Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: In some implementations, a user can specify an audio control curve for an audio track. The audio track can include an audio signal that is sampled at a specified rate. The audio control curve can be used to generate an audio control signal that can be sampled at the same rate as the audio signal. The audio signal and the audio control signal can be combined to produce sample-accurate adjustments of the audio signal. In some implementations, the audio control curve can correspond to gain adjustments for the audio signal. In some implementations, the audio control curve can be used to adjust the amplitude of the audio signal at particular frequencies. The audio control curve can be used to adjust treble or bass levels during the playback of an audio track, for example.

Abstract: An audio signal output device (1) includes an audio signal output portion (15), a volume setting information acquisition portion (15), and a control portion (10). The control portion is configured to calculate the output volume of an external device on the basis of volume information of content and volume setting information of the external device and perform control of lowering the output volume of the external device in the case where the output volume is higher than a prescribed volume threshold.

Abstract: An electronic device includes a volume adjustment activating module, a distance detection module, and an adjusting module. The volume adjustment activating module is configured for activating the distance detection module to measure distances between a volume adjusting member and the distance detection module. The distance detection module includes at least one distance sensor, the at least one distance sensor is configured for detecting distances between the volume adjusting member and the distance detection module to obtain a plurality of distance measure values. The adjusting module is configured to adjust a volume of the electronic device according to the distance measure values.

Abstract: The application discusses a computer implemented method and apparatus for performing audio equalisation in an audio receiver device, such as an integrated receiver/decoder or set top box, or integrated TV, connected to one or more audio playback devices, such as a television unit, computer screen and speakers, amplifier or home theatre equipment. The method and apparatus use an equalisation process which compares audio signals received in different audio formats (e.g. MPEG-1 Layer II, AC-3 2.0, AC-3 5.1 and HE-AAC) with one another, allowing a correction gain factor to be determined for equalising the perceived loudness of the signals when played-back at a connected playback device. The correction gain factor is then applied in the audio receiver device before output.

Abstract: The present invention relates to a process for adjusting the sound volume of a digital sound recording characterised 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 apparatus for adaptively controlling the audio output (122) of a communication device (114) according to the noise characteristics of the receiver listening environment (119). An output volume for the communication device (114) is set by a user (e.g., listener 118). The communication device (114) can intermittently sample ambient noise (116) of its environment (119). A minimum signal to noise threshold can be established for audio output (122). A total adjustment for the audio output (122) is established based on the ambient noise (116), the user set output volume, and the minimum signal to noise threshold. The total adjustment is a product of a software volume adjustment (230) and a hardware gain adjustment (240). The software volume adjustment (230) and the hardware gain adjustment (240) is adaptively applied when the communication device (114) outputs audio (122).

Abstract: This is directed to controlling the output of audio based on the mode of an electronic device. In particular, this is directed to an electronic device having a sound mode, silent mode, and mute mode controlling the output of audio along different audio output routes. In the sound mode, all audio can be output, while only registered or authorized audio can be output in the silent mode. In the mute mode, no audio can be output. The sound and silent modes can be enabled using an interface of the device (e.g., a switch having two possible states), while the mute mode can be enabled using an accessory coupled to the device. To disengage the mute mode, a user can provide a corresponding instruction using the accessory, or providing an instruction on the device related to volume control. For example, a user can change the device volume using an on-device interface, or toggle a sound/silent switch to a sound mode.

Abstract: A noise level of an output signal is maintained low. An adjacent interference detecting unit compares a signal level of a wide band signal which is a desired station signal of a relatively wide band and a signal level of a narrow band signal which is a desired station signal of a relatively narrow band, and judges that adjacent interference has occurred when the signal level of the narrow band signal is less than or equal to a predetermined level even though the signal level of the wide band signal is greater than or equal to a predetermined level. A variable amplifier (36) attenuates an L?R signal and forms a monophonic signal when the signal level of the narrow band signal is lower than a predetermined level. With this process, when there is adjacent interference, a signal level of the narrow band signal at which the attenuating unit starts attenuating the L?R signal is changed to a relatively high value.

