Abstract: The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Abstract: In accordance with embodiments of the present disclosure, a system may include a controller configured to be coupled to an audio speaker, wherein the controller receives an output signal indicative of a physical quantity associated with the audio speaker, compares the output signal to an audio input signal to determine if differences between the output signal and the audio input signal are present indicating at least one of distortion of the output signal, non-linearities of the audio speaker, and overexcursion of the audio speaker, and controls an audio signal communicated from the controller to the audio speaker and based on the audio input signal responsive to determining that differences between the output signal and the audio input signal are present indicating at least one of distortion of the output signal, non-linearities of the audio speaker, and overexcursion of the audio speaker.

Abstract: The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Abstract: There is provided a system for use in an environment, the system comprising a microphone, a plurality of speakers, a memory storing an executable code, and a processor executing the executable code to receive a media content including an audio component, play the audio component of the media content over at least one of the plurality of speakers, detect, while playing the audio component, an ambient sound in the environment using the microphone, and adjust playing of the audio component of the media content for the at least one of the plurality of speakers based on the ambient sound in the environment.

Abstract: A method of altering audio output from an electronic device based on image data is provided. In one embodiment, the method includes acquiring image data and determining one or more characteristics of the image data. Such characteristics may include sharpness, brightness, motion, magnification, zoom setting, and so forth, as well as variation in any of the preceding characteristics. The method may also include producing audio output, wherein at least one characteristic of the audio output is determined based on one or more of the image data characteristics. Various audio output characteristics that may be varied based on the video data characteristics may include, for instance, pitch, reverberation, tempo, volume, filter frequency response, added sound effects, or the like. Additional methods, devices, and manufactures are also disclosed.

Abstract: A noise control method and device are provided that relate to the field of noise control. The noise control method includes: acquiring noise information of an ambient environment; judging whether the noise information satisfies a predetermined condition, and if yes, acquiring a first message from a first device; and judging, according to the first message, whether the noise information is related to the first device, and if yes, sending a second message to the first device, the second message being used to notify the first device to adjust a volume. The noise control method and device in the embodiments of the present application may easily and quickly realize noise control over a specific device, thereby improving user experience.

Abstract: Automatic gain control systems disclosed herein can incorporate a confidence metric that can estimate the accuracy of gain adjustments calculated by an automatic gain control module. The confidence metric may be based on a percentage of valid audio samples in a given period of time. Based on the confidence metric, the AGC response may be reduced, delayed, frozen, or otherwise altered from the baseline gain adjustment. Time-averaging process may be used to estimate the input signal power level and determine an appropriate baseline gain adjustment. Additionally, weighting functions can be adjusted to prevent overestimation of the signal power.

Abstract: A noise control method and device are provided that relate to the field of noise control. A noise control method includes: acquiring noise information of an ambient environment; and judging whether the noise information satisfies a predetermined condition, and if so, sending a noise control message to another device, the noise control message being used to notify the other device to adjust a volume. Another noise control method includes: receiving, by a device, a noise control message from an external device; and adjusting a volume based on a volume adjustment policy according to the noise control message and a current volume of the device. The noise control method and device in the embodiments of the present application may easily and quickly realize control over ambient noise, thereby improving user experience.

Abstract: A method and apparatus for processing audio signals in an entertainment system from at least one audio source modify audio signals during playback by the entertainment system for adjustment to a psychoacoustic loudness set value, wherein this modification is performed in each case on the basis of an average psychoacoustic loudness maximum determined over a predefined time interval for the audio source concerned.

Abstract: Vehicle operational data can be acquired in response to receiving a vehicle occupant voice command. A voice input can be received from a vehicle occupant while a vehicle is in operation. The received voice input can be analyzed to determine whether a vehicle diagnostic command is included in the voice input. In response to determining that a vehicle diagnostic command is included in the received voice input, one or more vehicle sensors can be caused to acquire vehicle operational data based on the vehicle diagnostic command. The vehicle operational data can be presented to a user, such as a driver, a passenger, or a review entity.

