Устройство и способ повышения разборчивости речи для респираторной маски

В настоящем документе описано устройство повышения разборчивости речи и респираторные маски, содержащие устройство повышения разборчивости речи, а также способы повышения разборчивости передачи речи для пользователя респираторной маски. В одном или более воплощениях устройство и способы повышения разборчивости речи, описываемые в настоящем документе, обнаруживают акустическую энергию в пределах первого диапазона частот внутри чистого воздушного пространства респираторной маски и доставляют компенсирующую акустическую энергию наружу из этого чистого воздушного пространства с использованием динамика. В одном или более воплощениях указанная компенсирующая акустическая энергия обладает таким предварительно определенным профилем ослабления амплитуды, что компенсирующая акустическая энергия имеет амплитуду менее чем на 6 дБ больше, чем указанный профиль акустического ослабления корпуса маски по меньшей мере в 90% предварительно определенного диапазона ослабляемых частот. 2 н. и 21 з.п. ф-лы, ...

КОНТРОЛЛЕР ВЫРАВНИВАТЕЛЯ ГРОМКОСТИ И СПОСОБ УПРАВЛЕНИЯ

Настоящее изобретение в целом относится к обработке звуковых сигналов. Описаны контроллер выравнивателя громкости и способ управления. В одном варианте осуществления контроллер выравнивателя громкости содержит классификатор звукового содержимого для идентификации типа содержимого звукового сигнала в реальном времени и регулирующий блок для регулировки выравнивателя громкости в непрерывном режиме в зависимости от идентифицированного типа содержимого. Регулирующий блок может выполняться с возможностью положительной корреляции коэффициента динамического усиления выравнивателя громкости с типами информативного содержимого звукового сигнала и отрицательной корреляции коэффициента динамического усиления выравнивателя громкости с типами мешающего содержимого звукового сигнала. Изобретение направлено на обеспечение возможности автоматически настраивать устройства улучшения качества звука в непрерывном режиме в зависимости от воспроизводимого звукового содержимого. 3 н. и 8 з.п. ф-лы, 41 ил., 2 ...

TONVERSTAERKERANORDNUNG MIT EINER VON DER MOTORDREHZAHL ABHAENGIGEN LAUTSTAERKEREGELUNG

Volume control counteracting noisy surroundings - alters dynamic characteristics or volume in accordance with background noise level

The volume control, for making acoustic signals clearly audible in noisy environments, compresses the signal's dynamic characteristics and/or increases its volume depending on background noise. The background noise is measured by one or more transducers which then control circuitry altering the dynamic characteristics and/or volume. The degree to which compression and/or volume are affected by the background noise can be adjusted by a special control knob.

Loudspeaker or headphone automatic level adjustment - surrounding noise is picked up and converted to control electric signal

Loudspeaker etc. is energised, directly or indirectly, by a voltage (UL) obtained by addition of the input signal voltage (US) to a voltage (UP) which is proportional to the production of the incoming signal voltage (US) and a mean value of a rectified voltage (UM) obtained by picking up the back ground noise and converting to electric signal. The microphone (3) is fitted within the the room in the neighbourhood of the window to pick-up the interference coming mostly from traffic. The output (UM) from the microphone amplifier (4) reaches the analogue multiplier (7) via a rectifier (5) and a low pass filter (6) which forms the arithmetical mean value.

Verfahren und Einrichtung zur automatischen Regelung der Lautstärke eines Lautsprechers abhängig vom Störpegel der Umgebung

Umgebungsgeräuschabhängige Lautstärkeregelung für Nachrichten in einem Fahrzeug

VERFAHREN ZUM AUTOMATISCHEN EINSTELLEN EINES AUDIOFREQUENZGANGS ZUR BESSEREN VERSTÄNDLICHKEIT

Method for suppressing feedback in the case of hall or large-field audio systems and circuit arrangement for carrying out the method

In an audio system having an adaptively controllable compander and a microphone picking up interfering noise, the signals of which are used for controlling the characteristic of the compander (3) in order to suppress feedback effects, several microphones (Mi) are installed at a distance from the speaker or transmitting microphone (M1) at locations at which a higher sound pressure occurs or can be expected due to direct radiation, reflection, interference or room resonances or echoes and the compander characteristic is displaced in the direction of higher attenuation in dependence on the microphone signal level (ni) of the microphones (Mi). ...

Verfahren und Einrichtung zur Anpassung von Ausgangssignalen eines Tonwiedergabegerätes an den Geräuschpegel im Fahrgastraum eines Kraftfahrzeuges

Verfahren zur Anpassung von Ausgangssignalen eines Tonwiedergabegerätes (8) an den Geräuschpegel im Fahrgastraum eines Kraftfahrzeugs mit der Möglichkeit, am Tonwiedergabegerät (8) die Lautstärke sowie die Höhen und/oder Tiefen der Ausgangssignale manuell zu beeinflussen, bei dem unter Nutzung eines Fahrgeschwindigkeitssignals (V) ein Tonfrequenzgang der Ausgangssignale automatisch variiert wird, wobei nahezu der gesamte Tonfrequenzgang auf der Grundlage von fahrzeugspezifischen Kennlinien in Verbindung mit dem Fahrgeschwindigkeitssignal (V) sowie ggf. mit weiteren Betriebsparametern (n) variiert wird, wobei Schalldruckamplituden (L) der einzelnen Ausgangssignalfrequenzen (f) immer um einen aktuell zu berechnenden Wert über den Amplituden (L) ausgewählter gleichfrequenter Störgeräusche liegen, dadurch gekennzeichnet, daß der Öffnungsgrad (A) von verschließbaren Karosserieöffnungen und/oder Kennzahlen (G), die den Betriebszustand einer Lüftungs- oder Klimaanlage definieren, als Grundlage ...

Announcement outputting method for e.g. public place, involves determining environment volume at place or in area before output of announcement, and performing recording of events at place and/or within area by camera

The method involves determining an environment volume at a place or in an area before an output of an announcement. Verification is made whether the environment volume is too high to understand the announcement, or whether the environment volume is small enough to understand the announcement. An output device (AG) is connected with a control room (LW). The announcement is outputted when the environment volume is small enough to understand the announcement. Recording of events at the place and/or within the area is performed by a camera (C).

Howling frequency component emphasis method and apparatus

AUDIO-VISUAL REPRODUCTION

Automatic volume control for cellular telephones with hands-free operation

A method of controlling the volume of a telephone during hands-free operation is described. A background noise level is sensed and averaged over a predetermined period. The volume of the hands-free speaker is controlled in dependence upon the noise level sensed, so that the volume is raised if the average noise value is higher than the set volume level.

Dynamic volume adjustment in a slot machine

Control for vehicle sound output

Adjusting audio output in a physical listening environment, preferably the cabin of a vehicle, so as to create a simulated environment. Preferably the simulated environment includes a configurable venue which can be altered in size by the user interacting with a user interface (UI). Preferably the perceived distance from a stage of the configurable venue can be altered as well as the volume and the perceived positioning of the user in the configurable venue. Preferably the UI comprises a first control and a second control which are both displayed as concentric circles. Each circle being associated with the size of the configurable venue and the perceived distance from the virtual stage of the configurable venue respectively. Preferably there are further controls for changing the treble, bass balance and volume. Preferably there exists a third control in the shape of an arc. Preferably the colour of the controls changes based on the current settings. Preferably internal vehicle lights also ...

Speech clarity

Transient detection for speaker distortion reduction

Audio distortion by a speaker may be reduced by detecting onset audio events within an audio signal and modifying the audio to reduce the audio distortion perceived by a listener. The onsets may be detected using a psych-acoustic model by determining critical sub-band powers and corresponding masking thresholds. When a loudness value calculated from the CSBs and masking thresholds exceeds a threshold level, certain frequency bands may be attenuated and other frequency bands may be amplified. The audio modification may be performed on a frame by frame basis and each frame may be processed multiple times until the onset is sufficiently masked or attenuated. Transients are detected in distortion producing frequency bands which are typically low frequencies 202, the distortion producing bands may be attenuated 204, distortion masking bands may also be amplified 206 to reduce perceived distortion.

Audio signal amplifier circuit

An audio signal amplifier circuit for an audio signal reproducer for use in noisy surroundings, the audio signal at (10) fed to the reproducer having undergone volume compression, in which the amplifier circuit contains a volume expander (20), the expansion provided by the expander being controllable by a noise signal, such as from a microphone at (12), in such a way as to increase with decreasing noise amplitude. The expansion is additionally controlled by the audio signal amplitude by X which feeds via a function circuit (23) comprising two D/A converters which address a ROM or PROM which via a D/A converter provides an analog volume range signal to FET's (28). Filters (16) and (32) divide the audio and noise signals into a number of frequency bands and a volume expander (20) is provided for each band. The noise signal of a lower frequency band having a reduced amplitude is fed via an amplifier (40) and the volume expander of the adjacent higher frequency band. ...

AUTOMATIC VOLUME CONTROL SYSTEM

A method and system for noise cancellation

AUDITORY SUBLIMINAL MESSAGE SYSTEM

Gain control for an audio signal

Input gain adjusting apparatus and method

In a voice operated gain adjust for a voice processor included an input range adjust (10), a sixteen channel amplifier (12), an analog to digital converter (14) and a microcomputer (16). An audio signal including voice signals and background noise is applied to the input of the range adjust (10) and the amplifier (12). The output of the input range adjust (10) is applied through the A/D converter (14) to the microcomputer (16). The microcomputer (16) distinguishes voice signals from background noise and controls the gain of the amplifier (12) in accordance with the amplitude of the voice signals. Also, the microcomputer (16) controls the gain of the input range adjust (10) so that the input signal to the A/D converter (14) is within the operating range of the converter (14).

SOUND BROADCASTING EQUIPMENT

Gain control for intra-band carrier aggregation

Gain control for intra-band carrier aggregation

Gain control for intra-band carrier aggregation

PROCEDURE AND DEVICE FOR THE DYNAMIC SOUND OPTIMIZATION

AUDIO PROCESSING DEVICE AND - PROGRAM

ARRANGEMENT FOR THE SUPPRESSION OF UNWANTED SIGNALS

PROCEDURE AND DEVICE FOR THE ABSORPTION OF NOISE IN AN INPUT SIGNAL

CIRCUIT FOR INTERFERENCE COMPENSATION.

AUDIOVISUAL PLAYBACK UNIT

PROTECTIVE CIRCUIT AGAINST ELECTROMAGNETIC NOISE

SYSTEM FOR THE INCREASE OF THE LANGUAGE QUALITY

AUDIO CHARACTERISTIC ADJUSTMENT

A cable tie

Abstract A cable tie having: a head and tail which releasably engage with each other; an elongated flexible strap extending between the head and the tail; a releasing means; and a locking means; the cable tie arranged such that when in use and when it has been securely fixed to a surface a user is able to move the cable tie between an open configuration, a holding configuration which holds at least one cable and a fastened configuration which fastens at least one cable, the arrangement also being such that when the cable tie is in the fastened configuration the locking means prevents it from moving into the open configuration, the releasing means also enables a user to move the cable tie from the holding or fastened configuration into an open configuration so that a user can add more cables. WO2020/167142 PCT/NZ2020/050009 Figure 1 SUBSITUTE SHEET (Rule 26) ...

Radio transceiver

Method and apparatus for automatically adjusting speaker and microphone gains within a mobile telephone

AUTOMOTIVE AUDIO CONTROLLER WITH VIBRATION SENSOR

An automotive audio controller is provided for receiving an audio signal from an audio source and providing a filtered audio signal that may help compensate for ambient noise in the automobile. The audio controller may have a vibration sensor that generates a vibration signal indicative of vibrations in the automobile. A dynamic filter of the audio controller may receive the vibration signal and the audio signal, and change the gain level and/or tone of the audio signal based on the vibration signal to generate the filtered audio signal. The audio controller also may receive a speed signal, and change the gain level azldlor tone of the audio signal based on the speed signal to generate the filtered audio signal.

APPARATUS FOR THE DETECTION OF VOICE SIGNALS

APPARATUS FOR THE DETECTION OF VOICE SIGNALS A detection apparatus for receiving an input signal containing noise and for detecting the presence of voice signals therein comprises means for attenuating the input signal and a comparator of the amplitude of the attenuated input signal relative to a variable threshold voltage. This threshold voltage is generated by control means which make the threshold voltage dependent on the peak amplitude value of the input signal. A control signal corresponding to the detection of voice signals is supplied by decision means which carry out a time analysis of the output signal of the comparator.

AUDIO APPARATUS FOR VEHICLE

An audio apparatus for a vehicle, comprising: means for generating an audio signal; amplifying means operatively connected to said generating means for amplifying an audio signal received from said generating means; sound emitting means for emitting sounds and which is driven by said amplyfying means; means for detecting vehicle speed; and gain control means operatively connected to said speed detecting means for generating a control signal and providing it to said amplifying means, such that said control signal increases the gain of the ampifying means according to an increase in the output of said speed detecting means, wherein the output of said amplifying means, or the sound volume, is kept constant if the vehicle speed is below speed V1 where V1 is a predetermined, extremely low speed, and the output of said amplifying means is increased as the vehicle speed is increased above V1, and wherein said output of said amplifying means increases at a faster rate when the vehicle speed is ...

AUDIO SIGNAL

A computer device (100) for processing audio signals is described. The computer device (100) includes at least a processor and a memory. The computer device (100) is configured to receive a bitstream comprising a combined audio signal, the combined audio signal comprising a first audio signal including speech and a second audio signal. The computer device (100) is configured to compress the combined audio signal to provide a compressed audio signal. The computer device (100) is configured to control a dynamic range of the compressed audio signal to provide an output audio signal. In this way, a quality of the speech included in the output audio signal is improved.

SOUND MANAGEMENT METHOD AND SYSTEM

A computer implemented method for managing a sound emitting device comprising: receiving data associated with operation of the sound emitting device at a predetermined location; processing said data to determine an operating characteristic of that device for that location; comparing the operating characteristic with a predetermined mathematical relationship to determine whether a difference exists; and identifying an input adjustment to correct the difference wherein the input adjustment optionally is within a predetermined range and optionally does not exceed a predetermined maximum increment; wherein the predetermined mathematical relationship is between an input variable and an output variable in respect of the sound emitting device.

A COMPUTER-IMPLEMENTED METHOD FOR REDUCING CROSSTALK IN A COMPUTER-BASED AUDIOMETER

A computer-implemented method for reducing undesired crosstalk signals on an inactive channel of a device comprising the steps of: (i) determining system volume levels and associated signal amplitudes required to achieve a range of desired audio output attenuation levels on an active channel of the device; (ii) determining a crosstalk compensation signal comprising a signal amplitude and associated phase shift required to reduce undesired crosstalk on the inactive channel of the device for each desired audio output attention level in the range of desired audio output attenuation levels; and (iii) generating a desired audio output attenuation level on the active channel of the device by generating a signal at the determined system volume level and associated signal amplitude required to achieve said desired audio output attenuation level, and generating a contemporaneous crosstalk compensation signal on the inactive channel of the device by generating a signal at the determined signal amplitude ...

SYSTEMS AND METHODS FOR GENERATING A VOLUME-BASED RESPONSE FOR MULTIPLE VOICE-OPERATED USER DEVICES

Systems and methods are provided herein for responding to a voice command at a volume level based on a volume level of the voice command. For example, a media guidance application may detect, through a first voice-operated user device of a plurality of voice- operated user devices, a voice command spoken by a user. The media guidance application may determine a first volume level of the voice command. Based on the volume level of the voice command, the media guidance application may determine that a second voice-operated user device of the plurality of voice-operated user devices is closer to the user than any of the other voice-operated user devices. The media guidance application may generate an audible response, through the second voice-operated user device, at a second volume level that is set based on the first volume level of the voice command.

IMPROVED HEARING AID DEVICES WITH REDUCED BACKROUND AND FEEDBACK NOISES

A personal sound processing apparatus is disclosed that comprises an earphone; a microphone; an electronic sound processor connected between an automatic gain controller (AGC) and a digital volume controller, an automated feedback tracker operatively connected to the microcontroller, and an audio signal amplifier to amplify audio signals which level is controlled by the digital volume controller. The microcontroller is configured to execute electronic instructions for recognizing at least one of a feedback noise condition and a background noise condition in the audio signals and to send a control signal to the digital volume controller to adjust volume level in response to presence of the at least one of the feedback noise condition and the background noise condition.

RADIO WITH MESSAGE RECEPTION INDICATOR

Elektroakustische Übertragungsanlage für lärmgestörte Wiedergabeorte.

Elektroakustische Übertragungsanlage für lärmgestörte Wiedergabeorte.

Tonwiedergabesystem mit durch das Umgebungsgeräusch geregelter Lautstärke

DEVICE FOR AUTOMATIC GAIN CONTROL FOR A TO RECEIPT UNIT OF PULSE-MODULATED RADAR EQUIPMENT A BELONGING ZF-AMPLIFIER.

SWITCHING CONFIGURATION FOR ADJUSTING THE VOLUME OF A CAR RECEIVER AS A FUNCTION OF THE DRIVING NOISE.

SWITCHING CONFIGURATION FOR THE AUTOMATIC ADAPTATION OF THE VOLUME AT LEAST ONE LOUDSPEAKER TO A TO A LOUDSPEAKER PLACE DOMINANT ONE INTERFERENCE LEVEL FOR BROADCAST RECEIVERS.

MECHANISM FOR TONWIEDERGABE IN AN AREA WITH AN INDEPENDENT ACOUSTIC SOURCE.

Adaptive control of electrical signal amplification involves using Wiener filter, deriving normalized gain factor from estimated useful signal, estimated noise signal

The method involves feeding the signal (d(n)) to a Wiener filter whose normalized gain factor is computed by deriving an estimated useful signal from the smoothed power of the electrical signal and then deriving the estimated noise signal power and the normalized gain factor from a stated equation from these parameters. An Independent claim is also included for an application of the method to a hands-free device.

Volume control apparatus for an audio system.

