amplitude of amplitudes corrected by the first correction module is equal to or larger than a threshold. The adjuster adjusts first and second coefficients based on the first condition if the maximum amplitude is equal to or larger than the threshold. The second correction module adds a value obtained by multiplying the acoustic signal by the first coefficient to a value obtained by multiplying the correction amount of the amplitude by the second coefficient, and outputs a resulting value as an output signal.

Abstract: Disclosed herein is a sound source recording apparatus and method adaptable to an operating environment, which can record a target sound source at a predetermined level without being affected by characteristics of the sound source or ambient noise. A target sound source is separated from a sound source signal received through an array of microphones and a recording sound pressure level and a gain are estimated using a reference sound pressure level and a reference distance for the target sound source, thereby controlling or adjusting the gain of the microphones.

Abstract: Certain aspects relate to providing an at least one audio source to at least one user. Certain aspects relate to selectively modifying an at least one first sound source to be provided to the at least one user, wherein the at least one first sound source is combined with an at least one second sound source, and wherein the selectively modifying is performed relative to the at least one audio source based at least in part on at least some specific information of the at least one first sound source. Other aspects relate to selectively modifying the at least one first sound source to be provided to the at least one user relative to the at least one second sound source based at least in part on at least some specific information of the at least one first sound source.

Abstract: The present invention features systems for adjusting audio signals by applying a gain to the signal in a spectrally varying manner to compensate for ambient noise in the environment of the listener. The system allows a listener to hear what ought to be heard, over the ambient noise, by applying a gain to the source that varies according to the spectral composition of the noise, rather than cancelling or filtering the noise. The spectral composition of the source is thus preserved in the listener's awareness without the removal of the noise signal. After application of these corrective gains to the source, the listener's perception of the source sound is as if the noise was not present. Systems may be incorporated into apparatuses including but not limited to mobile phones and music players.

Abstract: An information processing device includes: an output device configured to perform notification to a user by outputting ringing sound; a sound pickup device configured to pick up surrounding sound as ambient sound; an adaptive filtering process device configured to perform an adaptive filtering process using the picked-up ambient sound and the ringing sound output from the output device, to thereby extract, from the ambient sound, estimated environmental sound from which the ringing sound picked up by the sound pickup device has been removed; and a control device configured to control, on the basis of the feature quantity of a predetermined feature extracted from the estimated environmental sound, the adjustment of at least one of the sound volume and the sound quality of the ringing sound.

Abstract: The present invention relates to a signal processing apparatus and method, a program, and a data recording medium configured such that the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. An analyzer 21 generates mapping control information in the form of the root mean square of samples in a given segment of a supplied audio signal. A mapping processor 22 takes a nonlinear function determined by the mapping control information taken as a mapping function, and conducts amplitude conversion on a supplied audio signal using the mapping function. In this way, by conducting amplitude conversion of an audio signal using a nonlinear function that changes according to the characteristics in respective segments of an audio signal, the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. The present invention may be applied to portable playback apparatus.

Abstract: An earpiece (100) and a method (640) for acoustic management of multiple microphones is provided. The method can include capturing an ambient acoustic signal from an Ambient Sound Microphone (ASM) to produce an electronic ambient signal, capturing in an ear canal an internal sound from an Ear Canal Microphone (ECM) to produce an electronic internal signal, measuring a background noise signal, and mixing the electronic ambient signal with the electronic internal signal in a ratio dependent on the background noise signal to produce a mixed signal. The mixing can adjust an internal gain of the electronic internal signal and an external gain of the electronic ambient signal based on the background noise characteristics. The mixing can account for an acoustic attenuation level and an audio content level of the earpiece.

Abstract: A system comprising audio processing circuitry is provided. The audio processing circuitry is operable to receive combined-game-and-chat audio signals generated from a mixing together of a chat audio signal and game audio signals. The audio processing circuitry is operable to process the combined-game-and-chat audio signals to detect strength of a chat component of the combined-game-and-chat audio signals and strength of a game component of the combined-game-and-chat audio signals. The audio processing circuitry is operable to automatically control a volume setting based on one or both of: the detected strength of the chat component, and the detected strength of the game component. The combined-game-and-chat audio signals may comprise a left channel signal and a right channel signal. The processing of the combined-game-and-chat audio signals may comprise measuring strength of a vocal-band signal component that is common to the left channel signal and the right channel signal.