Abstract: The invention provides one sound producing system including a speaker, a driving circuit and an audio controller. The speaker is used for producing an audible sound. The driving circuit is connected to the speaker. The audio controller is configured to receive an input audio signal and pre-amplify the input audio signal with an amplify gain to obtain a pre-amplified audio signal; operate a multiband dynamic range control on the pre-amplified audio signal to obtain an output audio signal; convert the output audio signal to a driving voltage; provide the driving voltage to the speaker through the driving circuit; detect over the driving circuit to obtain at least one excursion parameter to determine an estimated excursion of the speaker in response to the driving voltage; and adjust the amplify gain according to the estimated excursion.

Abstract: Systems and methods are operable to adjust an output audio volume level of presented audio content that is output as sound from one or more speakers in a media presentation environment. An exemplary embodiment detects movement of a user who is listening to the audio content; increases the output audio volume level when the detected user movement is away from a predefined location in the media presentation environment; and decreases the output audio volume level when the detected user movement is closer to the predefined location in the media presentation environment.

Abstract: The present disclosure involves a method of automatic volume adjustment based on context and location. A determination is made that a mobile computing device has been placed in a vehicle. The determination may be made by detecting that the mobile computing device has been plugged into a dock inside the vehicle, or by detecting that a movement speed of the mobile computing device is within a normal speed range of the vehicle. The mobile computing device has a programmable audio output volume. In response to the determination, the audio output volume of the mobile computing device is automatically increased to a predefined audio output volume. In some embodiments, the volume setting of the mobile computing device is recorded before the volume is automatically increased, and the mobile computing device restores the original volume setting after the mobile computing device is taken out of the vehicle.

Abstract: A power generation circuit and process for providing electrical power and data transfer by means of a module attached to the audio port of a mobile electronic device. Power and data may be exchanged with an auxiliary device, which may be a sensor or a data processing device. A description of several possible power generation circuits is provided, as well as a description of different sensors and data processing devices which may be attached to a power generation circuit. The power generation circuit is particularly designed to provide a low interference and low voltage ripple output with high total power availability and efficiency.

Abstract: A speaker array control method includes steps of detecting a position of an audience located in front of a speaker array, wherein the speaker array includes N speakers and N is a positive integer larger than one; defining a target and a non-target with respect to an i-th speaker of the N speakers according to the position of the audience, wherein i is a positive integer smaller than or equal to N; calculating a weighting vector for the i-th speaker according to the target and the non-target; adjusting a directionality of an output signal of the i-th speaker by the weighting vector and reducing energy of a plurality of side lobes of the output signal of the i-th speaker; and controlling the i-th speaker to output the adjusted output signal when the energy of each of the side lobes is smaller than a threshold.

Abstract: A computer program product, method, apparatus, and system determine location data of a performance. Further, the computer program product, method, and system provide the location data to an audio adjustment device that calculates a distance-based audio adjustment based upon a difference between a position of the performance indicated by the location data and a position of a speaker that is coupled to receive an audio signal from the audio adjustment device. The audio adjustment device adjusts at least one characteristic of the audio signal based upon the distance between the performance and the speaker.

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: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: A rendering device is configured to dynamically vary how audio and visual information is rendered by a rendering device responsive to changes in distance between the rendering device and a mobile device carried by a user. As the user moves about, the distance between the rendering device and the mobile device carried by the user will vary. As the distance between the mobile device and the rendering device changes, the rendering system can dynamically adapt how the audio and visual information is rendered. For example, the rendering device can increase or decrease the volume of an audio output, depending on changes in the distance. Also, the rendering device can change how visual information is rendered on a display responsive to changes in distance. In some embodiments, when the distance between the mobile device and rendering device exceeds a predetermined distance, the audio and/or visual information may be sent to the mobile device.

Abstract: A control terminal apparatus includes a transmission unit transmitting a control signal to audio output apparatuses, a display unit, an operation detector detecting an operation associated with display content, and a controller displaying individual volume setting sections which correspond to the audio output apparatuses and which include operation members performing variable operations of volume settings while volume setting states are displayed, displaying a master volume setting section including an operation member performing collective variable operation while the volume balance of the audio output apparatuses is maintained, executing clear display of a setting changeable range in which the volume balance is maintained in the master volume setting section, generating, when the operation detector detects an operation performed on one of the individual volume setting sections or the master volume setting section, a control signal corresponding to content of the operation, and causing the transmission unit

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: Mechanisms for controlling an audio level of an HDMI audio system are provided, the mechanisms comprising: causing audio data to be presented by an HDMI audio system at a current system volume level; receiving a requested volume level from a second screen device; and controlling a system volume level by: (a) determining the current system volume level; (b) determining a change in volume based on a difference between the requested volume level and the current system volume level; (c) determining a direction in which to cause the system volume level to change; (d) sending a volume control message to the system using a consumer electronic control bus connected to the system indicating whether to increase or decrease the system volume level based on the determined direction of system volume change; and (e) repeating (a)-(d) until the current system volume level reaches a predetermined value.