Abstract: An audio system encodes and decodes audio captured by a microphone array system in the presence of wind noise. The encoder encodes the audio signal in a way that includes beamformed audio signal and a “hidden” representation of a non-beamformed audio signal. The hidden signal is produced by modulating the low frequency signal to a high frequency above the audible range. A decoder can then either output the beamformed audio signal or can use the hidden signal to generate a reduced wind noise audio signal that includes the non-beamformed audio in the low frequency range.

Abstract: A headset having a talk-through microphones incorporates an audio circuit that disconnects or compresses a signal representing sounds detected by the talk-through microphones in response to the audio circuit detecting the onset of a peak in the signal that exceeds a predetermined voltage level, and that does so with a rate of change in voltage level that exceeds a predetermined rate of change in voltage level. The duration of the disconnection or compression may be controlled by a timing circuit set to a predetermined period of time that may be retriggerable while amidst the predetermined period of time.

Abstract: Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

Abstract: This specification describes, among other things, a computer-implemented method. The method can include receiving a stream of audio data at a computing device. The stream of audio data can be segmented into a plurality of audio segments. Respective intensity levels are determined for each of the plurality of audio segments. For each of the plurality of audio segments and based on the respective intensity levels, a determination can be made as to whether the audio segment includes a speech signal. Selective gain control can be performed on the stream of audio data by automatically adjusting a gain of particular ones of the plurality of audio segments that are determined to include a speech signal.

Abstract: A device-adaptable headset device is provided, comprising: a microphone; one or more speakers; an audio connector comprising: a microphone output pin; and one or more speaker input pins respectively connected to the one or more speakers; a microphone line between the microphone output pin and the microphone; a ground line; one or more input devices in communication with the microphone line; a first circuit and a second circuit respectively configured for active signalling and passive signalling on the microphone line; and, a processor configured to: determine a present signalling mode; when the signalling mode is active, and when the one or more input devices is actuated, generate active signals on the microphone line using the first circuit; and, when the signalling mode is passive, and when the one or more input devices is actuated, generate passive signals on the microphone line using the second circuit.

Abstract: Systems and methods for leveling loudness variation in an audio signal are described. Embodiments use both a perceptual leveling algorithm and a standards-based loudness measure together to minimize audio process artifacts and ensure that the measured loudness of the processed audio is close to a required measure, according to a particular standard measurement of loudness. These systems and methods can be used either offline or in real-time.

Abstract: Systems and methods for multi-sourced noise suppression are provided. An example system may receive streams of audio data including a voice signal and noise, the voice signal including a spoken word. The streams of audio data are provided by distributed audio devices. The system can assign weights to the audio streams based at least partially on quality of the audio streams. The weights of audio streams can be determined based on signal-to-noise ratios (SNRs). The system may further process, based on the weights, the audio stream to generate cleaned speech. Each audio device comprises microphone(s) and can be associated with the Internet of Things (IoT), such that the audio devices are Internet of Things devices. The processing can include noise suppression and reduction and echo cancellation. The cleaned speech can be provided to a remote device for further processing which may include Automatic Speech Recognition (ASR).

Abstract: A mobile electronic device (e.g., mobile phone) includes an acceleration sensor, a communication module, and a controller. The controller detects a moving state of an own device based on an acceleration detected by the acceleration sensor. When detecting that the own device in the moving state is stopped, the controller causes the communication module to search for the base station after the lapse of a first period of time. When detecting that the own device in the moving state is stood still, the controller causes the communication module to search for the base station before the lapse of the first period of time.

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: There is provided a system for use in an environment, the system comprising a microphone, a plurality of speakers, a memory storing an executable code, and a processor executing the executable code to receive a media content including an audio component, play the audio component of the media content over at least one of the plurality of speakers, detect, while playing the audio component, an ambient sound in the environment using the microphone, and adjust playing of the audio component of the media content for the at least one of the plurality of speakers based on the ambient sound in the environment.