Abstract: A method and device are provided for the objective evaluation of voice quality of a speech signal. The device includes: a module for extracting a background noise signal, referred to as a noise signal, from the speech signal; a module for calculating the audio parameters of the noise signal; a module for classifying the background noise contained in the noise signal on the basis of the calculated audio parameters, according to a predefined set of background noise classes; and a module for evaluating the voice quality of the speech signal on the basis of at least the resulting classification relative to the background noise in the speech signal.

Abstract: A method and apparatus to evaluate a quality of an audio signal, in which the number of effective channels is determined for each of a reference signal of a current frame and a test signal indicative of the reference signal that has passed through an audio codec, and an audio quality evaluation score of the current frame is calculated by evaluating an audio quality of the current frame based on the determined number of effective channels for each of the reference signal and the test signal by means of a predetermined evaluator.

Abstract: An apparatus and method are disclosed, which separates ambient noise from a voice signal in a Bluetooth headset with dual microphones, switches the connection path between the dual microphones and automatically controls output audio gain, based on the ambient noise. The method for outputting the voice signal includes receiving/comparing the first and second input signals from the first and second microphones, extracting the voice signal of substantially identical intensity from the first and second input signals, and transmitting the voice signal from the portable terminal communicating with the Bluetooth headset.

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

Abstract: A method and system for enhancing a target sound signal from multiple sound signals is provided. An array of an arbitrary number of sound sensors positioned in an arbitrary configuration receives the sound signals from multiple disparate sources. The sound signals comprise the target sound signal from a target sound source, and ambient noise signals. A sound source localization unit, an adaptive beamforming unit, and a noise reduction unit are in operative communication with the array of sound sensors. The sound source localization unit estimates a spatial location of the target sound signal from the received sound signals. The adaptive beamforming unit performs adaptive beamforming by steering a directivity pattern of the array of sound sensors in a direction of the spatial location of the target sound signal, thereby enhancing the target sound signal and partially suppressing the ambient noise signals, which are further suppressed by the noise reduction unit.

Abstract: An audio signal attenuation system and method for detecting an audio emergency warning signal (or alarm) in a vehicle in which an audio signal is being played. Embodiments of the system and method make it easier for a police, fire, or other emergency alarm or siren to be heard in a loud or noisy listening environment when audio signal is being reproduced. This is achieved using selective frequency attenuation, which identifies a frequency of the alarm and then selectively attenuates the alarm frequency in the audio signal. Moreover, direction data that includes information about from which direction the alarm is coming can be used to selectively attenuate the alarm frequency in certain channels (or speakers) of the audio signal. In some embodiments, audio cues are used to alert the listener to the alarm signal and are adjusted based on alarm distance from the vehicle, speed, and the type of alarm.

Abstract: An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

Abstract: An earpiece and method to control a level of audio reproduction within an occluded ear canal are provided. The method includes the steps of measuring an ambient sound level from an Ambient Sound Microphone (ASM) at an entrance to an occluded ear canal, measuring a residual background noise level within an occluded ear canal from an Ear Canal Microphone (ECM), measuring a level of audio content delivered to the occluded ear canal, and adjusting a gain of the audio content so as to maintain reproduction of the audio content within a predetermined level range.

Abstract: A method and an arrangement for adapting an alert signal level generated by a device to an ambient of the device is provided. The method may include processing a sound signal corresponding to a test signal or an alert signal, generating a correlation value by correlating the sound signal with a generated sound signal, comparing the correlation value with a value, and adjusting the alert signal to a level based on result of the comparison.