Die Erfindung betrifft eine Lautstärken-Regeleinrichtung für eine Audioanlage, welche eine Klangquelle aufweist. Die erzeugten Signale derselben sind zu zwei Primärwicklungen (24, 24´) eines Signalübertragers (21) leitbar, bei dem eine Mess- und Auswerteeinrichtung (28 A , 28 B ) der Lautstärke integriert ist. Es sind wenigstens eine zwischen den Primärwicklungen (24, 24´) einsetzbare Schalteinrichtung (30) mit einer Anzahl von Widerständen, die durch Schalter zuschaltbar oder überbrückbar sind, vorgesehen. Zudem ist eine elektronische Steuervorrichtung (26) mit einem Mikroprozessor (27) vorhanden, welche die Schalter der Schalteinrichtung (30) in Abhängigkeit der gemessenen Lautstärke derart betätigt, dass die festlegbare maximale Lautstärke der Audioanlage nicht oder nur kurzzeitig überschritten bzw. reduziert wird. Damit kann die Lautstärke auf analoger Basis praktisch ohne Veränderung seiner Form reguliert werden.

Voice acquisition method and device for intelligent wearable equipment, and related components

Automatic volume and dynamic range adjustment method and device for mobile audio devices

DISPOSITIF DE REGLAGE AUTOMATIQUE DU NIVEAU SONORE D'UNE SOURCE DE SONS UTILES EN FONCTION DU BRUIT AMBIANT

REGLAGE AUTOMATIQUE DU NIVEAU D'UNE SOURCE SONORE EN FONCTION DU BRUIT AMBIANT. DISPOSITIF DE REGLAGE AUTOMATIQUE DU NIVEAU SONORE D'UNE SOURCE 11 DE SONS UTILES EN FONCTION DU BRUIT AMBIANT, CONSTITUE, D'UNE PART, D'UN CIRCUIT EMETTEUR 21, COMPORTANT PRINCIPALEMENT UN AMPLIFICATEUR 23 A GAIN VARIABLE EN SERIE AVEC UN HAUT-PARLEUR 22, UN PREMIER CIRCUIT REDRESSEUR 31 EN DERIVATION SUR LE CIRCUIT EMETTEUR 21, ENTRE L'AMPLIFICATEUR 23 A GAIN VARIABLE ET LE HAUT-PARLEUR 22, FOURNISSANT UN PREMIER SIGNAL CONTINU S1 ET, D'AUTRE PART, D'UN CIRCUIT CAPTEUR 41, COMPORTANT PRINCIPALEMENT UN DEUXIEME CIRCUIT REDRESSEUR 51 EN SERIE AVEC UN MICROPHONE 42 FOURNISSANT UN DEUXIEME SIGNAL CONTINU S2. SELON L'INVENTION, LE PREMIER SIGNAL CONTINU S1 ET LE DEUXIEME SIGNAL CONTINU S2 SONT APPLIQUES RESPECTIVEMENT A UNE PREMIERE BORNE 62 D'ENTREE ET UNE DEUXIEME BORNE 63 D'ENTREE D'UN CIRCUIT COMPARATEUR 61 QUI DELIVRE A UNE BORNE DE SORTIE 64 UN SIGNAL DE COMMANDE DU GAIN DE L'AMPLIFICATEUR 23 A GAIN VARIABLE ...

APPARATUS FOR COMPENSATING FOR AMBIENT NOISE IN AN AUDIO AMPLIFIER

PROCESS OF ADJUSTMENT Of a NOISE LEVEL IN a Mobile phone

Sound information reproduction system - has two compressor units connected in series between radio receiver and power amplifier

DEVICE INTENDS TO AMPLIFY ACOUSTIC SIGNALS AND PROVIDED WITH MEANS TO REMOVE UNDESIRABLE INTERFERING SIGNALS

DISPOSITIF PERMETTANT L'AJUSTAGE AUTOMATIQUE DU NIVEAU DES SOURCES SONORES UTILES DANS LES MILIEUX A BRUYANCE VARIABLE

Dispositif permettant l'ajustage automatique du niveau des sources sonores utiles dans les milieux à bruyance variable. Il comprend un amplificateur linéaire à gain commandé piloté par une tension n.VB proportionnelle à la pression acoustique partielle de bruit. Cette tension est obtenue par calcul analogique en temps réel de la fonction : n.VB = n.(VM**2 -VS**2 )1/2 dans laquelle VM est une tension proportionnelle à la pression acoustique totale recueillie par un micro et VS une tension de signal recueillie aux bornes du haut-parleur. Le dispositif permet de conserver le rapport des intensités sonores signal/bruit dans des limites préalablement établies. L'invention est destinée à l'équipement des voitures automobiles et des installations sonores situées au voisinage d'aéroports, de lignes SNCF, de voies à grande circulation. (CF DESSIN DANS BOPI) ...

Device for analogue conditioning of vocal signal in chain of digital processing of speech in noisy environment, comprises automatic gain control of amplifier

L'invention concerne un dispositif de conditionnement analogique (CDA) d'un signal vocal dans une chaîne de traitement numérique de la parole équipée d'un alternat « appuyer pour parler » (ALT). Le dispositif de conditionnement comprend des moyens de pré-amplification (PAMP) du signal d'entrée (MIC), des moyens d'amplification (AMP) à gain variable du signal pré-amplifié (PMIC), délivrant un signal de sortie (CANMIC) vers un convertisseur analogique/ numérique (CAN) de la chaîne et des moyens de contrôle automatique du gain (CAG) des moyens d'amplification (AMP). Selon l'invention, les moyens de contrôle du gain (CAG) comprennent notamment des moyens (DET) de détection et de mémorisation du niveau crête en valeur absolue du signal pré-amplifié (PMIC) pendant la durée d'un appui de l'alternat, des moyens de calcul (LOG) à partir dudit niveau mémorisé, du gain à appliquer aux moyens d'amplification (AMP) et des moyens de validation (VAL) permettant l'autorisation, au moment du relâché de ...

PERFECTIONNEMENT AUX DISPOSITIFS PERMETTANT L'AJUSTEMENT DU NIVEAU DES SOURCES SONORES UTILES DANS LES MILIEUX A BRUYANCE VARIABLE

DISPOSITIF D'ELABORATION DE LA TENSION DE COMMANDE DES AMPLIFICATEURS A GAIN COMMANDES PERMETTANT L'AJUSTEMENT DU NIVEAU DES SOURCES SONORES UTILES DANS LES MILIEUX A BRUYANCE VARIABLE CARACTERISE PAR LE FAIT QU'IL COMPORTE UN MOYEN POUR COMPARER UNE TENSION FONCTION DU BRUIT AMBIANT A UNE TENSION FONCTION DU SIGNAL UTILE ET PAR LE FAIT QUE LE RAPPORT SIGNAL UTILEBRUIT AMBIANT EST CONSERVE DANS DES LIMITES PREALABLEMENT DEFINIES.

AUTO-CALIBRATION OF THE VOLUME LEVEL IN AN AUDIO SIGNAL CAPABLE OF SUITABLY CONTROLLING THE GAIN OF AN AUDIO OUTPUT SIGNAL

PURPOSE: An auto-calibration of the volume level in an audio signal is provided to dynamically determine the amount of sound of an audio input signal based on a signal statistics of a sound input signal or the combination of the same. CONSTITUTION: An audio signal analysis unit(30) determines the sound volume of an audio input signal by using a signal statistics and the psychological-sound localization model of sound of a person. A signal controller(40) which includes a gain control unit(41) and a delay element(42) receives the input of an audio input signal(19). A gain which is determined by the gain control unit is controlled through a user interface(50). COPYRIGHT KIPO 2012 ...

Method and apparatus for audio intelligibility enhancement and computing apparatus

Method and apparatus for audio intelligibility enhancement and computing apparatus are provided. The method includes the following steps. Environment noise is detected by performing voice activity detection according to a detected audio signal from at least a microphone of a computing device. Noise information is obtained according to the detected environment noise and a first audio signal. A second audio signal is outputted by boosting the first audio signal under an adjustable headroom by the computing device according to the noise information and the first audio signal.

Audio coding selection based on device operating condition

A sensor is configured to determine at least one operating condition of a device and a selector is configured to select an audio coding process for the device, based on the operating condition. The operating condition may include remaining battery life of the device and/or ambient noise level. The selected audio coding process may consume less power than another possible audio coding process during audio processing. The audio may include voice and/or audio playback, e.g., music playback.

CONTROLLING AN OUTPUT WHILE RECEIVING A USER INPUT

While an output is presented to a user, an audio input that can include spoken input from the user is monitored. Presentation of the output is controlled while monitoring the audio input based on the monitoring. In the case of an audio output, the presentation can be controlled by attenuating the audio output according to the monitoring of the audio input. For example, a level of the audio output is reduced for continued presentation to the user after a desired signal is detected in the audio input. The output can include a prompt soliciting an input from a user, and the monitoring can include detecting the user's spoken input in the input audio, for example, estimating a certainty that the audio input includes the user's spoken input, or that such spoken input is in a desired grammar, such as in a desired list of commands or phrases. The approach is also applicable to video outputs.

NOISE MODIFYING OVERHEAD AUDIO SYSTEM

An overhead audio system for use in a vehicle, the overhead audio system includes multiplexor for receiving an input signal from a plurality of audio sources and for transmitting an output signal; a noise detection sensor for detecting a noise and providing noise data; an audio processor for processing the output signal received from the multiplexor and for receiving the noise data from the noise detection sensor, wherein the audio processor processes the output signal based on the noise data and transmits the processed output signal to a speaker; and wherein the overhead audio system is housed within a headliner of the vehicle.

DATA-DRIVEN SOFTWARE ARCHITECTURE FOR DIGITAL SOUND PROCESSING AND EQUALIZATION

A digital sound processing design system for a vehicle audio system includes a computer and a design tool that is run by the computer. The design tool allows a user to define sound processing criteria that is stored in a template file. An audio signal processor is connected to the first and second real channel inputs of an audio source. Memory that is coupled to the audio signal processor stores the template file. The sound processing engine that is coupled to the audio signal processor and the memory reads the template file at run-time to obtain the sound processing criteria. The sound processing engine applies the sound processing criteria to the first and second real channel inputs. The design tool allows a user to create virtual channel inputs and outputs that are based, in part, on the first and second real channel inputs.

METHOD AND SYSTEM FOR ENSURING AUDIO SAFETY

The invention concerns a method (200) and a system (100) for ensuring audio safety in an audio device (110). The system includes a processor (112), a sensor (116), an analog safety circuit (114) and a first feedback loop (117). The processor is programmed to output an acoustic output signal (310, 410), the sensor monitors the acoustic output signal and the analog safety circuit is coupled to an output of the processor. The first feedback loop feeds the monitored acoustic signal from the sensor to the analog safety circuit. The analog safety circuit adjusts from a first level (312, 412) to a second level (316, 414) the acoustic output signal when the acoustic output signal exceeds a predetermined safety threshold (314, 420) such that the audio device has an output capacity that is capable of driving a sound pressure level of the acoustic output signal above the predetermined safety threshold.

METHOD AND APPARATUS FOR AUTOMATICALLY ADJUSTING SPEAKER AND MICROPHONE GAINS WITHIN A MOBILE TELEPHONE

The cellular telephone (100) is provided with the capability for automatically adjusting the gain of a microphone (102) based upon the detected noise level in which the cellular telephone (100) is operated. As the noise level increases, the gain of the microphone (102) is automatically decreased thereby compensating for the natural tendency of telephone users to speak more loudly in noisy environments. Also, by decreasing the microphone gain, any clipping that might otherwise occur as a result to the user speaking more loudly is avoided and the signal to noise ratio is not thereby decreased. Furthermore, because the microphone gain decreases, the volume level of the voice of the user as it is output from the other party's telephone is not unduly loud. Hence, the other party need not manually decrease the speaker gain of his or her telephone. In the exemplary embodiment, the cellular telephone (100) includes a digital signal processor (110) configured or programmed to apply the detected ...

METHOD FOR VOLUME CONTROL OF AN AUDIO REPRODUCTION AND DEVICE FOR CARRYING OUT SAID METHOD

The invention relates to radio electronic engineering, in particular to audio reproducing devices. In order to expand the functional capabilities, the inventive method for volume control of an audio reproduction consists in the following: upper and lower levels are pre-set prior to volume control; the loudness is adjusted at the lower pre-set level thereof during the volume control in the presence of human voices, and at the upper pre-set level in the absence thereof. Analysis and a control unit of the volume control system of audio reproduction is embodied in such a way that it is possible to form a control signal at the output thereof. Said signal reduces the volume in the audio reproducing device to the lower pre-set level in the presence of human voices in accordance with a first pre-set time function, and increases the volume in the audio reproducing device to the upper pre-set level in the absence of human voices in accordance with a second pre-set time function.

AUDIO PLAYBACK DEVICE WITH AUDIO SCENE SIMULATION

The invention relates to an audio playback device (100) comprising a signal processing chain (110), a scene simulation device (120), a user interface (130), a signal modifier (140), and a storage device (150). The signal processing chain (110) prepares a useful acoustic signal for a playback for a listener. The scene simulation device (120) provides audio scene information for a simulated audio scene, wherein the simulated audio scene influences the listener's perception of the useful signal playback and/or defines a useful signal type. The user interface (130) detects playback parameter adjustments by a user, said adjustments representing an individual preference of the listener with respect to the simulated audio scene. The signal modifier (140) receives the playback parameter adjustments and modifies the playback of the useful signal in accordance with the playback parameter adjustments. The playback parameter adjustment and the audio scene information are stored in the storage device ...

CONTROLLING AUDIO OUTPUT OF A MOBILE TERMINAL

A mobile terminal has an audio device to adjust the gain of an audio amplifier depending on whether the terminal is in an open or a closed position. The audio amplifier provides an audio signal to a speaker, and the volume of the audio signal is proportional to the gain of the audio amplifier. One of a plurality of gain control circuits is selected according to the position of the terminal and is provided to the audio amplifier to adjust the gain. Therefore, the gain of the audio amplifier may be greater when the terminal is in the closed position than in the open position. Alternatively, one of the gain control circuits may be selected according to a measure of the ambient noise surrounding the terminal, or may be manually selected by a user.

SYSTEMS AND METHODS FOR OPTIMIZING AN AUDIO COMMUNICATION SYSTEM

Methods and systems for optimizing and audio communication system, including detecting a surrounding noise profile, determining one or more states of one or more noise-related conditions corresponding to the surrounding noise profile, and associating the one or more states of the one or more noise-related conditions to the surrounding noise profile.

Transducer holder

A transducer comprising: a flat strip of metal formed into a closed metal loop with two substantially parallel sides and rounded corners, a solenoid fixed to a surface on a first parallel side, and fixed magnets attached to an opposite parallel side of the trapezoidal shape to bring the fixed magnets into proximity with the solenoid, wherein at least one lateral side of the trapezoidal shape coupling said parallel sides has a continuously varying width, resulting in the transducer being able to resonate with high fidelity over a range of frequencies in response to reception of a signal comprising varying frequencies.

Audio monitoring system

An earphone monitoring system is disclosed that provides for user information and control regarding ambient and monitored sound levels.

Automatic process of adjustment of the volume of sound reproduction

A control process and apparatus which automatically ensures adjustment of the volume of sound delivered by a sound reproduction source, such as a car radio, as a function of the ambient sound signals so that the user has the sensation of listening to an audio program which is maintained at the same sound level regardless of variations in disturbing noise level. The apparatus uses a microphone which picks up the ambient sound signals and employs computer circuitry operating under program control for determining the average level of the ambient sound signals; circuits are used which supply signals to the computer representing the level of the sound delivered by the sound reproduction source; under program control the computer determines the difference in level between the ambient sound signal and the delivered sound signal to obtain a result (SOM) representing the value of the user's listening comfort. This value (SOM) is subtracted from subsequent measurements of the difference between the ...

Voice control circuit for a communication terminal

A voice control circuit controls the volume range of the voice signals in a communication terminal, such as a video telephone having a hands-free facility. In a hands-free facility, the signal-to-noise ratio is lower than in telephones with handset. To avoid audible feedback, hands-free facility conventionally includes a volume control circuit which decreases or increases the gain in the receive path as the gain in the set path increases or decreases. The problem to be solved is to utilize the maximum possible loop gain without reducing the gain in the receive path. The idea is to place behind the microphone (50) a controllable compandor (52) in which the signal voltages from the microphone (50) are A. compressed if they lie above a predetermined level, B. expanded if they lie below the predetermined level, and C. imparted maximum gain if they have the predetermined level. The predetermined signal level, which corresponds to the compression set point, is varied by a control circuit (90) ...

On-vehicle sound control apparatus

An electric source signal supplied from a battery of a vehicle is composed of a direct current component and an alternating current component. The alternating component is classified into an alternator noise of which a frequency changes in proportional to an engine speed of the vehicle and other noises. Also, as the engine speed is heightened, a vehicle noise in the inside of the vehicle is heightened. The direct current component of the electric source signal is removed in an alternating current component passing circuit, the other noises are removed in a band pass filter, and the alternator noise is obtained. Thereafter, a frequency of the alternator noise is calculated in a microcomputer, and a volume of a sound reproduced by a speaker of an on-vehicle stereo and a tone of the sound are adjusted according to the frequency of the alternator noise. Therefore, a driver can be entertained with a music without being disturbed by the vehicle noise.

Voice control circuit for a communication terminal

A voice control circuit controls the volume range of the voice signals in a communication terminal, such as a video telephone having a hands-free facility. In a hands-free facility, the signal-to-noise ratio is lower than in telephones with handset. To avoid audible feedback, hands-free facility conventionally includes a volume control circuit which decreases or increases the gain in the receive path as the gain in the set path increases or decreases. The problem to be solved is to utilize the maximum possible loop gain without reducing the gain in the receive path. The idea is to place behind the microphone (50) a controllable compandor (52) in which the signal voltages from the microphone (50) are A. compressed if they lie above a predetermined level, B. expanded if they lie below the predetermined level, and C. imparted maximum gain if they have the predetermined level. The predetermined signal level, which corresponds to the compression set point, is varied by a control circuit (90) ...

KNOWLEDGE BASED RADIO

A knowledge based radio that monitors changed operating parameters and environmental conditions, and that selectively varies operating parameters of the radio in response to rules. The rules can be varied over time by the radio based upon the monitored response of the user to various environmental conditions.