Abstract: Intuitive methods of automatically configuring a remote control for multiple electronic devices are disclosed. The remote control can be automatically configured with the help of a first electronic device that is connected to one or more additional electronic devices. The first electronic device aids in the configuration of the remote control by gathering information about the one or more additional electronic devices and configuring the remote control in accordance. The information about the one or more additional electronic devices may be gathered from the devices themselves, from additional remote controls associated with the devices, and/or from a user, among other possibilities.

Abstract: Devices for monitoring a sound pressure level of sound generated by a speaker. For example, the device may sample voltages at various points along a path of an audio signal, determine a current associated with the audio signal, and receive or infer an efficiency of the speaker. The device may then determine the sound pressure level based on the voltage, the current, and the efficiency to more accurately monitor the sound pressure level exposure of a user.

Abstract: The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Abstract: In some examples, sound verification may include a speaker device that may be configured to transmit sound at a dynamic volume level and a listening device that may be configured to receive the sound and provide feedback to the speaker device based on the received sound. The primary transceiver device may be further configured to adjust the dynamic volume level based on the feedback provided by the secondary transceiver device.

Abstract: A system and a method for determining one or more wave characteristics from a moving platform are disclosed. A sonar system, such as an Acoustic Doppler Current Profiler, can profile the water motion relative to the platform, and an earth reference can determine a measure of the platform motion relative to a fixed earth reference. Both water profile and earth reference measurements can be synergistically employed to compensate for motion of the platform. Directional wave spectra and non-directional wave spectrum can be computed and translated via linear wave theory to surface height spectra and used to calculate characteristics, such as significant wave height, peak period, and peak direction.

Abstract: An apparatus comprises a receiver (203) receiving a multichannel signal comprising two channels for rendering by a first speaker (101) at a first position and a second speaker (103) at a second position respectively. A first signal generator (207) generates a correlated signal and a second signal generator (209) generates an uncorrelated signal from the multichannel signal, the signals comprising respectively correlated and uncorrelated signal components for the channels. A receiver (201) receives a microphone signal from the microphone (107). A first correlator (213) determines a first correlation signal from a correlation of the microphone signal and the correlated signal, and a second correlator (215) determines a second correlation signal from a correlation of the microphone signal and the uncorrelated signal. A position estimator (219) estimates a position of the microphone from the first and second correlation signals.

Abstract: A method and an electronic device of controlling an audio output are disclosed. The method includes detecting an audio output device connected to an electronic device, determining a type of audio output device associated with the audio output device connected to the electronic device, determining a first audio output level associated with the audio output device connected to the electronic device based on the type of audio output device and a predetermined audio output level, and outputting an audio output at the first audio output level using the audio output device.

Abstract: A method can include applying a patterned mask over a semiconductor structure, the semiconductor structure having a dielectric layer, forming using the patterned mask a material formation trench intermediate first and second spaced apart metal formations formed in the dielectric layer, and disposing a dielectric material formation in the material formation trench.

Abstract: The application relates to a loudspeaker system including an input unit providing an electric input audio signal; an equalization unit for modifying said electric input audio signal in dependence on frequency and to provide an equalized electric audio signal according to a predefined equalization function, a loudspeaker unit for converting said equalized electric audio signal to an acoustic output sound signal, and a user interface for modifying a volume level of said output sound signal in a multitude (N) of steps V0, V1, . . . , VN. The application further relates to a method communication device comprising the loudspeaker system and to its use. The present application provides an improved loudspeaker system in which the equalization unit is configured to apply a specific equalization function EQ0, EQ1, . . . , EQN to the electric input audio signal in each of said multitude of steps V0, V1, . . . , VN of the volume level.

Abstract: An information processing apparatus includes a learning unit which learns sound level manipulation relating to a sound level of a terminal device used by a user based on information relating to the user. The information processing apparatus includes a determination unit which determines a reproduction mode at the time when the terminal device reproduces an audio contained in a content according to a learning result of the learning unit. The information processing apparatus includes a controller which controls an audio contained in the content in the reproduction mode determined by the determination unit.

Abstract: A selectively coordinated audio playback system is disclosed. The audio playback system includes multiple audio players serving, receiving and rendering audio content audio content. A controller is in communication with the audio players and controls the audio content provided on each audio player. The controller includes a touchscreen user interface to receive user inputs and is configured to selectively coordinate the audio content from the multiple audio players according to gestures detected via the touchscreen interface.