Abstract: A method, an apparatus for adjusting volume and an electronic device using the same is provided. The method comprises: receiving a volume adjusting signal; detecting a current application scene based on a display content of the electronic device after receiving the volume adjusting signal; displaying a volume adjusting interface, the volume adjusting interface including an volume adjusting area which corresponds to the detected current application scene and a selecting area for indicating whether the volume adjusting signal is applied to all application scenes of the electronic device; adjusting volume based on the volume adjusting signal and applying the adjusted volume to the detected current application scene in case that a selected state of the selecting area is in a first state or applying the adjusted volume to all application scenes of the electronic device in case that the selected state of the selecting area is in a second state.

Abstract: A computer-implemented method of dynamic range control is disclosed. The method includes at a device with a display, displaying a volume (relative loudness level) control to control the volume level of an output audio signal of the device, the volume control including a dynamic resizable window control for controlling the dynamic range of the output audio signal. A method for adjusting dynamic range of an audio signal is also disclosed. The method includes providing an input audio signal with a first dynamic range, mapping the first dynamic range to a second dynamic range using a transfer function with a linear portion aligned to an average level of the input audio signal, and generating an output audio signal with the second dynamic range from the input audio 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: A method and an apparatus for controlling/adjusting volume, which can set multiple volume modes having different volume intervals and control volume by a volume interval of a corresponding volume mode according to a given circumstance in an electronic device having a volume change function. The method includes detecting a volume change event; when the volume change event is a volume mode change, changing a current volume mode to a volume mode corresponding to the volume change event; and when the volume change event is a volume value change, controlling volume up/down in response to the volume change event.

Abstract: A sound reproducing device having: a sound source data storage unit configured to store reference sound source data for reproducing a reference sound having a prescribed frequency band and comparative sound source data for reproducing comparative sounds having frequency bands different from each other; a reproduction processing unit configured to reproduce the reference sound and the comparative sounds on the basis of the reference sound source data and the comparative sound source data stored in the sound source data storage unit; a comparative sound volume adjustment operation unit configured to receive sound volume adjustment operation for the comparative sounds; and an acoustic characteristic information generation unit configured to simultaneously output the reference sound and a comparative sound of a sound volume in accordance with the received sound volume adjustment operation and record a result of the sound volume adjustment operation in association with the comparative sound related to reproduction

Abstract: A gain controlling system, a sound playback system, and a gain controlling method thereof are disclosed. The gain controlling system includes a main gain control unit, a sub gain control unit, and a logic control unit. The main gain control unit has a first step-by-step adjusting magnitude; the sub gain control unit has a second step-by-step adjusting magnitude; wherein the second step-by-step adjusting magnitude is smaller than the first step-by-step adjusting magnitude. The logic control unit is used for controlling the main and the sub gain control unit to transform an analog signal into a converted signal according to an adjustment command signal and further determining whether an adjusting magnitude required by the adjustment command signal is larger than a maximum gain range of the sub gain control unit. If yes, the logic control unit controls the main control unit and the sub gain control unit repeatedly.

Abstract: A system for processing signals to enhance patient audibility of a plurality of signals in an MRI environment is provided. The system includes an acoustic measuring device for measuring sound power levels generated by the MRI and a principal frequency component identifier for identifying principal frequencies measured by the acoustic measuring device. The system also includes an audio equalizer for controlling the amplitude and frequency of each of the plurality of signals in accordance with the principal frequencies. Further provided by the system is an attenuator for attenuating an overall sound level of the signals being processed and a dynamic range compression processor.

Abstract: A sound signal processing device that is capable of suitably extracting main sound from mixed sound in which unnecessary sound (for example, leakage sound and reverberant sound) is mixed with the main sound. More specifically, a mixed sound signal in the time domain including first sound and second sound, and a target sound signal in the time domain including sound corresponding to at least the second sound, which have temporal relation in their entirety or in part, are each divided into a plurality of frequency bands. A level ratio between the two signals is calculated at each frequency. Based on the level ratio, a signal of the first sound that is included in the mixed sound signal is extracted.