Automatic sound field correcting system

An automatic sound field correcting system with graphic equalizer means to be provided in an audio signal supply line. A test signal such as a pink noise is radiated in the listening space and detected at a predetermined listening point to determine the detection frequency characteristic of the test signal, thereby to adjust the frequency characteristic of the graphic equalizer means. Thereafter, the frequency characteristic of the graphic equalized means is made equal to different band-pass filter characteristics, one after another, and each time the equalization is effected, the detected test signal is obtained through the grapic equalizer means for determination of the detected frequency characteristic of the test signal.

Audibility enhancement

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

VIDEO OR AUDIO PROCESSING FORMING AN ESTIMATED QUANTILE

A method of audio processing receives a sequence of sample values, each corresponding with a location in video or audio content; forms an initially estimated quantile value; and then modifies the estimated value in dependence upon a count of the results of comparisons between sample values within a fixed-length interval of the sequence of samples values and the estimated value to form an estimated quantile of the sequence of sample values. The estimated quantile is used in the measurement or control of loudness.

SYSTEM AND METHOD FOR IMPROVED INFRARED COMMUNICATION BETWEEN CONSUMER APPLIANCES

Optical or electrical attenuation is provided at the infrared (IR) receiver of a first device, such as a remote control, together with a commensurate increase in output of the IR transmitter of an appliance, such as a set top box, to improve the signal-to-noise ratio of IR communications with respect to any background infrared radiation, such as radiation from a plasma display screen.

INFORMATION PROCESSING DEVICE, INFORMATION PROCESSING METHOD, VOICE OUTPUT DEVICE, AND VOICE OUTPUT METHOD

The present technology relates to an information processing device, an information processing method, a voice output device, and a voice output method that allow a user to hear a speech from a home agent unit regardless of where the user is. The voice output device outputs, on the basis of an image and noise from a predetermined noise source which are obtained in a unit capable of outputting a voice toward a user, information indicating hearing difficulty of a voice from the unit at a position of the user. The present technology can be applied to a household voice assistant device that performs a speech to a user.

Dynamic Range Improvement Technique

Apparatus and methods are disclosed for detecting and progressively attenuating specific frequencies prevalent in an audio signal. In contrast to conventional wide-band enhancement techniques over long time frames, narrow bandwidths and short attenuation times employed are commensurate with resonances and timing typical of speech. Apparent dynamic range is therefore increased through attenuation of longer-duration elements with declining informational contribution.

Bell sound outputting apparatus and method thereof

A method and apparatus for outputting a bell sound are disclosed. The method includes receiving an ambient noise; extracting a target frequency band from the ambient noise; changing volume of a pre-stored bell sound by using the target frequency band; and outputting the changed pre-stored bell sound. In the apparatus for outputting a bell sound, when a bell sound outputting apparatus receives an incoming call signal, a user may hear a bell sound including a signal in a frequency band not existing in a peripheral environment, and thus may easily recognize the bell sound.

Personalized adjustment of an audio device

Described herein are apparatuses, systems and methods that facilitate user adjustment of an audio effect of an audio device to match the hearing sensitivity of the user. The user can tune the audio device with a minimum perceptible level unique to the user. The audio device can adjust the audio effect in accordance with the minimum perceptible level. For example, a loudness level can adjust automatically to ensure that the user maintains a perceptible loudness, adjusting according to environmental noise and according to the minimum perceptible level. Also described herein are apparatuses, systems and methods related to an audio device equipped with embedded audio sensors that can maximize a voice quality while minimizing the effects of noise.

Volume control method and electronic device capable of automatic volume adjustment

A volume control method is to be performed by an electronic device for adjusting intensity of an output audio signal outputted from the electronic device according to an external audio signal of an environment in which the electronic device is located. The volume control method includes: configuring the electronic device to receive the external audio signal; configuring the electronic device to compare a sound track of the external audio signal with a preset sound track and to make a determination as to whether a degree of similarity between the two sound tracks reaches a preset ratio; and configuring the electronic device to stop outputting the output audio signal when a result of the determination is affirmative.

method and apparatus for providing information about the source of a sound via an audio device

A method and apparatus for providing information about the source of a sound via an audio device is described. An ambient sound is detected ( 200 ) and specific sounds are identified in the detected ambient sound ( 202 ). Information about the source of the identified specific sounds is determined ( 204 ). An operational control characteristic of a generated audio stream rendered by an audio device is changed ( 206 ) and information about the source is provided to the audio device upon detection of said identified specific sounds ( 208 ).

Fine/Coarse Gain Adjustment

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, including determining a coarse gain adjustment and a fine gain adjustment, the target gain adjustment corresponding to a combination of the coarse gain adjustment and the fine gain adjustment, applying the coarse gain adjustment to the signal carrying far-end speech using first gain adjustment circuitry to produce a first gain-adjusted signal, applying the fine gain adjustment to the signal carrying far-end speech using second gain adjustment circuitry to produce a second gain-adjusted signal, and providing a result of combining the first gain-adjusted signal and the second gain-adjusted signal to audio output from a wireless communications device.

System and methods for dynamically controlling atleast a media content with changes in ambient noise

Disclosed are methods and system method for dynamically maintaining at least an output volume of a media system at a predefined volume level in response to at least a change in an instantaneous ambient noise level. The method comprises the steps of: establishing the predefined volume level of the media system; measuring at least the change in the instantaneous ambient noise level; and adjusting dynamically the output volume of the media system against at least the change in the instantaneous ambient noise level to maintain the predefined volume level for a constant media experience.

Signal processing apparatus

A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Method and system for providing hearing assistance to a user

A method for providing hearing assistance to a user by capturing input audio signals; estimating a speech level of the input audio signals and an ambient noise level of the input audio signals; applying a gain model to the input audio signals to transform the input audio signals into filtered audio signals, wherein, for a each ambient noise level, the gain varies as a function of the speech level and wherein the function varies according to the ambient noise level changing the ratio of the gain at low speech levels and at high speech levels as a function of the ambient noise level; wirelessly transmitting the filtered audio signals to a receiver unit forming or being connected to a user hearing stimulator, the stimulator being worn at or in the user's ear; and stimulating the user's hearing by the stimulator according to audio signals supplied by the receiver unit.

Method and System

A method of warning and/or reduction of risk of ear damage of a hearer by excessive exposure to relatively high volume and/or frequency of sound, wherein a non-audio input representative of desired sound and/or ambient sound is employed to warn of said risk and/or said risk is automatically reduced.

Receiver with Variable Gain Control Transimpedance Amplifier

According to one embodiment, a compact low-power receiver comprises first and second analog circuits connected by a digitally controlled interface circuit. The first analog circuit has a first direct-current (DC) offset and a first common mode voltage at an output, and the second analog circuit has a second DC offset and a second common mode voltage at an input. The digitally controlled interface circuit connects the output to the input, and is configured to match the first and second DC offsets and to match the first and second common mode voltages. In one embodiment, the first analog circuit is a variable gain control transimpedance amplifier (TIA) implemented using a current mode buffer, the second analog circuit is a second-order adjustable low-pass filter, whereby a three-pole adjustable low-pass filter in the compact low-power receiver is effectively produced. 16-. (canceled)7. A variable gain control transimpedance amplifier (TIA) in a compact low-power receiver , said variable gain control TIA comprising:a current mode buffer configured for use with a low voltage power supply, said current mode buffer receiving differential current inputs and providing first and second differential outputs of said variable gain control TIA;at least one resistor and at least one capacitor coupled in parallel between said first and second differential outputs.8. The variable gain control TIA of claim 7 , wherein said variable gain control TIA is configured to provide a gain control range of at least approximately 30 dB.9. The variable gain control TIA of claim 7 , wherein said low voltage power supply comprises a less than approximately 1.3V power supply.1012-. (canceled)13. The variable gain control TIA of claim 7 , wherein said compact low-power receiver utilizes a three-pole low-pass Chebyshev filter.14. An adjustable low-pass filter (LPF) in a compact low-power receiver claim 7 , said adjustable LPF comprising:a resistance input block;a plurality of operational amplifiers ...

Audio signal adjustment method and audio player having audio signal adjustment function

A computerized method adjusts two presented audio signals using an audio player. A first audio signal is received and played through an audio playing unit. When a second audio signal that is inserted to be played is received in time during the playing of the first audio signal, a peak amplitude X of a volume of the first audio signal and a peak amplitude Y of a volume of the second audio signal are extracted. Each of the peak amplitude X and the peak amplitude Y are compared with a predetermined threshold value A. The first audio signal and/or the second audio signal are adjusted according to the comparison, and the first audio signal and/or the second audio signal is outputted to the audio playing unit and played through the audio playing unit according to the adjustment for the first audio signal and/or the second audio signal.

Electronic device, system and method for managing sound effect

A sound effect managing method is executed by an electronic device. In the method audio signals are generated and ambient noises are collected. Then audio data are obtained from the collected ambient noise. The audio data of the ambient noise are set as reference audio data and the reference audio data are adjusted by pre-set audio data to obtain target audio data. The audio signals are adjusted according to the target audio data and producing the adjusting audio signals.

METHOD, APPARATUS AND COMPUTER-READABLE MEDIA UTILIZING POSITIONAL INFORMATION TO DERIVE AGC OUTPUT PARAMETERS

Method and apparatus for automatic gain control utilizing sound source position information in a shared space having a plurality of microphones and a plurality of sound sources. Sound signals are received from the microphones. One or more processors locate position information corresponding to each of the sound sources. The processor(s) determine the distance to each of the sound sources from each of the microphones. The processor(s) define a predetermined gain weight adjustment for each of the microphones. The processor(s) apply the defined weight adjustments to the microphones to achieve a consistent volume of the desired plurality of sound sources. The processor(s) maintain a consistent ambient sound level regardless of the position of the sound sources and the applied gain weight adjustments. The processor(s) output a summed signal of the sound sources at a consistent volume with a constant ambient sound level across the plurality of sound source positions. 1. A method of controlling one or more microphones in a plurality of microphones , based on measured distance information from a sound source to each of the plurality of microphones , in a shared physical space having the plurality of microphones and the sound source , comprising:receiving from the plurality of microphones, sound signals corresponding to the measured distance information;determining, using one or more processors, position information corresponding to the sound source in the shared space;determining, using the one or more processors, the measured distance information to the sound source from each of the plurality of the microphones in the shared space, based on the position information;controlling, using the one or more processors, one or more microphones of the plurality of microphones, based on the measured distance information; (i) when the sound source is outside of a configurable threshold distance, all microphones of the plurality of microphones are enabled;', '(ii) when the sound source ...

HOWLING SUPPRESSION APPARATUS, AND METHOD AND PROGRAM FOR THE SAME

A howling suppression apparatus includes: an integration processing part that obtains the maximum value among L values corresponding to n-th frames of L i-th signals, for i=1, 2, . . . , L, L being any integer equal to or greater than 2, the L i-th signals being frequency-domain signals obtained from sound signals collected by multiple microphones; and a howling suppression processing part that performs howling suppression processing on at least any of the L i-th signals using the maximum value. 1. A howling suppression apparatus comprising:an integration processing part that obtains a maximum value among L values corresponding to n-th frames of L i-th signals, for i=1, 2, . . . , L, L being any integer equal to or greater than 2, the L i-th signals being frequency-domain signals obtained from sound signals collected by a plurality of microphones; anda howling suppression processing part that performs howling suppression processing on at least any of the L i-th signals using the maximum value.2. The howling suppression apparatus according to claim 1 , wherein the howling suppression processing part performs the howling suppression processing by utilizing a fact that howling components are out of phase among the sound signals collected by the plurality of microphones.3. The howling suppression apparatus according to or claim 1 , wherein claim 1 , at least either (i) if the maximum value is greater than a value indicating predetermined power or (ii) if a value indicating a variation in the maximum value is greater than a value indicating a predetermined variation claim 1 , the howling suppression processing part performs the howling suppression processing by multiplying at least any of the L i-th signals by a smaller one of a first gain obtained based on the maximum value and a second gain obtained based on the value indicating the variation in the maximum value.4. The howling suppression apparatus according to or claim 1 , comprising a smoothing processing part that ...

Minute Signal Detection Method and System

In an environment in which signal-to-noise is poor, a method and a system configuration for power-saving, low-cost, and general minute signal detection are provided. The system includes a circuit that converts and amplifies an input signal, a nonlinear analog front-end circuit that determines the existence of a minute signal from the input signal and that outputs information on the existence of the same as an event signal, an analog-to-digital-conversion circuit that drives operation-mode control based on the event signal and performs analog-to-digital conversion on the converted-and-amplified input signal, a data-transfer circuit that drives the operation-mode control by the event signal and transfers the analog-to-digital converted signal, a digital-signal-processing circuit that drives the operation-mode control by the event signal and performs digital-signal processing on the signal transmitted from the data-transfer circuit and detects the signal, and a parameter-control circuit that controls a characteristic parameter of the nonlinear analog front-end circuit.

Methods And Apparatus For Adaptive Gain Control In A Communication System

Methods and apparatus for a communication system having microphones and loudspeakers to determine a noise and speech level estimate for a transformed signal, determine a SNR from the noise and speech level estimates, and determine a gain for the transformed signal to achieve a selected SNR range at a given position. In one embodiment, the gain is determined by adapting an actual gain to follow a target gain, wherein the target gain is adjusted to achieve the selected SNR range. 1. A method , comprising:for a communication system having microphones and loudspeakers, transforming a signal received by a first one of the microphones to the frequency domain;determining a noise level estimate for the transformed signal;determining a speech level estimate for the transformed signal;determining a SNR from the noise and speech level estimates; and adapting an actual gain to follow a target gain, wherein the target gain is adjusted to achieve the selected SNR range;', 'comparing the target gain and the actual gain to determine a gain change increment;', 'increasing the actual gain if the SNR at the given position is lower than a minimum SNR in the SNR range; and', 'decreasing the actual gain if the SNR at the given position is higher than a maximum SNR in the SNR range., 'determining a gain for the transformed signal to achieve a selected SNR range at a given position, wherein determining the gain comprises2. The method according to claim 1 , further including adapting the SNR range in response to a change in the noise level.3. The method according to claim 1 , further including using a psychoacoustic measure for determining the noise and/or speech level estimate.4. The method according to claim 3 , further including determining a noise level estimate for the transformed signal using spectral weights based upon the psychoacoustic measure.5. The method according to claim 1 , further including performing voice activity detection on the transformed signal and generating ...

BALANCING MOBILE DEVICE AUDIO

Techniques related to balancing audio for mobile devices are discussed. Such techniques may include detecting a speaker of the device is at least partially obstructed, determining an audio output from the device is impeded due to the speaker being at least partially obstructed, and increasing an output from the speaker and/or an alternative speaker based on the audio output from the device being impeded. 1. A method for performing audio balancing for a device comprising:detecting a speaker of the device is at least partially obstructed;determining an audio output from the device is impeded due to the speaker being at least partially obstructed; andincreasing an output from at least one of the speaker or a second speaker based on the audio output from the device being impeded.2. The method of claim 1 , wherein detecting the speaker of the device is at least partially obstructed comprises detecting claim 1 , adjacent to the speaker claim 1 , at least one of physical pressure claim 1 , a change in infrared light claim 1 , a change in temperature claim 1 , a change in resistance claim 1 , a change in conductivity claim 1 , or a change in shape of a casing of the device.3. The method of claim 1 , wherein detecting the speaker of the device is at least partially obstructed comprises detecting claim 1 , via a sensor integrated into a casing of the device claim 1 , at least one of physical pressure claim 1 , a change in temperature claim 1 , a change in resistance claim 1 , a change in conductivity claim 1 , or a change in shape of the casing of the device.4. The method of claim 1 , wherein detecting the speaker of the device is at least partially obstructed comprises detecting claim 1 , via a miniature microphone adjacent to the speaker claim 1 , an audio sample and determining the audio sample is distorted.5. The method of claim 1 , wherein detecting the speaker of the device is at least partially obstructed comprises detecting claim 1 , via an infrared sensor claim 1 , a ...

PERSONALIZED ADJUSTMENT OF AN AUDIO DEVICE

Described herein are apparatuses, systems and methods that facilitate user adjustment of an audio effect of an audio device to match the hearing sensitivity of the user. The user can tune the audio device with a minimum perceptible level unique to the user. The audio device can adjust the audio effect in accordance with the minimum perceptible level. For example, a loudness level can adjust automatically to ensure that the user maintains a perceptible loudness, adjusting according to environmental noise and according to the minimum perceptible level. Also described herein are apparatuses, systems and methods related to an audio device equipped with embedded audio sensors that can maximize a voice quality while minimizing the effects of noise.

PORTABLE COMPUTING DEVICES AND COMMAND INPUT METHODS FOR THE PORTABLE COMPUTING DEVICES

Methods for inputting commands to portable computing devices are disclosed. One method includes detecting a sound, analyzing attenuation of the detected sound in a predetermined frequency, and inputting a command to a portable computing device based on analysis of the detected sound. Apparatus and computer program products implementing the methods are also disclosed. 1. An apparatus , comprising:a microphone configured to detect a sound;an analysis unit configured to analyze attenuation of the detected sound in a predetermined frequency; anda command input unit configured to input a volume command based on analysis of the detected sound by the analysis unit,wherein at least a portion of said units comprise one or more of a set of hardware circuits, a set of programmable hardware devices, and executable code stored on a set of non-transitory computer-readable storage mediums.2. The apparatus of claim 1 , wherein:the analysis unit determines a direction that a user swipes the apparatus with a finger relative to the microphone; andthe command input unit inputs the volume command corresponding to the direction that swiping is performed with the finger relative to the microphone.3. The apparatus of claim 2 , wherein:the volume command is based on one of a first direction in which the user swipes the portable computing device with the finger in an approaching direction relative to the microphone to decrease volume and a second direction in which the user swipes the portable computing device with the finger in a moving away direction relative to the microphone to decrease volume.4. The apparatus of claim 3 , further comprising:a detection unit configured to detect a tapping operation performed on the apparatus,wherein the command input unit inputs the volume command based on detecting the tapping operation by the detection unit.5. The apparatus of claim 4 , wherein:the command input unit inputs the volume command that corresponds to a combination of the analysis by the ...