Abstract: The application discusses a computer implemented method and apparatus for performing audio equalization 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 theater equipment. The method and apparatus use an equalization 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 equalizing 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: A multi-channel audio system that can provide a variable sub-sample delay between two or more audio channels. In one embodiment, a variable timing clock generator generates multiple clock signals where each may have different phase, and the clock generator can vary the phase difference, in accordance with a sub-sample delay setting input. These clock signals are used by respective digital-to-analog converters (DACs) to convert the digital audio channels into analog form. In another embodiment, a variable delay block is added to an oversampling DAC, on a per channel basis. Other embodiments are also described and claimed.

Abstract: A method on a mobile device for voice recognition training is described. A voice training mode is entered. A voice training sample for a user of the mobile device is recorded. The voice training mode is interrupted to enter a noise indicator mode based on a sample background noise level for the voice training sample and a sample background noise type for the voice training sample. The voice training mode is returned to from the noise indicator mode when the user provides a continuation input that indicates a current background noise level meets an indicator threshold value.

Abstract: A sound field measuring device (1) includes an external output unit (6) configured to output a measurement signal composed of a periodic function having a code length of 2n?1 (n is a natural number) to a speaker (9), a microphone (7) configured to pick up the measurement signal outputted from the speaker (9), a Fourier transform unit (12) configured to obtain frequency characteristics by Fourier transforming measurement sound picked up with a sample length of 2m (m is a natural number), a thinning-out unit (13) configured to remove line spectra except for the (k×2m-n+1)th line spectra (k=0, 1, 2, and the like) from the obtained frequency characteristics, and an averaging unit (14) configured to obtain averaged frequency characteristics of a sound field on the basis of frequency characteristics thinned-out.

Abstract: Systems, methods, apparatus, and articles of manufacture to control audio playback devices are disclosed. An example first playback device includes a speaker driver, a processor, and a computer readable medium including a set of instructions that, when executed by the processor, cause the first playback device to implement a method. The example method includes receiving, from a first audio information source, first audio information. The example method includes playing back the first audio information. The example method includes receiving, from a second audio information source, (i) a first message, and (ii) second audio information.

Abstract: A noise cancellation system, comprising: an input for a digital signal, the digital signal having a first sample rate; a digital filter, connected to the input to receive the digital signal; a decimator, connected to the input to receive the digital signal and to generate a decimated signal at a second sample rate lower than the first sample rate; and a processor. The processor comprises: an emulation of the digital filter, connected to receive the decimated signal and to generate an emulated filter output; and a control circuit, for generating a control signal on the basis of the emulated filter output. The control signal is applied to the digital filter to control a filter characteristic thereof.

Abstract: An information processing method, electronic apparatus, and a system are described where the method includes establishing a connection with a second electronic apparatus; receiving a first control instruction from the second electronic apparatus; parsing the first control instruction, to obtain an adjusting parameter for an audio adjusting unit; adjusting a gain for the audio adjusting unit based on the adjusting parameter; and processing and outputting audio information received from the second electronic apparatus based on the adjusted gain.

Abstract: An apparatus and method for outputting audio signals of a portable communication device are provided. The apparatus includes a plurality of audio tables including different volume control values or tone control values, and allows a user to selectively set audio tables corresponding to each of audio paths equipped with the portable communication device. Further, the apparatus analyzes a surrounding environment during a voice call, determines the absence or presence of noise, and suitably controls a volume or a tone of audio signals based on the result of the determination before outputting the audio signals. Accordingly, the apparatus can provide an improved calling condition in various calling environments.

Abstract: Large numbers of similar-looking or identical physical controls on an audio control surface can make it difficult for operators to rapidly locate desired controls. Highlighting of selected controls focuses the operator's visual attention on controls the operator is manipulating, or has just manipulated and is likely to manipulate again. Controls peripheral to or otherwise related to manipulated controls may also be highlighted. Highlighting methods include the use of control illumination intensity, color, and labeling display text. Visual focusing may include highlighting relationships between channel strips or between individual physical controls, or between the physical controls of a surface and one or more digital audio workstations.