Abstract: A system and method for enhancing two-way conversation between a user wearing at least one communication system or earphone and a second individual in proximity to the user is provided. The method includes detecting a spoken voice of the user from sounds captured at an ambient sound microphone, automatically activating a voice timer and setting a voice activity threshold for the voice timer responsive to detecting spoken voice, and adjusting a mixing gain of an audio content signal delivered to the earphone or communication system by way of the internal speaker with an ambient sound pass-through during activation of the voice timer. Separate mixing gains can be adjusted independently in accordance with spoken voice activity and reproduced audio content characteristics. Other embodiments are disclosed.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a communication device having a controller to monitor ambient noise in proximity to the communication device, monitor a transmitted voice signal associated with the communication device, adjust a receiver volume for the transmitted voice signal based at least in part on a comparison of the transmitted voice signal with the ambient noise, monitor a received voice signal associated with the communication device, and adjust a speaker volume for the received voice signal based at least in part on a comparison of the received voice signal with the ambient noise, where the adjusting of the receiver volume is independent of the adjusting of the speaker volume. Other embodiments are disclosed.

Abstract: Systems and methods for altering display output based on seat position are described. One embodiment of a method includes receiving desired frequency response data for a plurality of different audio outputs and a nominal seat position, determining a current seat position of a seat in the vehicle, and implementing a test sequence based on the seat position. Some embodiments include determining from the test sequence received frequency response data for the amplifier to utilize for the seat position, comparing the received frequency response data and the desired frequency response data, and in response to determining that the desired frequency response data does not substantially match the received frequency response data, altering settings on the amplifier such that the received frequency response data substantially matches the desired frequency response data.

Abstract: Smart notification appliances used in a mass notification system (MNS) have integrated software and distributed hardware for real time information to in-building, immediate vicinity and distributed recipients during emergency situations. Programmable control configurations in the smart notification appliances provide flexible installations. A distributed architecture system provides distributed intelligence in the smart notification appliances for maximum survivability and robust operation of the MNS. Audio messages are stored in each smart notification appliance with a one-to-one relationship to a speaker circuit. This configuration provides any circuit with either a live page or a plurality of preconfigured messages, in effect a multi-channel system limited only by the number of stored messages and controllable by software. Similarly, programmable light strobe intensity, flash rate and color may be controlled through the smart notification appliances.

Abstract: An audio device and a volume adjusting method are provided. The audio device includes a speed sensor, a first FIFO buffer unit, and a second FIFO buffer unit. The audio device generates audio signals by playing multimedia files and stores audio signals to the first FIFO buffer unit. The audio device collects ambient sound signals and stores collected sound signals to the second FIFO buffer unit. The audio device further analyzes stored audio signals and sound signals to determine a waveform of environmental noise signals, and determines a SPL of the environmental noise signals according to the determined waveform. The audio device then compares the determined SPL with a preset SPL and compares the sensed speed with a preset speed if the determined SPL is greater than the preset SPL, and adjusts the volume of audio signals according to a comparison result between the sensed speed and the preset speed.

Abstract: Devices and methods for digital sound leveling are disclosed. The subject devices and methods can be used to reduce the risk of noise induced hearing loss. According to an embodiment, an audio file can be parsed into frames that are filtered using an A-weighted digital filter and scaled to a desired power level. Accordingly, the dynamic range of an audio output at a particular volume level can be controlled.

Abstract: Provided are a method and an apparatus for encoding and decoding an audio signal. A method for encoding an audio signal includes receiving a transformed audio signal, dividing the transformed audio signal into a plurality of subbands, performing a first sinusoidal pulse coding operation on the subbands, determining a performance region of a second sinusoidal pulse coding operation among the subbands on the basis of coding information of the first sinusoidal pulse coding operation, and performing the second sinusoidal pulse coding operation on the determined performance region, wherein the first sinusoidal pulse coding operation is performed variably according to the coding information. Accordingly, it is possible to further improve the quality of a synthesized signal by considering the sinusoidal pulse coding of a lower layer when encoding or decoding an audio signal in an upper layer by a layered sinusoidal pulse coding scheme.