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 disclosed apparatus extracts particular instrument sounds included in music in real time during music reproduction. The audio processing section 12 has an attack decision section 42 that decides attack positions of the bass drum sounds in audio signals, a bass drum sounds decision section 43 that decides sound periods of the bass drum in the audio signals, and a bass drum sounds extraction section 44 that extracts bass drum sounds using a time varying type filter for the attack positions and an adaptive type filter for the sound periods.

Abstract: The invention is directed to systems, methods and computer program products for adjusting audio properties after recording an audio. An exemplary method includes recording audio using an audio capturing system, wherein the recorded audio is associated with an audio level; determining one or more audio signal properties associated with the recorded audio; comparing the audio level with a first predetermined threshold level; and in response to determining the audio level is greater than a first predetermined threshold level, adjusting the recorded audio based at least partially on modifying the one or more determined audio signal properties, wherein the adjusting step is performed a finite time interval following the recording step.

Abstract: The application describes a method and an apparatus to prevent clipping of an audio signal when protection against signal clipping by received audio metadata is not guaranteed. The method may be used to prevent clipping for the case of downmixing a multichannel signal to a stereo audio signal. According to the method, it is determined whether first gain values (4) based on received audio metadata are sufficient for protection against clipping of the audio signal. The audio metadata is embedded in a first audio stream (1). In case a first gain value (4) is not sufficient for protection, the respective first gain value (4) is replaced with a gain value sufficient for protection against clipping of the audio signal. Preferably, in case no metadata related to dynamic range control is present in the first audio stream (1), the method may add gain values sufficient for protection against signal clipping.

Abstract: A microphone array device includes a first sound reception unit configured to obtain a first sound signal that is input from a first microphone, a second sound reception unit configured to obtain a second sound signal that is input from a second microphone, a noise state evaluation unit configured to compare the first sound signal and the second sound signal and to obtain an evaluation parameter to evaluate an influence of a non-target sound included in the second sound signal on a target sound included in the first sound signal according to a result of the comparison, a subtraction adjustment unit configured to set a suppression amount for the second sound signal based on the evaluation parameter and to generate a third sound signal; and a subtraction unit configured to generate a signal to be output based on the third sound signal and the first sound signal.

Abstract: A method for enhancing speech includes extracting a center channel of an audio signal, flattening the spectrum of the center channel, and mixing the flattened speech channel with the audio signal, thereby enhancing any speech in the audio signal. Also disclosed are a method for extracting a center channel of sound from an audio signal with multiple channels, a method for flattening the spectrum of an audio signal, and a method for detecting speech in an audio signal. Also disclosed is a speech enhancer that includes a center-channel extract, a spectral flattener, a speech-confidence generator, and a mixer for mixing the flattened speech channel with original audio signal proportionate to the confidence of having detected speech, thereby enhancing any speech in the audio signal.

Abstract: The present disclosure refers to a digital microphone device providing a single-bit Pulse Density Modulation PDM output signal. The digital microphone comprises a microphone, arranged to convert an acoustic input signal into an analog electrical signal, and a preamplifier, having a variable gain, arranged to receive the analog electrical signal and to provide an amplified analog electrical signal, depending on the variable gain. The variable gain depends on a gain control signal. The digital microphone further comprises an Analog-to-Digital Converter block, arranged to receive the amplified analog electrical signal and to convert it into a respective digital signal; and a compensation block, arranged to receive the digital signal and to perform a digital operation on such digital signal, on the basis of a compensation signal, to generate a compensated signal.

Abstract: There is provided a method for audio signal adjustment comprising: determining a loudness estimate for at least one frame of an audio signal; determining a level value for the at least one frame based on the loudness estimate; and adjusting the audio signal based on the level value. An apparatus for carrying out the method is also provided.

Abstract: Systems and methods to deliver live sound with enhanced quality conveyed by a mixing desk to a mobile computing device user contemporaneously with external live sounds. The mobile computing device is operable to receive enhanced audio signals produced by a remote audio signal processing device through a communication work. A memory resident application is configured to playback the enhanced audio signals in phase with the external sounds using an audio rendering device. An attendee at the live event can hear the sounds from the playback through an earphone coupled to the mobile computing device as well as from the ambient environment.