Methods and Apparatus for Adjusting a Level of an Audio Signal

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal. 1. A method for adjusting a level of an audio signal in an audio processing apparatus , the method comprising:dividing an audio signal into a plurality of frequency bands;delaying at least one of the plurality of the frequency bands;obtaining modification parameters for at least one of the plurality of frequency bands, the modification parameters comprising amplitude scale factors, each of the amplitude scale factors respectively operating in a frequency band of the plurality of frequency bands and each amplitude scale factor representing an average energy over the frequency band and a time segment;deriving gain factors for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors;smoothing the gain factors, wherein the smoothing of the gain factors is optional;determining a level of noise from noise compensation factors;applying the gain factors to at least one of the frequency bands to generate gain adjusted frequency bands;adjusting the level of noise based on the gain adjusted frequency bands; andsynthesizing the plurality of frequency bands to generate an output ...

AUTOMATED PAUSE OF MEDIA CONTENT PLAYBACK BASED ON SOUND LEVEL

An example system for playing media content with a media playback device in a vehicle can be programmed to obtain a sound measurement indicative of a sound level associated with playback of the media content by the media playback device in the vehicle. The example system also can determine a deviation in an expected sound level based upon the sound measurement. Finally, the system can modify playback of the media content by the media playback device () based upon the deviation. 1. A method for playing media content , the method comprising:transmitting, by a first device, media content to a second device for playback on the second device;obtaining, by the first device, a sound measurement indicative of a sound level associated with the playback of the media content on the second device;determining, by the first device, a deviation in the sound level from an expected sound level based upon the sound measurement; andresponsive to the determination, causing, by the first device, the playback of the media content on the second device to stop.2. The method of claim 1 , wherein obtaining the sound measurement further comprises removing claim 1 , by the first device claim 1 , sounds from sources other than the second device playing back the media content.3. The method of claim 1 , wherein the sound level is a volume of the media content item being played back on the second device.4. The method of claim 1 , wherein determining the deviation in the sound level comprises determining the deviation in the sound level is lower than the expected sound level.5. The method of claim 4 , wherein determining the deviation in the sound level comprises determining the sound level is lower than the expected sound level for a predetermined duration of time.6. The method of claim 1 , wherein the expected sound level is a dynamic threshold based on a learned behavior of a user associated with the first device.7. The method of claim 1 , wherein the expected sound level is a dynamic threshold ...

Loudness Matching

An example method may involve a device determining a first loudness representation for a playback device based on a first equalization setting applied to a representation of average music. The device may also determine a second loudness representation for the playback device, based on a second equalization setting applied to the representation of average music. The device may also determine a loudness adjustment factor based on the first and second loudness representations, and then causing the playback device to play back media based on the second equalization setting and the determined loudness adjustment factor. 1. A device comprising:a processor;a non-transitory computer-readable medium; and determining a first loudness representation for a playback device, wherein the playback device comprises a first equalization setting, and wherein the first loudness representation is based on the first equalization setting applied to a representation of average music;', 'determining a second loudness representation for the playback device, wherein the playback device comprises a second equalization setting, and wherein the second loudness representation is based on the second equalization setting applied to the representation of average music;', 'determining a loudness adjustment factor based on the first and second loudness representations; and', 'causing the playback device to play back media based on the second equalization setting and the determined loudness adjustment factor., 'program instructions stored on the non-transitory computer-readable medium that, when executed by the processor, cause the device to perform operations comprising2. The device of claim 1 , wherein the first and second equalization settings are associated with a first and second media source claim 1 , respectively.3. The device of claim 1 , wherein the representation of average music comprises a power-frequency dropoff curve.4. The device of claim 1 , wherein the program instructions claim 1 , ...

POST-PROCESSING GAINS FOR SIGNAL ENHANCEMENT

A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Content reproducing electronic apparatus and controlling method thereof

A content reproducing electronic apparatus includes a communicator including circuitry, an audio output device, and a processor configured to receive state information indicating a state of an external electronic apparatus from the external electronic apparatus located around the electronic apparatus via the communicator, obtain noise information corresponding to the received state information, and control the audio output device to output an audio signal of a content based on a volume corresponding to the obtained noise information.

SYSTEM AND METHOD FOR CONTROLLING VOLUME

Systems and methods provide a solution to a common nuisance of loud background noise encountered when watching television, allowing the issue to be solved with little or no user interaction. Systems and methods automatically modulate device volume based on background noise level. The systems and methods record a background level of noise, and if that background level of noise increases, the systems and methods automatically increase the content volume which had been set by the user. Exceptions are made for brief sounds or for viewer conversations. 1. A method of improving delivery of audio content during a content playback experience , comprising:a. detecting an initial BBNL;b. determining a desired CAL;c. detecting a sustained increase in the BBNL; andd. automatically causing an increase in the CAL to compensate for the detected sustained increase in the BBNL.2. The method of claim 1 , further comprising automatically causing display of closed-captioning to further compensate for the detected sustained increase in the BBNL.3. The method of claim 1 , further comprising causing the CAL to decrease when the increased BBNL is no longer detected.4. The method of claim 1 , wherein the increase in the CAL is at least proportional to the increase in the BBNL.5. The method of claim 1 , further comprising determining a metric based on the initial BBNL and the desired CAL claim 1 , wherein the automatic increase in the CAL is such that the value of the metric is unchanged.6. The method of claim 5 , wherein the metric is a signal-to-noise ratio.7. The method of claim 1 , further comprising detecting proximate speech claim 1 , wherein if proximate speech is detected claim 1 , and if the proximate speech causes an increase in the BBNL claim 1 , then suppressing the automatic increase in the CAL caused by the increase in BBNL due to the proximate speech.8. The method of claim 1 , further comprising detecting if the increase in BBNL exceeds a threshold claim 1 , and if so claim 1 ...

Method and Apparatus for Adjusting Volume of User Terminal, and Terminal

A volume adjustment method and apparatus, and a terminal is presented. Perform analysis on the collected sound signal surrounding a user terminal, to obtain composition information, where the composition information includes sound types included in the sound signal and proportions of sounds of the various types, and the sound types include blank sound, human sound, and noise; determine a current scene mode of the user terminal according to the composition information; and adjust volume of the user terminal according to the determined scene mode, thereby significantly reducing occurrence of a case, caused by mistaken determining of the scenario, in which play volume adjustment does not conform to the scenario, and enhancing user experience. 1. A method for adjusting volume of a user terminal , comprising:collecting a sound signal surrounding the user terminal;performing analysis on the collected sound signal in order to obtain composition information of the sound signal, wherein the composition information comprises sound types comprised in the sound signal and proportions of sounds of various types, and wherein the sound types comprise blank sound, human sound, and noise;determining a current scene mode of the user terminal according to the composition information of the sound signal; andadjusting a volume of the user terminal according to the determined scene mode.2. The method according to claim 1 , wherein collecting the sound signal surrounding the user terminal comprises collecting a sound signal of a current environment in which the user terminal is located in response to detection of a call signal arrival.3. The method according to claim 1 , wherein collecting the sound signal surrounding the user terminal comprises periodically collecting a sound signal of a current environment in which the user terminal is located.4. The method according to claim 1 , wherein adjusting the volume of the user terminal according to the determined scene mode comprises: ...

PERCEPTUAL POWER REDUCTION SYSTEM AND METHOD

A method of reducing power consumed by an audio playback system includes: receiving a signal from a digital audio interface; comparing a sound pressure level (SPL) response of a first frequency band of the signal to a total SPL response of the signal, the first frequency band including frequencies lower than a selected frequency; and attenuating signal components in the first frequency band of the signal based upon the comparison of the SPL response of the first frequency band of the signal to the total SPL response of the signal. 1. A method of reducing power consumption by an audio playback system , comprising:receiving a signal from a digital audio interface;comparing a sound pressure level (SPL) response of a first frequency band of the signal to a total SPL response of the signal, the first frequency band including frequencies lower than a selected frequency; andattenuating signal components in the first frequency band of the signal based upon the comparison of the SPL response of the first frequency band of the signal to the total SPL response of the signal.2. The method of claim 1 , wherein comparing the SPL response of the first frequency band of the signal to the total SPL response of the signal includes:determining a ratio of the SPL response of the first frequency band of the signal to the total SPL response of the signal; andcomparing the determined ratio to a threshold ratio, wherein the signal components in the first frequency band of the signal are attenuated when the determined ratio is less than the threshold ratio.3. The method of claim 2 , wherein determining the ratio of the SPL response of the first frequency band of the signal to the total SPL response of the signal includes:determining a root mean square value of the SPL response of the first frequency band of the signal;determining a root mean square value of the total SPL response of the signal; anddetermining a ratio of the root mean square value of the SPL response of the first frequency ...

MEASUREMENT SYSTEM AND MEASUREMENT METHOD

A measurement method of this disclosure is for evaluating an electronic device () that transmits air-conducted sound, generated by vibration of a vibrating body (), to a user by the vibrating body () being placed in contact with a human ear. The measurement method includes placing the vibrating body () in contact with an ear model unit (), measuring air-conducted sound generated by the electronic device () and air-conducted sound generated in the ear model unit (), and selecting a correction value from among a plurality of correction values. 1. A measurement system for evaluating an electronic device that transmits air-conducted sound , generated by vibration of a vibrating body , to a user by the vibrating body being placed in contact with a human ear , the measurement system comprising:an ear model unit contacted by the vibrating body;a sound pressure gauge configured to measure air-conducted sound generated by the electronic device and air-conducted sound generated in the ear model unit; anda data corrector configured to correct a sound pressure measured by the sound pressure gauge, whereinthe data corrector corrects the sound pressure using a correction value selected from among a plurality of correction values.2. The measurement system of claim 1 , wherein the user selects the correction value from among the plurality of correction values.3. The measurement system of claim 1 , further comprising a vibration detector configured to measure vibration transmitted to the model ear unit.4. The measurement system of claim 3 , wherein the measurement system selects the correction value from among the plurality of correction values using a value detected by the vibration detector and one of a sound pressure of air-conducted sound before correction and a sound pressure of air-conducted sound when temporarily selecting a temporary correction value.5. The measurement system of claim 4 , whereina first correction value is selected when measuring a first electronic device ...

Ambient sound activated device

In device having at least one microphone and one or more speakers, environmental sound may be recorded using the microphone, classified and mixed with source media sound to produce a mixed sound depending on the classification. The mixed sound may then be played over the one or more speakers.

DYNAMIC ALARM SYSTEM FOR REDUCING ALARM FATIGUE

A dynamic alarm system that can assess the background noise in a clinical environment with sub-minute time sampling and change the alarm output to a desired signal-to-noise ratio (SNR) below ambient noise (negative SNR) to minimize clinician fatigue and ameliorate deleterious patient outcomes, while preserving clinician performance. 1. A dynamic alarm system comprising:a housing;an electronic processor;a microphone electronically coupled to the electronic processor, the microphone configured to sense an ambient noise level proximal to the housing;a user interface on the housing and electronically coupled to the electronic processor, the user interface configured to allow a user to select a signal-to-noise ratio based on the ambient noise level; anda speaker electronically coupled to the electronic processor, the speaker configured to output an audible tone according to the selected signal-to-noise ratio when a physiological parameter detected by a sensor coupled to a patient exceeds a preselected threshold for that physiological parameter.2. The system of claim 1 , wherein the housing includes a port configured to receive data from the sensor.3. The system of claim 1 , wherein the physiological parameter is blood pressure claim 1 , heart rate claim 1 , ECG signal claim 1 , or blood oxygen saturation.4. The system of claim 1 , wherein the electronic processor is configured to control power to the speaker to control an alarm volume based on the selected signal-to-noise-ratio.5. The system of claim 4 , wherein the alarm volume is less than the ambient noise level.6. The system of claim 4 , wherein the alarm volume is greater than the ambient noise level.7. The system of claim 1 , wherein the signal-to-noise-ratio is between −11 dB and +4 dB.8. The system of claim 1 , wherein the speaker output is between 33 dB and 81 dB.9. The system of claim 8 , wherein the microphone input is between 50 dB and 90 dB.10. A dynamic alarm system comprising:a microphone configured to ...

System and Method for Non-Destructively Normalizing Loudness of Audio Signals within Portable Devices

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. 1. A method for decoding an encoded input signal to generate an audio output signal , wherein the method comprises:receiving the encoded input signal, wherein the encoded input signal includes encoded audio information and associated metadata including one or more decoding-control parameters, one or more first parameters specifying dynamic range compression suitable for use by a first mode of decoding that uses a first reproduction level, and one or more second parameters specifying dynamic range compression suitable for use by a second mode of decoding that uses a second reference reproduction level;applying a decoding process to the encoded audio information to obtain subband signals, wherein the decoding process is adapted in response to the one or more decoding-control parameters;modifying the subband signals to obtain modified subband signals with changed dynamic range characteristics, wherein the modifying is adapted in response to the one or more second parameters;applying a synthesis filter bank to the modified subband signals to obtain a time-domain audio signal; andapplying a gain and a limiter to the time-domain audio signal, wherein the application of the gain raises an effective reference reproduction level of the audio output signal, and wherein the application of the limiter prevents amplitudes of the audio output ...

Speaker with Automatic Volume Control

An example speaker for providing alerts includes: a housing and a cover forming an integrated unit; a microphone configured to sense an ambient noise level; and a processor configured to adjust a volume of the speaker based upon the ambient noise level. A hinge can be positioned between the housing and the cover to allow the cover to be pivoted relative to the housing from a close position to an open position. 120-. (canceled)21. An apparatus , comprising:a housing;a cover coupled to the housing;a speaker coupled to the housing and operable to provide alerts;an amplifier configured to amplify alert signals provided to the speaker; anda control board including a processor with a programmable logic controller (PLC) interface configured to allow third parties to control and monitor the speaker, the processor being configured to initiate playback, using the speaker, of the alerts, the alerts including one or more of digitally stored tones, voice messages, and predefined relay output sequences.22. The apparatus of claim 21 , further comprising a wire electrically connecting a first component positioned in claim 21 , and fixed relative to claim 21 , the housing to a second component positioned in claim 21 , and fixed relative to claim 21 , the cover.23. The apparatus of claim 22 , wherein the second component is the control board.24. The apparatus of claim 21 , wherein the processor is configured to communicate one or more conditions at the apparatus to one or more third parties.25. The apparatus of claim 24 , wherein the one or more conditions include a voltage claim 24 , a current and a temperature.26. The apparatus of claim 21 , wherein a wiring harness electrically connects the control board to a power supply positioned in claim 21 , and fixed relative to claim 21 , the housing.27. The apparatus of claim 21 , further comprising a power supply positioned in claim 21 , and fixed relative to claim 21 , the housing.28. The apparatus of claim 21 , wherein the amplifier is ...

VOLUME LEVELER CONTROLLER AND CONTROLLING METHOD

Volume leveler controller and controlling method are disclosed. In one embodiment, A volume leveler controller includes an audio content classifier for identifying the content type of an audio signal in real time; and an adjusting unit for adjusting a volume leveler in a continuous manner based on the content type as identified. The adjusting unit may configured to positively correlate the dynamic gain of the volume leveler with informative content types of the audio signal, and negatively correlate the dynamic gain of the volume leveler with interfering content types of the audio signal. 1. A volume leveler controller comprising:an audio content classifier for identifying the content type of an audio signal in real time; andan adjusting unit for adjusting a volume leveler in a continuous manner based on the content type as identified;wherein the adjusting unit is configured to positively correlate the dynamic gain of the volume leveler with informative content types of the audio signal, and negatively correlate the dynamic gain of the volume leveler with interfering content types of the audio signal.2. The volume leveler controller according to claim 1 , wherein the content type of the audio signal comprises one of: speech claim 1 , short-term music claim 1 , noise and background sound.3. The volume leveler controller according to claim 1 , wherein noise is regarded as an interfering content type.4. The volume leveler controller according to claim 1 , wherein the adjusting unit is configured to adjust the dynamic gain of the volume leveler based on the confidence value of the content type.5. The volume leveler controller according to claim 4 , wherein the adjusting unit is configured to adjust the dynamic gain via a transfer function of the confidence value of the content type.6. The volume leveler controller according to claim 1 , wherein the audio content classifier is configured to classify the audio signal into multiple content types with corresponding ...

METHOD FOR SYNCHRONIZING LEFT AND RIGHT CHANNELS IN SMART AUDIO AMPLIFIERS

A method for synchronizing the left and right channels in smart audio amplifiers for stereo application, where each channel gain is determined by speaker limitation nonlinearly. Gain information is exchanged in real time and a threshold is set such that, if the difference in the gain between the two channels is above the threshold, the gain correction action is taken, and if the difference is below the threshold, it continues to compare the gain of the two channels in the next time frame. 1. A method for synchronizing a left channel and a right channel in a smart audio amplifier , comprisingbuilding a communication channel between a left channel and a right channel,exchanging gain information of the left channel and the right channel in real time,calculating difference in the gain between the left channel and the right channel and comparing the difference with a set threshold, andadjusting the gain when the difference is above the threshold, or continuing to compare in a next time frame when the difference is below the threshold.2. The method of claim 1 , wherein the communication channel is built in a communication protocol that is in I2S format.3. The method of claim 1 , wherein value of the set threshold is 2 Db.4. The method of claim 1 , wherein the channel having the higher gain is adjusted to have the same gain as the channel having the lower gain when the difference is above the threshold.5. The method of claim 1 , wherein the channel having the lower gain is adjusted to have the same gain as the channel having the higher gain when the difference is above the threshold. The subject application claims priority on Chinese patent application No. 201610666367.6 filed on Aug. 13, 2016. The Chinese priority application is incorporated herein by reference.The present invention relates to audio amplifier synchronization, particularly, method for two-channel synchronization in an audio amplifier.An audio amplifier is a device to amplify audio signals and to drive the ...

Audio equalization metadata

Introduced here are systems and methods to enable recording artists and engineers to specify exactly how the audio track should be played as well as perceived by the user in the case where the frequency transfer functions of the sound playback system and/or listening mechanisms (user's own hearing) can be measured and compensated for. For example, the acoustic environment during recording and mastering can be measured, and the measurements can be recorded in an inaudible portion of an audio track. The acoustic environment can include speaker frequency, distortion, reverberation, channel separation, room acoustics, etc. In addition, a hearing profile of the audio creator, such as the recording artist, sound engineer, mastering person, etc., can be included within the inaudible data. Further, the acoustic environment and/or the hearing profile of the audio consumer can also be used to modify the audio prior to reproducing the audio to the audio consumer.