Abstract: Ambient audio event detection in a personal audio device headset provides for directive response to external audible events. Depending on the type of event, an alert may be issued, speech may be communicated to another device, program material may be interrupted and/or resumed with or without repositioning, and program material may be modified or selected for compatibility with, or to overcome, the ambient environment indicated by the detected event.

Abstract: An audio processing system that includes an audio filter having one or more elements capable of having state, such as a capacitor, an inductor or a delay. A saturation detector is configured to detect saturation of the audio filter and to generate an output when saturation of the filter is detected, such as a switch control signal. A switch is connected to the audio filter and the saturation detector, wherein the state of one or more of the elements of the audio filter is changed when the saturation detector provides the output to the switch, such as when the switch shorts the element and causes the energy stored in the element to be dissipated.

Abstract: A content reproduction device including: a microphone that collects noise in the surroundings of a casing; a feature amount extractor that extracts a plurality of feature amounts; a distance calculator that calculates an intervector distance between the extracted feature amount vector and a feature amount vector with the same dimensions which is set in advance as a feature amount of a waveform of a music signal; a determinator that determines whether or not music is included in the sounds collected by the microphone; a processor that processes the signal of the sounds collected by the microphone to change the volume or frequency characteristics of the sounds collected by the microphone; and an adder that adds and outputs the signal of the sounds collected by the microphone and the signal of sounds of reproduced content.

Abstract: An adaptive power control monitors the noise level within a primary acoustic signal and compares the noise level to a threshold. If the noise level is lower than the threshold, a noise suppression system is deactivated and bypass filtering and cross fading are enabled. If the noise level is higher than the threshold, the noise suppression system is activated.

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: Methods and apparatuses for deriving a signal-to-noise ratio based at least in part on a measured level of a signal carrying far-end speech, and a measured level of a signal carrying ambient acoustic noise; determining a target gain adjustment based at least in part on the derived signal-to-noise ratio; applying the target gain adjustment to the signal carrying far-end speech to produce a gain-adjusted signal; and providing the gain-adjusted signal for audio output from a communications device.

Abstract: In a system and method for maintaining the spatial stability of a sound field a background noise estimate may be estimated for each of a first signal and a second signal. A first gain coefficient may be calculated responsive to the first audio signal and the background noise estimate of the first audio signal. A second gain coefficient may be calculated responsive to the second signal and the background noise estimate of the second signal. The gain coefficients may be calculated using one or more gain coefficient calculators. A common gain coefficient may be selected from one of the first gain coefficient and the second gain coefficient. The selected common gain coefficient may be one that results in a least amount of audio signal modification and may be applied to each of the first signal and the second signal.

Abstract: An apparatus and method are disclosed, which separates ambient noise from a voice signal in a Bluetooth headset with dual microphones, switches the connection path between the dual microphones and automatically controls output audio gain, based on the ambient noise. The method for outputting the voice signal includes receiving/comparing the first and second input signals from the first and second microphones, extracting the voice signal of substantially identical intensity from the first and second input signals, and transmitting the voice signal from the portable terminal communicating with the Bluetooth headset.

Abstract: A method and vehicle-based communication system are provided that control a remote microphone by determining that one or more of the remote microphone and a user of the remote microphone is in a field of view (FOV) of a video camera and, in response, instructing the remote microphone to configure itself to receive ambient audio. In various embodiments, the remote microphone may configure itself, or be explicitly instructed to configure itself, to receive ambient audio by adjusting one or more of a beam forming or omni-directional pattern, potentially including noise cancellation algorithms to facilitate reception of ambient audio in contrast to user directed audio. When the one or more of the remote microphone and a user of the remote microphone no longer is in a FOV of the video camera, the method and vehicle-based communication system may instruct the remote microphone to reconfigure itself to receive user directed audio.

Abstract: Comfort noise, such as can be used in voice communications can be generated using methods in the frequency domain and/or in the time domain. In various embodiments, a comfort noise spectrum can be generated in the frequency domain as the product of a background noise sample and a random noise sample. In other embodiments, the comfort noise can be generated directly in the time domain as the convolution of a background noise sample and a random noise sample.