Abstract: A system includes a plurality of sources configured to provide a plurality of audio channels, and a control system including a processor, and a computer-readable storage media having computer-executable instructions embodied thereon. When executed by the processor, the computer-executable instructions cause the processor to adjust a master volume, identify a channel volume type associated with a first audio channel of the plurality of audio channels, and determine a channel volume associated with the first audio channel based at least partially on at least one of the adjustment of the master volume and the channel volume type.

Abstract: Integrated processing of multimedia signals can eliminate unnecessary signal processors and converters without losing the functionality of typical home entertainment system components. The integrated multimedia system includes a main player that captures and processes signals digitally. The main player may adjust the audio signal to provide audio output of equal loudness across all frequencies by accounting for sensitivity of the human ear for sounds of varying frequencies. The main player can also account for perceived differences in loudness based on the angle of a listener to a speaker by detecting the position of a user and making an adjustment accordingly. The invention further provides a speaker that has embedded performance characteristics or an identifier that allows the system to provide an optimal speaker driving current for a particular system or determine how that speaker would be best implemented in the integrated system.

Abstract: A method by an electronic device for controlling a frequency response of audio output includes: receiving an audio signal at the device; estimating a sound pressure level of the audio signal based on one or more attributes or settings of the electronic device and/or the audio signal; generating values of an adaptive loudness control curve along a range of frequencies, wherein the adaptive loudness control curve is generated based on a difference between values of an equal loudness curve at the estimated sound pressure level along and values of an equal loudness curve at a reference sound pressure level; filtering the audio signal using values of the adaptive loudness control curve; and controlling output of the filtered audio signal as an audio output having substantially the same loudness along the range of frequencies.

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 for automatic volume adjustment performed by a physical computing system includes, with the computing system, determining a virtual proximity between a user and a number of applications running on the computing system, the virtual proximity based in part on a type of one of the number of applications, and with the computing system, automatically adjusting a volume of the number of applications linked with a sound peripheral connecting to the computing system, the volume of each of the number of applications being based on the virtual proximity.

Abstract: In one example, a technique may include outputting, by a computing device associated with a user and for playback at a first volume level by an audio output device, first audio data, receiving, by the computing device, audio input data, and responsive to determining, by the computing device, that the audio input data includes speech associated with an entity different from the user, determining, by the computing device and based at least in part on the audio input data, whether to output second audio data. The method may also include, responsive to determining to output the second audio data: determining, by the computing device, a second volume level lower than the first volume level, and outputting, by the computing device and for playback at the second volume level by the audio output device, the second audio data.

Abstract: An apparatus and method for managing a sound volume of a wireless connection device in a mobile communication terminal are provided. The method includes pre-storing a sound volume level of the wireless connection device with respect to each system supported by the mobile communication terminal, and when a call is generated, requesting to determine a sound volume by transmitting to the wireless connection device a sound volume level of a system in which the call is generated among the sound volume levels pre-stored for the respective systems.

Abstract: An emergency message output amplifier includes a plurality of audio output channels. Channels can be combined together to increase output power, and can be isolated from respective output speakers in response to detected faults and then disabled. The functional amplifier can be activated by itself and output current can be monitored to lower the output level enough to enable the amplifier to properly drive the existing load.

Abstract: An ultrasonic speaker system has a first variable gain block, a modulation block, a second variable gain block, an ultrasonic actuator, and a controller, wherein the controller controls gain of the first variable gain block so that a modulation degree of a modulated signal becomes a predetermined value, and when the controller raises the gain of the first variable gain block, the controller lowers the gain of the second variable gain block, while when the controller lowers the gain of the first variable gain block, the controller raises the gain of the second variable gain block.

Abstract: A signal processing method for enhancing the dynamic range of a signal is disclosed. The method comprises: a) forming an attenuated signal from an input signal; b) filtering each of the input and the attenuated signals such that the sum of their bandwidths is less than or equal to the bandwidth of a transmission channel; c) modulating a first one of the filtered input signal and the filtered attenuated signal, whereby the filtered input signal and the filtered attenuated signal occupy respective non-overlapping frequency ranges within the bandwidth of the transmission channel; and d) combining the modulated signal with the second one of the filtered input signal and the filtered attenuated signal to form a composite output signal.