METHOD FOR AMPLIFYING AUDIO SIGNAL BASED ON ITS AMPLITUDE AND ELECTRONIC DEVICE SUPPORTING THE SAME

An electronic device includes a processor. The processor is configured to examine the sound level of an audio signal obtained from an external object for a preset time; change, if the frequency with which the sound level becomes higher than or equal to a specified level satisfies a preset condition, a reference level, at which the gain value for audio signal amplification is changed in accordance with the amplitude change of the audio signal, for at least one section of the audio signal satisfying the preset condition; and amplify the at least one section of the audio signal according to the changed reference level. 1. An electronic device comprising a processor wherein the processor is configured to:examine a sound level of an audio signal for a preset time, the audio signal obtained from an external source;change, if a frequency with which the sound level becomes higher than or equal to a specified level satisfies a preset condition, a reference level at which a gain value for audio signal amplification is changed in accordance with an amplitude change of the audio signal, for at least one section of the audio signal satisfying the preset condition; andamplify the at least one section of the audio signal according to the changed reference level.2. The electronic device of claim 1 , wherein the processor is configured to:filter the audio signal into a first frequency band signal and a second frequency band signal by using a filter, and for each of the first frequency band signal and second frequency band signals:examine a sound level of the frequency band signal for a preset time;change, if the frequency with which the sound level becomes higher than or equal to a specified level satisfies a preset condition, a reference level at which a gain value for audio signal amplification is changed in accordance with an amplitude change of the frequency band signal, for at least one section of the frequency band signal satisfying the preset condition; andamplify the at ...

Electronic apparatus and control method thereof

An electronic apparatus and a method for controlling the electronic apparatus are provided. The electronic apparatus includes an audio processor that outputs an audio including a first audio component and a second audio component and a controller that controls the audio processor to change a ratio of the first audio component to the second audio component according to the changed audio level, in response to receiving an instruction to change a level of the output audio.

Method and apparatus of playing music based on surrounding situations

Disclosed is a method and apparatus for playing music that acquires surrounding situation information including a driving situation and a movement of an object relative to a driving vehicle, generates sound information to represent the surrounding situation information as a change in a sound transferred through an audio device including speakers in the vehicle, and controls the audio device based on the sound information to play the music.

Sound processing device and non-transitory computer-readable storage medium

A sound processing device includes a processor configured to generate a first frequency spectrum of a first sound signal corresponding to a first sound received at a first input device and a second frequency spectrum of a second sound signal corresponding to the first sound received at a second input device, calculate a transfer characteristic based on a first difference between an intensity of the first frequency spectrum and an intensity of the second frequency spectrum, generate a third frequency spectrum of a third sound signal transmitted from the first input device and a fourth frequency spectrum of a fourth sound signal transmitted from the second input device, specify a suppression level of an intensity of the fourth frequency spectrum based on a second difference between an intensity of the third frequency spectrum and an intensity of the fourth frequency spectrum.

Information processing apparatus, control method, and non-transitory computer readable storage medium

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.

MAINTAINING SPATIAL STABILITY UTILIZING COMMON GAIN COEFFICIENT

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. 1. A computer implemented method for maintaining the spatial stability of a sound field comprising:estimating a background noise estimate for each of a first audio signal and a second audio signal;calculating a first gain coefficient responsive to the first audio signal and the background noise estimate of the first audio signal;calculating a second gain coefficient responsive to the second audio signal and the background noise estimate of the second audio signal;selecting a common gain coefficient from the first gain coefficient and the second gain coefficient; andapplying the selected common gain coefficient to each of the first audio signal and the second audio signal.2. The method for maintaining the spatial stability of a sound field of claim 1 , where selecting the common gain coefficient comprises selecting a gain coefficient claim 1 , from the first gain coefficient and the second gain coefficient claim 1 , that will result in a least amount of audio signal modification.3. The method for maintaining the spatial stability of a sound field of claim 1 , where selecting the common gain coefficient comprises selecting a gain coefficient claim 1 , from the ...

USER-CONTROLLED BEAM STEERING IN MICROPHONE ARRAY

Various aspects include approaches for controlling acoustic beams. In some particular aspects, a system includes: a microphone array; and a control system coupled with the microphone array, the control system programmed to control the microphone array by: processing a signal from a beacon device to identify a physical location of the beacon device relative to the microphone array; and focusing the microphone array in a direction based upon the physical location of the beacon device prior to receiving an audio input at the microphone array. 1. A system comprising:a microphone array; and processing a signal from a beacon device to identify a physical location of the beacon device relative to the microphone array; and', 'focusing the microphone array in a direction based upon the physical location of the beacon device prior to receiving an audio input at the microphone array., 'a control system coupled with the microphone array, the control system programmed to control the microphone array by2. The system of claim 1 , wherein the beacon device includes a focus mode for pre-focusing the microphone array toward the beacon device or a negation mode for pre-focusing the microphone array away from the beacon device.3. The system of claim 2 , wherein the beacon device is pre-set to the focus mode or the negation mode prior to sending the signal indicating the physical location of the beacon device.4. The system of claim 1 , further comprising a camera system connected with the control system claim 1 , the camera system for detecting a visual characteristic of a user claim 1 , the control system further programmed to:receive a camera signal from the camera system indicating the visual characteristic of the user; andfocus the microphone array in response to the visual characteristic of the user matching a visual characteristic in a stored visual characteristic library.5. The system of claim 1 , wherein the control system includes a digital signal processor for modifying the ...

Methods and Apparatus for Adjusting a Level of an Audio Signal

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal. 1. A method for adjusting a level of an audio signal in an audio processing apparatus , the method comprising:dividing an audio signal into a plurality of frequency bands;delaying at least one of the plurality of the frequency bands;obtaining modification parameters for at least one of the plurality of frequency bands, the modification parameters comprising filter coefficients and amplitude scale factors, each of the amplitude scale factors respectively operating in a frequency band of the plurality of frequency bands and each amplitude scale factor representing an average energy over the frequency band and a time segment;deriving gain factors for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors;smoothing the gain factors, wherein the smoothing of the gain factors is optional;determining a level of noise from noise compensation factors;applying the gain factors to at least one of the frequency bands to generate gain adjusted frequency bands;adjusting the level of noise based on the gain adjusted frequency bands;filtering at least one of the frequency bands with a ...

AUTOMATIC VOLUME CONTROL FOR LAND MOBILE RADIO

The present disclosure provides a radio and method for automatically adjusting, in response to ambient noise, the volume setting of a radio. The radio comprises a microphone for receiving ambient audio and generating a microphone signal; a codec for receiving the microphone signal and generating a processor signal representative of the ambient audio; and processor circuitry operable to receive the processor signal, determine an ambient audio level in response to the processor signal, and determine an adjusted radio volume to generate a radio volume control signal, wherein the adjusted radio volume is determined by calculating a difference between a baseline volume level and ambient audio level and adding a current volume level setting, wherein the codec is also operable to receive the radio volume control signal and generate an output signal to adjust the radio volume setting to maintain a net difference between the adjusted radio volume and ambient audio. 1. A land mobile radio for use in a land mobile radio (LMR) system , the land mobile radio configured to adjust a current volume level setting of the land mobile radio in response to ambient audio and comprising:a first circuitry configured to receive a processor input signal based on the ambient audio, and in response to the processor input signal based on the ambient audio, calculate a first amplitude difference between the ambient audio and the current volume level setting,wherein the first circuitry is configured to provide an upper volume limit and a lower volume limit,wherein the first circuitry is configured to adjust the current volume level setting to maintain the current volume level setting within the upper volume limit and the lower volume limit,wherein the first circuitry is configured to set the current volume level setting of the land mobile radio such that (i) a difference equal to the first amplitude difference is maintained between the current volume level setting and the ambient audio as the ...

Calibration of Playback Device to Target Curve

Techniques described herein may involve dynamically adjusting the equalization of multiple speakers in a playback device. An example implementation involves a playback device emitting, via a speaker, a first audio signal. While emitting the first audio signal, the playback device detects, via a microphone, data representing a second audio signal, the second audio signal comprising one or more reflections of the first audio signal as emitted by the speaker within a surrounding environment. The playback device determines a frequency response curve based on the second audio signal, the frequency response curve representing acoustic characteristics of a system comprising the playback device and the surrounding environment. The playback device determines filter coefficients based on a difference between the determined frequency response curve and a target frequency response curve, and applies the filter coefficients to one or more filters during playback of audio content by the playback device. 1. A playback device comprising:one or more speakers;one or more microphones;one or more processors; and emitting, via the one or more speakers, a first audio signal;', 'while emitting the first audio signal, detecting, via the one or more microphones, data representing a second audio signal, the second audio signal comprising one or more reflections of the first audio signal as emitted by the one or more speakers within a surrounding environment;', 'determining a frequency response curve based on the second audio signal, the frequency response curve representing acoustic characteristics of a system comprising the playback device and the surrounding environment;', 'determining filter coefficients based on a difference between the determined frequency response curve and a target frequency response curve; and', 'applying the determined filter coefficients to one or more filters during playback of audio content by the playback device, wherein the applied filter coefficients cause the ...

Loudness-Based Audio-Signal Compensation

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. 1. A system for loudness-based audio-signal compensation , comprising:(a) an input line that accepts an input audio signal;(b) a loudness estimator, coupled to the input line, that processes the input audio signal to obtain an audio playback loudness level and that also provides a production loudness level for the input audio signal;(c) a filter generator/selector coupled to an output of the loudness estimator that provides an audio compensation filter based on the production loudness level and the audio playback loudness level,wherein the input signal is processed by the audio compensation filter to provide an output signal, andwherein when the loudness estimator obtains at least one of a new audio playback loudness level or a new production loudness level for the input audio signal, the filter generator/selector provides a new audio compensation filter which is then used to process the input signal in order to provide the output signal.2. A system according to claim 1 , wherein the output signal is coupled to an audio output device that comprises at least one of a speaker or a headphone.3. A system according to claim 1 , wherein the production loudness level is fixed based on a setting provided by a user.4. A system according to claim 3 , wherein the setting provided by the user is a selected deviation ...

Loudness Matching

An example method may involve a device determining a first loudness representation for a playback device based on a first equalization setting applied to a representation of average music. The device may also determine a second loudness representation for the playback device, based on a second equalization setting applied to the representation of average music. The device may also determine a loudness adjustment factor based on the first and second loudness representations, and then causing the playback device to play back media based on the second equalization setting and the determined loudness adjustment factor. 1. A device comprising:a processor;a non-transitory computer-readable medium; and determining a first loudness representation for a playback device, wherein the playback device comprises a first equalization setting, and wherein the first loudness representation is based on the first equalization setting applied to a representation of average music;', 'determining a second loudness representation for the playback device, wherein the playback device comprises a second equalization setting, and wherein the second loudness representation is based on the second equalization setting applied to the representation of average music;', 'determining a loudness adjustment factor based on the first and second loudness representations; and', 'causing the playback device to play back media based on the second equalization setting and the determined loudness adjustment factor., 'program instructions stored on the non-transitory computer-readable medium that, when executed by the processor, cause the device to perform operations comprising2. The device of claim 1 , wherein the first and second equalization settings are associated with a first and second media source claim 1 , respectively.3. The device of claim 1 , wherein the representation of average music comprises a power-frequency dropoff curve.4. The device of claim 1 , wherein the program instructions claim 1 , ...

Audio system with noise interference mitigation

Generally, this disclosure provides systems, devices, methods and computer readable media for an audio device with noise interference mitigation capability. The audio device may include an interference detection module to detect an interfering voice audio signal received from an interfering audio source. The device may also include an interference identification module to identify the interfering audio source. The device may further include an impact estimation module to estimate an impact of the interfering voice audio signal on a user of the audio device and a communication module to transmit a volume correction request to the interfering audio source. The volume correction request may be based on the identification and the estimated impact. Other embodiments may be described and claimed. 1. An audio device with noise interference mitigation , said device comprising:an interference detection module to detect an interfering voice audio signal received from an interfering audio source;an interference identification module to identify said interfering audio source;an impact estimation module to estimate an impact of said interfering voice audio signal on a user of said audio device; anda communication module to transmit a volume correction request to said interfering audio source, said volume correction request based on said identification and said estimated impact.2. The device of claim 1 , wherein said impact estimation module is further to measure volume and persistence of said interfering voice audio signal claim 1 , and to estimate distance between said audio device and said interfering audio source claim 1 , wherein said estimated impact is based on said measured volume claim 1 , said measured persistence or said estimated distance.3. The device of claim 1 , wherein said communication module is further to receive a location broadcast claim 1 , said location broadcast comprising a location associated with said interfering audio source claim 1 , wherein said ...

DEVICE AND METHOD FOR VOLUME CONTROL

A device for volume control for at least two audio sources includes a loudness analyzer and a volume regulator. The loudness analyzer is configured to analyze an audio signal of the first one and of the second one of the two audio sources over a time period, to determine a first and a second loudness value, respectively, as a function thereof, and to store same while associating them with the first and the second audio source. The volume regulator is configured to adapt the audio signal of the currently selected first and/or second one of the two audio sources in accordance with the corresponding first and/or second loudness value. 1. A device for volume control for at least two audio sources , comprising:a loudness analyzer configured to analyze an audio signal of the first one of the two audio sources over a time period and to determine a first loudness value as a function thereof, and to store said loudness value while associating it with the first one of the two audio sources, and to analyze an audio signal of the second one of the two audio sources over a time period, and to determine a second loudness value as a function thereof, and to store said loudness value while associating it with the second one of the two audio sources;a volume regulator configured to adapt the audio signal of the currently selected first one of the two audio sources in accordance with the corresponding first loudness value and to adapt the audio signal of the currently selected second one of the two audio sources in accordance with the corresponding second loudness value;wherein the time period in a learning phase amounts to at least 30 seconds, advantageously at least 60 or 150 seconds, for each audio source, wherein the loudness analyzer performs the learning phase when no loudness value has yet been stored for the respective audio source;wherein the time period during an adaptation phase amounts to at least 5 seconds, or advantageously at least 60 seconds or 300 seconds for the ...

Method and Apparatus for Adjusting Volume of User Terminal, and Terminal

A volume adjustment method and apparatus, and a terminal is presented. Perform analysis on the collected sound signal surrounding a user terminal, to obtain composition information, where the composition information includes sound types included in the sound signal and proportions of sounds of the various types, and the sound types include blank sound, human sound, and noise; determine a current scene mode of the user terminal according to the composition information; and adjust volume of the user terminal according to the determined scene mode, thereby significantly reducing occurrence of a case, caused by mistaken determining of the scenario, in which play volume adjustment does not conform to the scenario, and enhancing user experience. 1. A method for adjusting volume of a user terminal , comprising:collecting a sound signal surrounding the user terminal;performing analysis on the collected sound signal in order to obtain composition information of the sound signal, wherein the composition information comprises sound types comprised in the sound signal and proportions of sounds of various types, and wherein the sound types comprise blank sound, human sound, and noise;determining a current scene mode of the user terminal according to the composition information of the sound signal; andadjusting a volume of the user terminal according to the determined scene mode.2. The method according to claim 1 , wherein collecting the sound signal surrounding the user terminal comprises collecting a sound signal of a current environment in which the user terminal is located in response to detection of a call signal arrival.3. The method according to claim 1 , wherein collecting the sound signal surrounding the user terminal comprises periodically collecting a sound signal of a current environment in which the user terminal is located.4. The method according to claim 1 , wherein adjusting the volume of the user terminal according to the determined scene mode comprises: ...

AUTOMATIC VOLUME CONTROL FOR LAND MOBILE RADIO

The present disclosure provides a radio and method for automatically adjusting, in response to ambient noise, the volume setting of a radio. The radio comprises a microphone for receiving ambient audio and generating a microphone signal; a codec for receiving the microphone signal and generating a processor signal representative of the ambient audio; and processor circuitry operable to receive the processor signal, determine an ambient audio level in response to the processor signal, and determine an adjusted radio volume to generate a radio volume control signal, wherein the adjusted radio volume is determined by calculating a difference between a baseline volume level and ambient audio level and adding a current volume level setting, wherein the codec is also operable to receive the radio volume control signal and generate an output signal to adjust the radio volume setting to maintain a net difference between the adjusted radio volume and ambient audio. 1. A land mobile radio for use in a land mobile radio (LMR) system , the land mobile radio configured to automatically adjust a volume setting of the land mobile radio in response to ambient audio , the land mobile radio comprising:a first microphone configured to receive the ambient audio and generate a microphone signal in response to receiving the ambient audio;an analog-to-digital converter configured to receive the microphone signal and generate a processor input signal representative of the ambient audio;first circuitry configured to receive the processor input signal, determine an ambient audio level in response to the processor input signal, and determine an adjusted radio volume to generate a radio volume control signal, wherein the adjusted radio volume is determined by calculating a difference between a baseline volume level and the ambient audio level and adding a current volume level setting; anda digital-to-analog converter configured to receive the radio volume control signal and generate an output ...

AUDIO CONTROL SYSTEM

The invention includes a system for classifying sounds using a microphone, microphone array and/or a camera. The system can classify sounds and respective levels at different zones in an area such as a room, hall or vehicle passenger compartment. The system can use Active Noise Cancellation (ANC) to reduce unwanted sound and Automatic Loudness Control (ALC) to adjust levels of audio output such as music. The system can maintain a loudness ratio in each zone to create an optimal environment for music listening and/or conversation. The system can also be used to optimize the sound levels of broadcasted messages such as service announcements and advertisements. 1. A system for controlling of audio output in an area , said system comprising:a first sensor to capture audio data corresponding to a human;a second sensor to capture video data of the human, wherein the video data comprises a lip region of the human;an audio output device; and analyze the video data to determine movement of lips of the human; and', 'adjust audio output from the audio output device based on detection of presence of human speech, and, 'an automatic loudness control unit connected to the first sensor, the second sensor and the audio output device towherein the automatic loudness control unit detects the presence of human speech based at least on analysis of the movement of lips of the human,wherein the automatic loudness control unit detects whether any other color different from the lips is present at a region between the lips, and judges the presence of human speech occurred based on a frequency of detection of the any other color at the region between the lips within a predetermined period, andwherein the automatic loudness control unit simultaneously reduces a volume of the audio output during the human speech being present.2. The system of claim 1 , wherein active noise cancellation is applied to the area to decrease background noise.3. The system of claim 1 , wherein the first sensor ...