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 M RI 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 program asset is retrieved. The program asset has audio encoded at a first loudness setting and includes metadata specifying the first loudness setting. Dialog of the audio is identified. The loudness of the dialog is determined. The determined loudness is compared to the first loudness setting. The program asset is re-recorded at a second loudness setting corresponding to the determined loudness, if the first loudness setting and the determined loudness are different by more than a predetermined amount.

Abstract: A user terminal apparatus connected to an electronic device, and a method thereof, are provided. The apparatus includes a communicator that performs communication with the electronic device, an audio processor that processes an audio source to be transmitted to the electronic device, and a controller that adjusts a sound level of the audio source based on a reference volume unit of the electronic device, and controls the adjusted audio source to be transmitted to the electronic device in a streaming method.

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: The present disclosure provide a volume display method and apparatus for smoothly and flexibly displaying a volume change on the premise of ensuring high time effectiveness and accuracy. The method includes: acquiring a volume value of a first audio input during a current time cycle and a volume value of a second audio input during a previous time cycle; determining volume values to be displayed according to the volume values of the first audio input and the second audio input as well as a volume unit or a threshold of the quantity of volume values to be displayed; determining, according to the quantity of the volume values and duration of the time cycle, a unit display time of volume values to be displayed; sequencing all volume values to be displayed; and successively displaying an image that corresponds to each volume value on a device screen according to the unit display time and the sequence.

Abstract: Provided are, among other things, systems, methods and techniques for loudness-based audio-signal compensation. According to one such system, an input line accepts an input audio signal; a loudness estimator, coupled to the input line, processes the input audio signal to obtain an audio playback loudness level and also provides a production loudness level for the input audio signal; and a filter generator/selector coupled to an output of the loudness estimator provides an audio compensation filter based on the production loudness level and the audio playback loudness level. The input signal is processed by the audio compensation filter to provide an output signal.

Abstract: A system and a method for mixing at least two audio signals are provided that comprise transferring the audio signals with respective transfer functions, the audio signals each having an amplitude and a phase; adding the audio signals to provide an output signal representative of the mixed audio signals, the output signal having an amplitude and a phase; controlling at least one of the transfer functions of the signal lines so that the phase of the output signal is adapted to the phase of the audio signal with a higher signal strength than the other audio signal(s), the signal strengths corresponding to the amplitudes of the audio signals.

Abstract: A level of an input signal is detected according to a first following rate, and a first level signal indicating the detected level is generated. A level of the input signal is detected according to a second following rate lower than the first following rate, and a second level signal indicating the detected level is generated. One of the first and second level signals is selected based on a relation (e.g., ratio) between the first and second level signals so that a gain is determined based on the selected one of the first and second level signals. The level of the first input signal is adjusted according to the determined gain. For example, if the level variation is dominant, the gain adjustment suitable for the level variation can be performed, whereas, if the stable level is dominant, the gain adjustment suitable for the stable level can be performed.

Abstract: An audio normalization gain value is applied to an audio signal to produce a normalized signal. The normalized signal is processed to compute dynamic range control (DRC) gain values in accordance with a selected one of several pre-defined DRC characteristics. The audio signal is encoded, and the DRC gain values are provided as metadata associated with the encoded audio signal. Several other embodiments are also described and claimed.

Abstract: Presently described are systems and methods for identifying a mute/sound attribute of an audio sample set having first-channel samples and second-channel samples. An embodiment takes the form of a method that includes: (i) receiving the audio sample set; (ii) for each first-channel sample, determining a first max amplitude and a first min amplitude; (iii) calculating a first span based on a first function of the first max amplitude and the first min amplitude; (iv) for each second-channel sample, determining a second max amplitude and a second min amplitude; (v) calculating a second span based on a function of the second max amplitude and the second min amplitude; and (vi) identifying the audio sample set as having a sound attribute if both the first span and the second span are greater than a min-volume threshold.