COMPENSATION FOR AMBIENT SOUND SIGNALS TO FACILITATE ADJUSTMENT OF AN AUDIO VOLUME

Embodiments of the invention relate generally to audio device and wearable computing devices to detect and characterize an ambient sound, to adjust an output volume of an audio device. More specifically, disclosed are systems, components and methods to generate audio signals associated with an audio device and an output volume, receive ambient sound signals associated with an ambient sound source, detect the ambient sound signals reaching a threshold intensity, analyzing digital data representing the ambient sound and adjust the audio signals to change the output volume according to a category of the ambient sound signals.

Dynamic Sound Adjustment

Among other things, one or more non-transitory machine-readable media storing instructions are described. The stored instructions are executable by one or more processing devices to perform operations comprising analyzing an effect of noise in a spatial unit on sound in the spatial unit, at least part of the sound being produced by audio signals; selecting an adjustment curve among a group of adjustment curves based on one or more characteristics of the noise; and determining an amount of adjustment to be made to the audio signals based on the analyzed effect and the selected adjustment curve. 1. One or more non-transitory machine-readable media storing instructions that are executable by one or more processing devices to perform operations comprising:obtaining a first signal that is present in a cabin of a vehicle comprising a first unwanted noise signal and a second desired audio signal that is produced by a vehicle audio system of the vehicle provided with an input signal;adaptively filtering the first obtained signal to form an estimate of the first unwanted noise signal;comparing a first measurement in a first frequency band of the estimated first unwanted noise signal with a second measurement in a second frequency band of the estimated first unwanted noise signal, the second frequency band being different from the first frequency band;determining a first operating condition of the vehicle;selecting, based on both the comparison and the determined first operating condition, a first gain to be applied to the vehicle audio system input signal to compensate for noise interference, and;applying the first gain to one or more first signal components of the vehicle audio system input signal.2. The one or more non-transitory machine-readable media of claim 1 , wherein comparing the first noise measurement with the second noise measurement comprises calculating a ratio of the first noise measurement in the first frequency band and the second noise measurement in the ...

PERSONALIZED ADJUSTMENT OF AN AUDIO DEVICE

User adjustment of an audio effect of an audio device is facilitated to match a hearing sensitivity of a user. The user can tune the audio device with a minimum perceptible level unique to the user. The audio device can adjust the audio effect in accordance with the minimum perceptible level. For example, a loudness level can adjust automatically to ensure that the user maintains a perceptible loudness, adjusting according to environmental noise and according to the minimum perceptible level. Also described herein are apparatuses, systems and methods related to an audio device equipped with embedded audio sensors that can maximize a voice quality while minimizing the effects of noise. 1. A system , comprising:a memory that stores computer-executable components; and a tuning component configured to generate first tuning data based on a tuning process associated with a first user and performed by a user interface of a device; and', 'a noise suppression component configured to adjust an audio signal associated with speech data based on a digital transformation that employs the first tuning data and second tuning data generated based on a defined tuning value that is associated with a second user identity that is different than the first user identity., 'a processor, communicatively coupled to the memory, that executes or facilitates execution of the computer-executable components, the computer-executable components comprising2. The system of claim 1 , wherein the noise suppression component is configured to adjust the audio signal in response to a determination that noise data indicative of environmental noise is above a defined threshold.3. The system of claim 2 , wherein the noise suppression component is configured to adjust the audio signal based on the noise data.4. The system of claim 1 , wherein the noise suppression component is configured to adjust the audio signal based on a set of filter bands associated with the digital transformation.5. The system of claim ...

TECHNIQUES FOR USING COMPUTER VISION TO ALTER OPERATION OF SPEAKER(S) AND/OR MICROPHONE(S) OF DEVICE

In one aspect, a first device includes at least one processor and storage accessible to the at least one processor. The storage includes instructions that may be executable by the processor to receive input from a camera and identify a second device based on the input from the camera. The second device may include at least one speaker and at least one microphone. The instructions may also be executable to identify a current location of the second device within an environment based on the input from the camera and to identify a current location of an object within the environment that is different from the second device. The instructions may then be executable to provide a command to alter operation of the at least one speaker and/or the at least one microphone based on the current location of the second device and the current location of the object. 1. A first device , comprising:at least one processor; andstorage accessible to the at least one processor and comprising instructions executable by the at least one processor to:receive input from a camera;identify a second device based on the input from the camera, the second device comprising at least one speaker and at least one microphone;identify a current location of the second device within an environment based on the input from the camera;identify a current location of at least one object within the environment that is different from the second device; andprovide a command to alter operation of one or more of the at least one speaker and the at least one microphone based on the current location of the second device and the current location of the at least one object.2. The first device of claim 1 , wherein the command is transmitted to the second device for the second device to alter operation of one or more of the at least one speaker and the at least one microphone.3. The first device of claim 1 , wherein the current locations are identified using at least one computer vision algorithm.4. The first device of ...

Adaptation of dynamic range enhancement based on noise floor of signal

An analog signal path portion of a signal path may have: (i) an audio input for receiving an analog signal, an audio output for providing an output signal, and a selectable analog gain, and may be configured to generate the output signal based on the analog signal and in conformity with the selectable analog gain; and (ii) a digital path portion having a selectable digital gain and configured to receive a digital input signal and convert the digital input signal into the analog signal in conformity with the selectable digital gain. A control circuit may be configured to modify the digital and analog gains in response to an indication to switch between gain modes of the signal path, determine a noise floor of an audio signal comprising the digital input signal or a signal derived therefrom, and control modification of the digital and analog gains based on the noise floor.

AUTOMATIC ADJUSTMENTS OF AUDIO ALERT CHARACTERISTICS OF AN ALERT DEVICE USING AMBIENT NOISE LEVELS

The automatic adjustment of audio alert characteristics of an alert device using ambient noise levels is described. In one aspect of the invention, a machine-readable medium has executable instructions to cause a machine to perform a method to receive an audio sample of ambient noise and adjust a characteristic of the audio alert, such as, the volume level of the audio alert, based on the ambient noise level. 141-. (canceled)42. A machine readable medium having instructions stored thereon , which when executed by one or more machines , cause the machines to:sample ambient audio corresponding to receipt of a signal;select an audio alert from a plurality of audio alerts based on parameters of the sampled ambient audio, wherein the audio alert is different from the ambient audio; andgenerate the audio alert as an audible audio alert.43. The machine-readable storage medium of claim 42 , wherein generating the audio alert comprises adjusting characteristics of the audio alert.44. The machine-readable storage medium of claim 43 , wherein the audio alert characteristics are adjusted as a function of the parameters of the sampled ambient audio.45. The at least one machine-readable storage medium of claim 44 , wherein a decibel level of the audio alert is adjusted as a function of a decibel level of the ambient sound sample.46. The at least one machine-readable storage medium of claim 44 , wherein a frequency of the audio alert is adjusted as a function of a frequency of the ambient sound sample.47. The at least one machine-readable storage medium of claim 43 , wherein the characteristics of the audio alert are adjusted according to an audio alert rule.48. The at least one machine-readable storage medium of claim 47 , wherein the audio alert rule includes instructions to adjust the audio alert characteristics based on the parameters of the ambient audio.49. The at least one machine-readable storage medium of claim 48 , wherein the instructions to adjust the audio alert ...

SYSTEM AND METHOD TO SILENCE OTHER DEVICES IN RESPONSE TO AN INCOMING AUDIBLE COMMUNICATION

A user's set top box detects the ringing of a telephone, mobile device or other type of audible notification of an incoming call or message and, in response, sends a command causing a current audio level of the television programming or other media currently being played to decrease such that the audible notification can be heard by the user. A device silencing manager on the set top box may detect the audible notification occurring by the sound waves of the audible notification from the mobile device picked up by a microphone of the receiving device. Additionally or alternatively, a device silencing manager on the set top box may wirelessly receive, either directly or indirectly from the mobile device, an electronic notification of the audible notification. 1. A method in a device silencing manager system , the method comprising:receiving, by at least one computer processor that is in control of the device silencing manager system, information regarding an audible notification of an incoming contact;detecting, by the at least one computer processor, that the audible notification is occurring based on the received information;in response to the detection that the audible notification is occurring, deciding, by the at least one computer processor, whether to cause a current audio level of a device to decrease; andcausing, by the at least one computer processor, the current audio level of the device to decrease based on the decision.2. The method of wherein the device is an electronic device selected from the group comprising of: a television claim 1 , a speaker claim 1 , a sound system claim 1 , a stereo claim 1 , a radio claim 1 , earphones claim 1 , a mobile device claim 1 , an electronic toy claim 1 , an electronic game claim 1 , a computer and a gaming system.3. The method of wherein the device silencing manager system is in a receiving device.4. The method of wherein the receiving device is a set top box.5. The method of wherein the audible notification is an ...

SIGNAL PROCESSING DEVICE, METHOD, SPEAKER AND ELECTRONICS APPARATUS

The present invention discloses a signal processing device, a signal processing method and a speaker. The signal processing device comprises a multi-band dynamic range controller, wherein the multi-band dynamic range controller receives an audio signal and includes a first band splitting unit and a resonant band adjustment unit; the first band splitting unit is configured to split the audio signal into multiple bands and obtain at least one resonant band therefrom, which has a resonant frequency band signal in a resonant frequency range of the audio signal; and the resonant band adjustment unit is configured to adjust the resonant frequency band signal based at least on a resonant band dynamic range control gain and output an adjusted resonant frequency band signal for combination with other band signals into a compression output signal. 1. A signal processing device for a speaker , comprising a multi-band dynamic range controller ,wherein the multi-band dynamic range controller receives an audio signal and includes a first band splitting unit and a resonant band adjustment unit;the first band splitting unit is configured to split the audio signal into multiple bands and obtain at least one resonant band therefrom, which has a resonant frequency band signal in a resonant frequency range of the audio signal; andthe resonant band adjustment unit is configured to adjust the resonant frequency band signal based at least on a resonant band dynamic range control gain and output an adjusted resonant frequency band signal for combination with other band signals into a compression output signal.2. The signal processing device according to claim 1 , wherein the resonant frequency range covers a predetermined frequency range with the resonant peak of the speaker as its center frequency.3. The signal processing device according to claim 1 , wherein the resonant band adjustment unit further includes:a resonant band displacement predictor, which generates a resonant band ...

Audio Settings Based On Environment

Techniques described herein may involve audio settings based on an environment. An example implementation may involve a playback device playing back first audio content and during playback of at least a portion of the first audio content, detecting, via the at least one microphone, an audio signal. At least a portion of the detected audio signal may include a reflection of the first audio content played back by the playback device. The playback device may determine an equalization setting based on at least the determined one or more reflection characteristics and apply the determined equalization setting during playback of second audio content. 1. A system comprising a playback device , a server , and a control device , play back first audio content via the audio transducer;', 'during playback of at least a portion of the first audio content, detect, via the at least one microphone, an audio signal, wherein at least a portion of the detected audio signal comprises a reflection of the first audio content played back by the playback device;', 'determine one or more reflection characteristics based on at least the detected audio signal;', 'determine an equalization setting based on at least the determined one or more reflection characteristics; and', 'apply the determined equalization setting during playback of second audio content via the audio transducer; and, 'wherein the playback device comprises a first network interface, at least one microphone within a housing of the playback device, an audio transducer within the housing, at least one first processor and at least one first non-transitory computer-readable medium including instructions that are executable by the at least one first processor such that the playback device is configured to 'send, via the second network interface to the playback device, data representing one or more instructions to play the second audio content;', 'wherein the control device comprises a second network interface, at least one second ...

Devices and Methods for Audio Volume Adjustment

Systems, devices and methods are provided for audio volume adjustment. For example, a noise volume level of current ambient noise and a first volume level of a first current audio signal of a first application are acquired using a sound collecting device. The first volume level and the noise volume level are compared using one or more data processors to obtain a first comparison result. The first volume level is adjusted based on at least information associated with the first comparison result and a reference volume level.

SYSTEM AND METHOD FOR AUTONOMOUS MULTI-TRACK AUDIO PROCESSING

A method, apparatus, and computer readable medium for mixing a plurality of audio signals within an audio mixing system are disclosed. The method comprises receiving a plurality of audio signals, extracting at least one audio feature from each of the plurality of audio signals, determining, for each of the audio signals, an associated processing control function in accordance with a plurality of the extracted audio features, processing each audio signal in accordance with each associated processing control function, and outputting the plurality of processed audio signals. 121-. (canceled)22. A method for automatic audio mixing , the method comprising:receiving at least one audio track;extracting at least one feature from each of the at least one audio track;enabling a configuration for each of at least one processing control function for the at least one audio track to be controlled;processing each audio track using the at least one processing control function configured for that track, in accordance with the at least one extracted feature; andoutputting at least one processed audio track.23. The method of claim 22 , further comprising performing a feedback operation to compare at least one output produced to a desired result prior to said outputting the at least one processed audio track.24. The method of claim 22 , further comprising analyzing a relationship between at least one pair of extracted features from at least one track claim 22 , and processing each audio track using the at least one processing control function in accordance with the at least one extracted feature and the relationship.25. The method of claim 22 , wherein configuring the at least one processing function comprises establishing at least one subgroup of audio tracks having a particular configuration of the at least one processing control function.26. The method of claim 25 , wherein a plurality of sub-groups are configured claim 25 , at least a pair of the plurality of sub-groups being ...

ELECTRONIC APPARATUS AND CONTROL METHOD THEREOF

An electronic apparatus and a method for controlling the electronic apparatus are provided. The electronic apparatus includes an audio processor that outputs an audio including a first audio component and a second audio component and a controller that controls the audio processor to change a ratio of the first audio component to the second audio component according to the changed audio level, in response to receiving an instruction to change a level of the output audio. 1. An apparatus comprising:an audio processor configured to output an audio including a first audio component and a second audio component; anda controller configured to control the audio processor to, in response to receiving an instruction to change a level of the output audio, change a ratio of the first audio component to the second audio component based on the changed audio level and a predetermined condition,wherein the predetermined condition comprises a condition input from a user, a condition received from an outside, or a condition recognized by the apparatus, andwherein the controller is configured to control the audio processor to change the ratio of the first audio component by increasing a proportion of the first audio component compared to the second audio component based on the changed audio level being smaller than a first volume, and increasing a proportion of the second audio component compared to the first audio component based on the changed audio level being larger than a second volume.2. The apparatus according to claim 1 , wherein the controller is further configured to control the audio processor to separate the audio into the first audio component and the second audio component and to adjust a gain of at least one of the first audio component and the second audio component to change the ratio of the first audio component to the second audio component.3. The apparatus according to claim 1 , wherein the controller is further configured to control the audio processor to ...

ENTERTAINMENT AUDIO PROCESSING

An entertainment audio processing system includes a music limiter configured to receive an audio signal, volume control information for music, and volume control information for announcement. The music limiter is further configured to calculate a gain parameter based on a music-to-announcement ratio. In another aspect, an entertainment audio processing system includes a mixer configured to substantially match a left audio signal channel and a right audio signal channel. The entertainment audio processing system further includes an equalizer configured to introduce magnitude and phase changes between the left audio signal channel and the right audio signal channel. 1. An apparatus comprising:a mixer configured to at least partially collapse an audio signal to mono by making a left audio signal channel and a right audio signal channel similar;an equalizer coupled to an output of the mixer, the equalizer configured to introduce magnitude and phase differences between the left audio signal channel and the right audio signal channel to generate a processed audio signal;a latency compensator configured to receive an unprocessed audio signal and introduce a time delay that matches a time delay associated with the processed audio signal to generate a time-delayed audio signal; andan x-fade coupled to an output of the equalizer and an output of the latency compensator, the x-fade configured to output a digital audio signal by selecting to output one of the processed audio signal or the time-delayed audio signal based, at least in part, upon an announcement trigger signal received from a non-entertainment audio processing system.2. The apparatus of claim 1 , wherein:the announcement trigger signal provides an indication that an announcement audio is present.3. The apparatus of claim 1 , where in the time delay associated with the processed audio signal is due to at least one of the mixer and the equalizer.4. The apparatus of claim 1 , wherein the magnitude and phase ...

Apparatus and Method for Intelligent Audio Levels for Car, Home or Public Entertainment

The present principles generally relate to audio apparatus, methods, and computer program products and in particular, to improvements that adjust the sound level or levels of one or more audio outputs of an audio system based on the determined origin and/or direction of propagation of a detected human voice in a location. Such an adjustment may be to decrease, mute, or increase the sound level of an audio output producing sound in the direction of the origin of the voice. A sound level produced by other audio outputs may be unchanged. 2. The apparatus of wherein the adjustment comprises one of a decrease claim 1 , a muting claim 1 , and an increase in the level of the at least one sound output of the plurality of sound outputs of the one or more audio drivers.3. The apparatus of wherein the one or more processors are further configured to maintain a level of sound output of other of the plurality of sound outputs of the one or more audio output drivers not being projected in the direction toward the determined origin of the voice.4. The apparatus of wherein the detector comprises a plurality of microphones claim 1 , and the plurality of microphones are configured to provide one or more inputs to the one or more processors for the determining whether the ambient noise includes a voice of a person in the location.5. The apparatus of further comprising a camera wherein the camera is further configured to provide one or more inputs to the one or more processors for the determining whether the ambient noise includes a voice of a person in the location claim 1 , and wherein the camera is used to determine a presence of a human in the location.6. The apparatus of wherein the one or more processors is further configured to determine one or more of the plurality of sound outputs of the one or more audio output drivers located closest to the determined origin of the voice claim 1 , and enable the adjustment in a sound level of the determined one or more of the plurality of ...

SOUNDER DYNAMIC VOLUME ADJUSTMENT

A sounder apparatus includes a sounder having at least two volume settings, at least one detector , a micro-controller wherein the micro controller is configured to receive an input signal from the at least one detector , and to control the volume setting of the sounder based on the input signal. 1. A sounder apparatus comprising ,a sounder having at least two volume settings,at least one detector, anda micro-controller,wherein the micro controller is configured to receive an input signal from the at least one detector, and to control the volume setting of the sounder based on the input signal.2. A sounder apparatus as claimed in claim 1 , further comprising a potentiometer for manual input of a desired volume claim 1 , wherein the micro controller receives an input signal from the potentiometer and is further configured to control the volume setting of the sounder based on the input signal from the potentiometer claim 1 , and wherein when the micro-controller receives an input signal from at least one detector and an input signal from the potentiometer claim 1 , the micro-controller controls the volume of the sounder based on the input signal from the potentiometer combined with the input signal from the at least one detector.3. A sounder apparatus as claimed in claim 2 , wherein when the micro-controller receives an input signal from at least one detector and an input signal from the potentiometer claim 2 , the micro-controller controls the volume of the sounder based on the potentiometer input which overrides the input signal from the at least one detector.4. A sounder apparatus as claimed in claim 1 , wherein the at least one detector includes one or more of: a detector for detecting the volume of ambient sound in the space claim 1 , a detector for detecting the dimensions of a space claim 1 , a detector for detecting the spatial configuration of a space and/or a detector for detecting the occupation of a space.5. A sounder apparatus as claimed in claim 1 , ...

Apparatus and method for automatic volume control with ambient noise compensation

An electronic device and method that automatically adjusts an audio output volume level based on a live environmental acoustic scenario input via a microphone using a machine learning algorithm trained with Human Activity Recognition (HAR). Equipped with such an intelligence the electronic device classifies ambient sounds occurring in the environment of the listening area in which the device is situated into different acoustic scenario mappings such a voice or conversation, for an ambient human conversation detected event, and noise, such as for example a vacuum cleaner or dish washer noise detected event, and automatically adjust the audio output volume accordingly.

APPARATUS, SYSTEMS AND METHODS FOR BUFFERING OF MEDIA CONTENT

Volume limiting systems and methods are operable to limit volume output from media presentation devices. An exemplary embodiment detects a sound using a microphone, wherein the sound corresponds to an audio output of at least one controlled media presentation device, and wherein the microphone is remotely located from the at least one controlled media presentation device; compares a level of the detected sound with a predefined maximum volume limit; generates a volume output limit command in response to the detected sound exceeding the predefined maximum volume limit; and communicates the volume output limit command to the media presentation device. The media presentation device then reduces a volume level of its audio output. In some instances, volume may be limited during user specified periods. 1. A media device configured to limit volume output from media presentation devices , comprising:an interface configured to receive a user specification that defines an enforcement period;an audio detector interface configured to receive audio information communicated by an audio detector; and determine a volume level of the detected sound based on the audio information received from the audio detector;', 'determine a start of the enforcement period based on the enforcement period;', 'compare the volume level of the detected sound with a predefined maximum volume limit in response to a real time reaching the start time of the enforcement period; and', 'generate a volume output limit command in response to the detected sound exceeding the predefined maximum volume limit during the enforcement period,, 'a processor system communicatively coupled to the audio detector interface and the interface, wherein the processor system is configured towherein the volume output limit command is configured to reduce the output volume of a controlled media presentation device that is presenting the detected sound.2. The media device of claim 1 , further comprising:a memory configured to ...

METHOD AND APPARATUS FOR MODE BALANCE FOR ANALOG FM, DIGITAL RADIO BLEND LOGIC IN AN AUTOMOTIVE ENVIRONMENT

A radio includes a first tuner and a second tuner. A processor compares a first perceivable volume level of a station tuned by the first tuner to at least one second perceivable volume level of at least one background station tuned by the second tuner. The processor enables automatic volume knob changes using a pre-calibrated lookup table that associates a volume step of the volume knob with a difference between the first perceivable volume level and the second perceivable volume level. 1. A radio , comprising:a first tuner and a second tuner; anda processor configured to:compare a first perceivable volume level of a station tuned by the first tuner to at least one second perceivable volume level of at least one background station tuned by the second tuner; andenable automatic volume knob changes using a pre-calibrated lookup table that associates a volume step of the volume knob with a difference between the first perceivable volume level and the second perceivable volume level.2. The radio of claim 1 , wherein the automatic volume knob changes comprise volume increases or volume decreases depending upon an existing volume level of the radio.3. The radio of claim 1 , wherein the station tuned by the first tuner is an analog station.4. The radio of claim 1 , wherein the background station is an analog station.5. The radio of claim 1 , comprising temporally computing an average energy of the station tuned by the first tuner claim 1 , the first perceivable volume level of the station tuned by the first tuner being determined dependent upon the temporally computed average energy.6. The radio of claim 1 , comprising temporally computing an average energy of the background station claim 1 , the second perceivable volume level of the background station being determined dependent upon the temporally computed average energy.7. The radio of claim 1 , wherein the processor is configured to implement the automatic volume knob changes in association with switching from audibly ...

Methods and Apparatus For Adjusting A Level of An Audio Signal

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal. 1. A method for adjusting a level of an audio signal in an audio processing apparatus , the method comprising:dividing an audio signal into a plurality of frequency bands;delaying at least one of the plurality of the frequency bands;obtaining modification parameters for at least one of the plurality of frequency bands, the modification parameters comprising filter coefficients and amplitude scale factors, each of the amplitude scale factors respectively operating in a frequency band of the plurality of frequency bands and each amplitude scale factor representing an average energy over the frequency band and a time segment;deriving gain factors for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors;smoothing the gain factors;determining a level of noise from noise compensation factors;applying the gain factors to at least one of the frequency bands to generate gain adjusted frequency bands;adjusting the level of noise based on the gain adjusted frequency bands;filtering at least one of the frequency bands with a filter generated with the filter coefficients; ...

Methods and Apparatus For Adjusting A Level of An Audio Signal

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal. 1. A method for adjusting a level of an audio signal in an audio processing apparatus , the method comprising:dividing an audio signal into a plurality of frequency bands;obtaining modification parameters for at least one of the plurality of frequency bands, the modification parameters comprising filter coefficients and amplitude scale factors, each of the amplitude scale factors respectively operating in a frequency band of the plurality of frequency bands and each amplitude scale factor representing an average energy over the frequency band and a time segment;deriving gain factors for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors;smoothing the gain factors;determining a level of noise from noise compensation factors;applying the gain factors to at least one of the frequency bands to generate gain adjusted frequency bands;adjusting the level of noise based on the gain adjusted frequency bands;filtering at least one of the frequency bands with a filter generated with the filter coefficients, wherein the filter coefficients are applied to time-varying filters; ...

Methods and Apparatus For Adjusting A Level of An Audio Signal

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal. 1. A method for adjusting a level of an audio signal in an audio processing apparatus , the method comprising:dividing an audio signal into a plurality of frequency bands;obtaining modification parameters for at least one of the plurality of frequency bands, the modification parameters comprising filter coefficients and amplitude scale factors, each of the amplitude scale factors respectively operating in a frequency band of the plurality of frequency bands and each amplitude scale factor representing an average energy over the frequency band and a time segment;deriving gain factors for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors;smoothing the gain factors, wherein the smoothing of the gain factors is optional;determining a level of noise from noise compensation factors;applying the gain factors to at least one of the frequency bands to generate gain adjusted frequency bands;adjusting the level of noise based on the gain adjusted frequency bands;filtering at least one of the frequency bands with a filter generated with the filter coefficients; ...

Methods and Apparatus For Adjusting A Level of An Audio Signal

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal. 1. A method for adjusting a level of an audio signal in an audio processing apparatus , the method comprising:dividing an audio signal into a plurality of frequency bands;delaying at least one of the plurality of the frequency bands;obtaining modification parameters for at least one of the plurality of frequency bands, the modification parameters comprising filter coefficients and amplitude scale factors, each of the amplitude scale factors respectively operating in a frequency band of the plurality of frequency bands;deriving gain factors for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors;smoothing the gain factors;determining a level of noise from noise compensation factors;applying the gain factors to at least one of the frequency bands to generate gain adjusted frequency bands;adjusting the level of noise based on the gain adjusted frequency bands;filtering at least one of the frequency bands with a filter generated with the filter coefficients, wherein the filter coefficients are applied to time-varying filters; andsynthesizing the plurality of frequency ...

Methods and Apparatus For Adjusting A Level of An Audio Signal

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal. 1. A method for adjusting a level of an audio signal in an audio processing apparatus , the method comprising:dividing an audio signal into a plurality of frequency bands;delaying at least one of the plurality of the frequency bands;obtaining modification parameters for at least one of the plurality of frequency bands, the modification parameters comprising filter coefficients and amplitude scale factors, each of the amplitude scale factors respectively operating in a frequency band of the plurality of frequency bands;deriving gain factors for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors;smoothing the gain factors, wherein the smoothing of the gain factors is optional;determining a level of noise from noise compensation factors;applying the gain factors to at least one of the frequency bands to generate gain adjusted frequency bands;adjusting the level of noise based on the gain adjusted frequency bands;filtering at least one of the frequency bands with a filter generated with the filter coefficients, wherein the filter coefficients are applied to time- ...

Methods and Apparatus For Adjusting A Level of An Audio Signal

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal. 1. A method for adjusting a level of an audio signal in an audio processing apparatus , the method comprising:dividing an audio signal into a plurality of frequency bands;delaying at least one of the plurality of the frequency bands;obtaining modification parameters for at least one of the plurality of frequency bands, the modification parameters comprising filter coefficients and amplitude scale factors, each of the amplitude scale factors respectively operating in a frequency band of the plurality of frequency bands and each amplitude scale factor representing an average energy over the frequency band and a time segment;deriving gain factors for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors;smoothing the gain factors;determining a level of noise from noise compensation factors;applying the gain factors to at least one of the frequency bands to generate gain adjusted frequency bands;adjusting the level of noise based on the gain adjusted frequency bands;filtering at least one of the frequency bands with a filter generated with the filter coefficients, wherein ...

Methods and Apparatus For Adjusting A Level of An Audio Signal

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal. 1. A method for adjusting a level of an audio signal in an audio processing apparatus , the method comprising:dividing an audio signal into a plurality of frequency bands;delaying at least one of the plurality of the frequency bands;obtaining modification parameters for at least one of the plurality of frequency bands, the modification parameters comprising filter coefficients and amplitude scale factors, each of the amplitude scale factors respectively operating in a frequency band of the plurality of frequency bands and each amplitude scale factor representing an average energy over the frequency band and a time segment;deriving gain factors for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors;smoothing the gain factors, wherein the smoothing of the gain factors is optional;determining a level of noise from noise compensation factors;applying the gain factors to at least one of the frequency bands to generate gain adjusted frequency bands;adjusting the level of noise based on the gain adjusted frequency bands;filtering at least one of the frequency bands with a ...

Methods and Apparatus For Adjusting A Level of An Audio Signal

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal. 1. A method for adjusting a level of an audio signal in an audio processing apparatus , the method comprising:dividing an audio signal into a plurality of frequency bands;obtaining modification parameters for at least one of the plurality of frequency bands, the modification parameters comprising filter coefficients and amplitude scale factors, each of the amplitude scale factors respectively operating in a frequency band of the plurality of frequency bands and each amplitude scale factor representing an average energy over the frequency band and a time segment;deriving gain factors for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors;smoothing the gain factors, wherein the smoothing of the gain factors is optional;determining a level of noise from noise compensation factors;applying the gain factors to at least one of the frequency bands to generate gain adjusted frequency bands;adjusting the level of noise based on the gain adjusted frequency bands;filtering at least one of the frequency bands with a filter generated with the filter coefficients, wherein the ...

Methods and Apparatus For Adjusting A Level of An Audio Signal

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal. 1. A method for adjusting a level of an audio signal in an audio processing apparatus , the method comprising:dividing an audio signal into a plurality of frequency bands;obtaining modification parameters for at least one of the plurality of frequency bands, the modification parameters comprising filter coefficients and amplitude scale factors, each of the amplitude scale factors respectively operating in a frequency band of the plurality of frequency bands;deriving gain factors for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors;smoothing the gain factors, wherein the smoothing of the gain factors is optional;determining a level of noise from noise compensation factors;applying the gain factors to at least one of the frequency bands to generate gain adjusted frequency bands;adjusting the level of noise based on the gain adjusted frequency bands;filtering at least one of the frequency bands with a filter generated with the filter coefficients, wherein the filter coefficients are applied to time-varying filters; andsynthesizing the plurality of frequency bands to ...

Methods and Apparatus For Adjusting A Level of An Audio Signal

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal. 1. A method for adjusting a level of an audio signal in an audio processing apparatus , the method comprising:dividing an audio signal into a plurality of frequency bands;delaying at least one of the plurality of the frequency bands;obtaining modification parameters for at least one of the plurality of frequency bands, the modification parameters comprising filter coefficients and amplitude scale factors, each of the amplitude scale factors respectively operating in a frequency band of the plurality of frequency bands;deriving gain factors for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors;smoothing the gain factors, wherein the smoothing of the gain factors is optional;determining a level of noise from noise compensation factors;applying the gain factors to at least one of the frequency bands to generate gain adjusted frequency bands;adjusting the level of noise based on the gain adjusted frequency bands;filtering at least one of the frequency bands with a filter generated with the filter coefficients; andsynthesizing the plurality of frequency bands to ...

INTELLIGENT AUDIO OUTPUTDEVICES

An audio output device may include one or more audio sensors configured to detect ambient sound and noises around the audio output device. The audio output device may include a controller configured to analyze the detected sound and noises and may adjust audio settings of the audio output device accordingly. In an embodiment, the controller may analyze and recognize that the detected noise is another person's attempt to get the user's attention or to talk to the user, such as audio sound of the user's name. In response, the controller may stop the audio output or lower the volume of the audio output. 120.-. (canceled)21. A computer-implemented method for automatically controlling an audio output device communicatively coupled to an audio device , the method comprising:interactively receiving, while concurrently receiving a first ambient noise, a change in audio settings of the audio output device, wherein the audio settings include a volume setting of the audio output device;detecting a first sound based on the first ambient noise;automatically generating an audio preference for training the audio output device, wherein the audio preference includes the detected first sound and the received change in the audio settings;receiving, by the trained audio output device, a second ambient noise;detecting the second sound based on the received second ambient noise; andupdating, based at least on the detected second sound and the generated audio preference, the audio settings.22. The computer-implemented method of claim 21 , the method further comprising:training, using the detected second sound and the generated audio preference, the audio output device for recognizing the first sound based on the detected second sound.23. The computer-implemented method of claim 21 , the method further comprising: the detected first sound,', 'the interactively received change in audio settings, and', 'the generated audio setting., 'training the audio output device based at least on24. The ...

ADVANCED COMMUNICATION EARPIECE DEVICE AND METHOD

An earpiece device (′) provides advanced communication to the user thereof, for controlling a total sound dose (TND), including an audio output from an audio source () reaching an ear of the user occluded at the outer ear canal () by an in-ear device (), for allowing an external ambient sound to be heard by the ear of the user, and for allowing a voice (NSV) of the user to be transmitted to a telecommunication transmission link (TTL). The present invention also contemplates a method of operation of the earpiece device (′). 1. A sound dose limiter device for controlling a total sound dose reaching an ear of a user , said limiter device comprising:an in-ear device for substantially occluding an outer ear canal of the user, said in-ear device including a speaker for providing an audio signal input from an audio output of an audio source inside the outer ear canal, an internal microphone for measuring a sound pressure level inside the occluded outer ear canal, and an external microphone for measuring an external ambient sound pressure level reaching the user's ear; anda controller device connecting to the internal and external microphones for receiving respective internal and external measured sound pressure levels and for receiving the audio output, said controller device performing an evaluation of the internal and external measured sound pressure levels, calculating a cumulative total sound dose from the internal sound pressure level and the audio signal input depending on the evaluation, and decreasing the audio signal input transmitted to the speaker based on the calculated cumulative total sound dose approaching a maximum acceptable sound dose threshold.2. The sound dose limiter device of claim 1 , wherein said controller device further calculates claim 1 , based on a level of the audio signal input claim 1 , an estimated sound dose decrease so as to account for a fatigue recuperation of the user's ear over time claim 1 , the estimated sound dose decrease being ...

Automatic Tuning of a Gain Controller

A gain control system for applying gain to a far-end signal, the system comprising: a signal identifier configured to detect an echo of the far-end signal in a microphone signal; and a path estimator configured to estimate a characteristic of an echo path of the detected echo, wherein: the signal identifier is further configured to detect a near-end signal from the microphone signal; and in response to detecting the near-end signal, the gain control system is configured to adjust the gain applied to the far-end signal in dependence on the estimated characteristic of the echo path.

METHOD, DEVICE, AND COMPUTER-READABLE MEDIUM FOR ADJUSTING VOLUME

Methods, devices, and computer-readable media for adjusting volume are provided in the disclosure. A method for adjusting volume may include acquiring, with a device, ambient noise information about ambient noise present during playing of sound. The method may further include adjusting volume of the sound based on the acquired ambient noise information. By acquiring ambient noise information, the method may provide that volume of the sound is adjusted based on the acquired ambient noise information. Because the acquired ambient noise information may reflect the current ambient noise level, the method may adjust volume of the sound based on the acquired ambient noise information without requiring manual adjusting of volume by a user, which may be convenient to the user. 1. A method for adjusting volume , comprising:acquiring, with a device, ambient noise information about ambient noise present during playing of sound; andadjusting, with the device, volume of the sound based on the acquired ambient noise information.2. The method of claim 1 , wherein acquiring the ambient noise information comprises:detecting the ambient noise information with a noise capture sensor built into the device; orreceiving the ambient noise information from a noise capture device separate from the device.3. The method of claim 1 , wherein adjusting the volume of the sound based on the acquired ambient noise information comprises:acquiring an amplitude of the ambient noise from the acquired ambient noise information;increasing the volume of the sound when the acquired amplitude of the ambient noise is equal to or above a first predetermined threshold; anddecreasing the volume of the sound when the acquired amplitude of the ambient noise is equal to or below a second predetermined threshold.4. The method of claim 1 , wherein adjusting the volume of the sound based on the acquired ambient noise information comprises:acquiring an amplitude of the ambient noise from the acquired ambient noise ...

METHOD AND SYSTEM FOR AN ULTRA LOW POWER DOSIMETER

Certain embodiments of the invention may be found in an ultra low power dosimeter assembly. The ultra low power dosimeter assembly may comprise a low power voltage source and a table circuit. The table circuit may be adapted to convert an input voltage to a second voltage level. The second voltage level may correspond to noise dose. The ultra low power dosimeter assembly may also comprise a switch adapted to trigger a control circuit. The control circuit may provide progressive attenuation of an output signal as the second voltage level increases. In certain embodiments, the control circuit may also be adapted to send one or more warning signals to a user of the ultra low power dosimeter. The one or more warning signals may be sent when the control circuit determines the second voltage level has reached one or more pre-determined threshold voltage levels. 1. An ultra low power dosimeter apparatus , comprising:a table circuit adapted to convert an input voltage to a second voltage level, wherein the second voltage level corresponds to an increment in noise dose;a switch adapted to reset the second voltage level to zero after a pre-determined threshold voltage level has been reached;a counter adapted to count the number of times the switch resets the second voltage level to zero; andan attenuation control circuit adapted to provide attenuation of an output signal as the counter reaches at least one pre-determined threshold count.2. The apparatus of further comprising an audio source adapted to provide the input voltage to the ultra low power dosimeter.3. The apparatus of wherein the audio source is a portable audio device.4. The apparatus of wherein the attenuation of the output signal is progressive as the count increases.5. The apparatus of wherein the progressive attenuation introduced into the output signal is approximately 1-10 dB at each of the at least one pre-determined threshold counts.6. The apparatus of wherein the table circuit uses a logarithmic amplifier ...

Sidetone-Based Loudness Control for Groups of Headset Users

Methods and computer-readable media embodying instructions executable by a computer to perform functions comprise: obtaining one or more indications of a sound level produced by a group of users of audio communication devices; and adjusting sidetone gains of the audio communication devices based on the one or more indications of the sound level produced by the group of users. 1. One or more non-transitory computer-readable storage media having computer-executable instructions stored thereon which , when executed by one or more computers , cause the one more computers to perform operations comprising:obtaining one or more indications of a sound level in a space, wherein a plurality of users operating a plurality of communication devices are located within the space; andadjusting a sidetone gain of the plurality of communication devices responsive to the one or more indications of the sound level in the space.2. The one or more non-transitory computer-readable storage media of claim 1 , wherein the adjusting the sidetone gain of the plurality of communication devices comprises:decreasing the sidetone gain of the plurality of communication devices responsive to the one or more indications of the sound level being above a threshold; andincreasing the sidetone gain of the plurality of communication devices responsive to the one or more indications of the sound level being below the threshold.3. The one or more non-transitory computer-readable storage media of claim 1 , wherein:the one or more indications indicate whether a user in the plurality of users has indicated that the plurality of users is talking too loudly.4. The one or more non-transitory computer-readable storage media of claim 1 , wherein:one or more of the indications of the sound level represent outputs of one or more microphones; andthe adjusting comprises comparing the indications with a threshold.5. The one or more non-transitory computer-readable storage media of claim 1 , wherein:one or more of the ...

GAIN ADJUSTMENT DEVICE, REMOTE CONVERSATION DEVICE, AND GAIN ADJUSTMENT METHOD

A gain adjustment method includes using a sensor to measure a physical quantity that varies according to a distance, and adjusting a level of an audio signal obtained by a microphone according to a measurement result of the sensor. 1. A gain adjustment device comprising:a sensor configured to measure a physical quantity that varies according to a distance;at least one memory configured to store instructions; and 'adjust a level of an audio signal obtained by a microphone according to a measurement result of the sensor.', 'at least one processor configured to execute the instructions to2. The gain adjustment device according to claim 1 , wherein the physical quantity varies according to a distance between the microphone and a sound source.3. The gain adjustment device according to claim 1 , wherein the sensor includes a plurality of microphones and measures a level ratio in the plurality of microphones as the physical quantity.4. The gain adjustment device according to claim 1 , wherein the at least one processor is further configured to execute the instructions to:extract a predetermined band of an input signal from the microphone.5. The gain adjustment device according to claim 1 ,wherein the sensor is configured to transmit electromagnetic waves and measure a time difference from transmission to reception of the electromagnetic waves as the physical quantity.6. A gain adjustment device comprising:a level adjuster configured to adjust a level of an audio signal obtained by a microphone;a sensor configured to measure a physical quantity that varies according to a distance; anda setter configured to set a gain of the level adjuster according to a measurement result of the sensor.7. The gain adjustment device according to claim 6 , wherein the physical quantity varies according to a distance between the microphone and a sound source.8. The gain adjustment device according to claim 6 , wherein the sensor includes:a plurality of microphones; anda calculator configured ...

Microphone with adjustable signal processing

A microphone may comprise a microphone element for detecting sound, and a digital signal processor configured to process a first audio signal that is based on the sound in accordance with a selected one of a plurality of digital signal processing (DSP) modes. Each of the DSP modes may be for processing the first audio signal in a different way. For example, the DSP modes may account for distance of the person speaking (e.g., near versus far) and/or desired tone (e.g., darker, neutral, or bright tone). At least some of the modes may have, for example, an automatic level control setting to provide a more consistent volume as the user changes their distance from the microphone or changes their speaking level, and that may be associated with particular default (and/or adjustable) values of the parameters attack, hold, decay, maximum gain, and/or target gain, each depending upon which DSP is being applied.

NOISE CONTROL METHOD AND DEVICE

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 1. A method , comprising:acquiring, by a device comprising a processor, noise information of an ambient environment; andjudging whether the noise information satisfies a predetermined condition, and in response to the noise information satisfying the predetermined condition, sending a noise control message to another device, the noise control message being configured to notify the other device to adjust a volume.2. The method of claim 1 , wherein the noise information comprises: a noise intensity value.3. The method of claim 2 , wherein the predetermined condition being satisfied comprises the noise intensity value being determined to be greater than a predetermined threshold.4. The method of claim 3 , wherein the noise control message comprises: the noise intensity value and the predetermined threshold.5. The method of claim 4 , wherein the noise control message further comprises: a transmit power value of the noise control message.6. The method of claim 4 , wherein the noise control message further comprises: position information of a sender of the noise control message.7. The method of claim 4 , wherein the noise information further comprises: noise ...

Respirator mask speech enhancement apparatus and method

Speech enhancement apparatus and respirator masks including speech enhancement apparatus, as well as methods of enhancing speech transmission for the wearer of a respirator mask are described herein. In one or more embodiments, the speech enhancement apparatus and methods described herein detect acoustic energy within a first frequency range in the clean air envelope of a respirator mask and deliver compensating acoustic energy outside of the clean air envelope using a speaker. The compensating acoustic energy, in one or more embodiments, exhibits a predetermined attenuated amplitude profile such that the compensating acoustic energy has an amplitude less than 6 dB greater than the acoustic attenuation profile of the mask body over at least 90% of a predetermined attenuated frequency range.

Loudness-Based Audio-Signal Compensation

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. 1. A system for loudness-based audio-signal compensation , comprising:(a) an input line that accepts an input audio signal;(b) a loudness estimator that provides an audio playback loudness level and a production loudness level for the input audio signal;(c) a filter generator/selector coupled to an output of the loudness estimator that provides an audio compensation filter based on the production loudness level and the audio playback loudness level,wherein the input signal is processed by the audio compensation filter to provide an output signal, andwherein when the loudness estimator provides at least one of a new audio playback loudness level or a new production loudness level for the input audio signal, the filter generator/selector provides a new audio compensation filter which is then used to process the input signal in order to provide the output signal.2. A system according to claim 1 , wherein the output signal is coupled to an audio output device that comprises at least one of a speaker or a headphone.3. A system according to claim 1 , wherein the production loudness level is fixed based on a setting provided by a user.4. A system according to claim 3 , wherein the setting provided by the user is a selected deviation from a default value that has been established based on an other input.5. A ...

Method and system for intelligently adjusting volume, and storage medium

A voice module of a television collects environmental volume Vr of an environment where a user is located; first current volume Vc of the television is obtained and compared with the environmental volume Vr, so as to determine whether the television volume needs to be adjusted; a desired volume value of the television requiring volume adjustment is calculated by means of a volume curve dynamic adjustment algorithm, and the television automatically adjusts its volume to the desired volume value.

AUDIO LOUDNESS CONTROL METHOD AND SYSTEM BASED ON SIGNAL ANALYSIS AND DEEP LEARNING

The present disclosure relates to a method and system for controlling loudness of an audio based on signal analysis and deep learning. The method includes analyzing an audio characteristic in a frame level based on signal analysis, analyzing the audio characteristic in the frame level based on learning, and controlling loudness of the audio in the frame level, by combining the analysis results. Accordingly, reliability of audio characteristic analysis can be enhanced and audio loudness can be optimally controlled. 1. A method for controlling loudness of an audio , the method comprising:first analyzing, at a first processor, an audio characteristic in a frame level based on signal analysis;second analyzing, at a second processor, the audio characteristic in the frame level based on learning; andcontrolling, at a controller, loudness of the audio in the frame level, by combining a result of the first analyzing and a result of the second analyzing.2. The method of claim 1 , wherein a type of the audio frame characteristic analyzed at the second analyzing is more diverse than a type of the audio frame characteristic analyzed at the first analyzing.3. The method of claim 2 , wherein the audio frame characteristic analyzed at the first analyzing comprises loudness and silence.4. The method of claim 3 , wherein the first analyzing compares the loudness of the audio frame with a threshold claim 3 , and divides the audio frame characteristic into a voice and silence.5. The method of claim 4 , wherein the threshold is variable according to audio data.6. The method of claim 3 , wherein the audio frame characteristic analyzed at the second analyzing comprises a foreground sound claim 3 , a background sound claim 3 , and silence.7. The method of claim 1 , wherein the controlling comprises:determining importance of frames forming the audio by combining the result of the first analyzing and the result of the second analyzing; andadjusting levels of loudness of the frames based on ...

NOISE ESTIMATION USING COHERENCE

The technology described herein can be embodied in a method for estimating a power spectral density of noise, the method including receiving an input signal representing audio captured using a microphone. The input signal includes a first portion that represents acoustic outputs from two or more audio sources, and a second portion that represents a noise component. The method also includes iteratively modifying a frequency domain representation of the input signal, such that the modified frequency domain representation represents a portion of the input signal in which effects due to the first portion are substantially reduced. The method further includes determining, from the modified frequency domain representation, an estimate of a power spectral density of the noise, and generating a control signal configured to adjust one or more gains of an acoustic transducer. The control signal is generated based on the estimate of the power spectral density of the noise. 1. A method for estimating a power spectral density of noise , the method comprising:receiving, at one or more processing devices, an input signal representing audio captured using a microphone, the input signal comprising a first portion that represents acoustic outputs from two or more audio sources, and a second portion that represents a noise component;iteratively modifying, by the one or more processing devices, a frequency domain representation of the input signal, such that the modified frequency domain representation represents a portion of the input signal in which effects due to the first portion are substantially reduced;determining, from the modified frequency domain representation, an estimate of a power spectral density of the noise; andgenerating a control signal configured to adjust one or more gains of an acoustic transducer corresponding one or more frequency ranges, the control signal being generated based on the estimate of the power spectral density of the noise.2. The method of claim 1 ...

Method and Apparatus for Adjusting Volume of User Terminal, and Terminal

A volume adjustment method and apparatus, and a terminal is presented. Perform analysis on the collected sound signal surrounding a user terminal, to obtain composition information, where the composition information includes sound types included in the sound signal and proportions of sounds of the various types, and the sound types include blank sound, human sound, and noise; determine a current scene mode of the user terminal according to the composition information; and adjust volume of the user terminal according to the determined scene mode, thereby significantly reducing occurrence of a case, caused by mistaken determining of the scenario, in which play volume adjustment does not conform to the scenario, and enhancing user experience. 1. A method for adjusting volume of a user terminal , comprising:acquiring microphone volume of the user terminal in real time after detection of a call signal connection;increasing an earpiece volume of the user terminal when the acquired microphone volume is greater than pre-stored reference volume; anddecreasing the earpiece volume of the user terminal when the acquired microphone volume is less than the pre-stored reference volume.2. The method of claim 1 , further comprising:periodically collecting a sound signal surrounding the user terminal;calculating volume of the sound signal;obtaining an average value according to the calculated volume of the sound signal and the pre-stored reference volume; andstoring the average value as a new reference volume.3. The method of claim 2 , wherein before calculating volume of the sound signal claim 2 , the method further comprises:performing analysis on the sound signal in order to obtain composition information of the sound signal; andcalculating the volume of the sound signal when the sound signal comprises only one person's sound.4. The method of claim 3 , wherein performing the analysis on the sound signal in order to obtain composition information of the sound signal comprises: ...

ELECTRONIC DEVICE AND METHOD FOR ANALYZING AND PLAYING SOUND SIGNAL

An electronic device and a corresponding method for analyzing and playing sound signals are provided. The electronic device includes a microphone, a processor, and a speaker. The microphone receives a sound and generates a sound signal according to the sound. The processor is coupled to the microphone for analyzing the sound signal to obtain an analysis parameter, determining a dynamic range parameter according to the analysis parameter, and adjusting the sound signal according to the dynamic range parameter. The speaker is coupled to the processor for playing the adjusted sound signal. 1. An electronic device , comprising:a microphone, receiving a sound and generating a sound signal according to the sound;a processor, coupled to the microphone, analyzing the sound signal to obtain an analysis parameter, determining a dynamic range parameter according to the analysis parameter, and adjusting the sound signal according to the dynamic range parameter; anda speaker, coupled to the processor, playing the adjusted sound signal.2. The electronic device according to claim 1 , wherein the analysis parameter comprises a histogram of a sound pressure level of the sound signal.3. The electronic device according to claim 1 , wherein the analysis parameter comprises a mean and a variance of a sound pressure level of the sound signal.4. The electronic device according to claim 1 , wherein the analysis parameter comprises a histogram of the sound signal in a frequency domain.5. The electronic device according to claim 1 , wherein the analysis parameter comprises a histogram of the sound signal in a time domain.6. The electronic device according to claim 1 , wherein the processor classifies the sound signal into one of a plurality of types according to the analysis parameter claim 1 , and the dynamic range parameter determined by the processor is a dynamic range parameter corresponding to the one type.7. The electronic device according to claim 1 , wherein the dynamic range ...

Alarm tone adjustment circuit and electronic device applying the circuit

An alarm tone adjustment circuit includes a sound collection unit with a function of collecting sound of a surrounding environment, an analog-digital (A/D) convertor, and a processing unit. The A/D convertor transforms the sound of a surrounding environment from the sound collection unit to a digital signal. The processing unit is coupled to the A/D convertor and an alarm tone unit, and determines a decibel range of the sound of surrounding environment according to a duty cycle of the digital signal, and adjusts a volume of a ring output from the alarm tone unit according to the decibel range of the sound of surrounding environment. Therefore, the alarm tone of the electronic device can be changed effectively according to the sound of surrounding environment.

USER-CONTROLLED BEAM STEERING IN MICROPHONE ARRAY

Various aspects include approaches for controlling acoustic beams. In some particular aspects, a system includes: a microphone array; a camera system for detecting a visual characteristic of a user; and a control system coupled with the microphone array and the camera system, the control system programmed to control the microphone array by: receiving a camera signal from the camera system; and focusing the microphone array in response to the camera signal indicating the visual characteristic of the user, prior to receiving an audio input from the user at the microphone array. 1. A system comprising:a microphone array;a camera system for detecting a visual characteristic of a user; and receiving a camera signal from the camera system; and', 'focusing the microphone array in response to the camera signal indicating the visual characteristic of the user, prior to receiving an audio input from the user at the microphone array., 'a control system coupled with the microphone array and the camera system, the control system programmed to control the microphone array by2. The system of claim 1 , wherein the control system is further configured to compare the visual characteristic of the user detected by the camera system with a stored visual characteristic library claim 1 , wherein focusing the microphone array is performed in response to the detected visual characteristic matching a visual characteristic in the stored visual characteristic library.3. The system of claim 2 , wherein the visual characteristic library comprises a thresholding mechanism for comparing the camera signal with pre-sorted video data or pre-sorted image data to determine whether the detected visual characteristic is linked with a focus direction for the microphone array.4. The system of claim 1 , wherein the visual characteristic comprises at least one of: a detected presence of the user claim 1 , a detected location of the user claim 1 , a detected identity of the user claim 1 , or a detected ...

Audio Settings Of Multiple Speakers in a Playback Device

Techniques described herein may involve dynamically adjusting the equalization of multiple speakers in a playback device. An example implementation involves a playback device emitting, via a first speaker, a first audio signal and emitting, via a second speaker, a second audio signal. The playback device may detect, via a microphone, an audio signal representing (i) at least a portion of the first audio signal and (ii) at least a portion of the second audio signal. The implementation may further involve the playback device determining one or more reflection characteristics based on the detected audio signal. The example implementation may also involve identifying a first audio setting and a second audio setting based on one or more reflection characteristics and applying (i) the first audio setting to playback of media content via the first speaker and (ii) the second audio setting to playback of media content via the second speaker. 120.-. (canceled)21. A playback device comprising:a housing;a transducer in the housing;a microphone in the housing;a processor; and recording, via the microphone, audio data, wherein the recording the audio data comprises recording (i) data indicative of sound emitted via the transducer and (ii) data indicative of a reflection of the emitted sound;', 'determining, in the recorded audio data, a feature of the operating environment based on the recorded audio data;', 'identifying a first audio setting corresponding to the determined feature of the operating environment; and', 'applying the first audio setting to playback of media content via the transducer., 'memory storing instructions that, when executed by the processor, cause the playback device to perform functions for dynamically adjusting audio played back via the playback device in an operating environment, the functions comprising22. The playback device of claim 21 , wherein the transducer is a first transducer claim 21 , the playback device further comprising: emitting, via the ...