Abstract: Provided are a loudspeaker assembly and an electronic device. The loudspeaker assembly includes: a housing having an internal cavity; a loudspeaker unit built in the internal cavity and including a loudspeaker diaphragm; and a separator provided in the internal cavity, the internal cavity being divided by the separator and the loudspeaker unit into a first cavity and a second cavity; where the housing has a through hole, the first cavity communicates with an exterior of the housing by the through hole, and the loudspeaker diaphragm is positioned in the first cavity; and the separator is provided with a balance hole in communication with the first cavity, and the second cavity is in communication with the first cavity through the balance hole.

Abstract: A system configured to improve audio processing by performing dereverberation and noise reduction during a communication session. The system may apply a two-channel dereverberation algorithm by calculating coherence-to-diffuse ratio (CDR) values and calculating dereverberation (DER) gain values based on the CDR values. While the device calculates the DER gain values prior to performing acoustic echo cancellation (AEC) processing, the device applies the DER gain values after performing residual echo suppression (RES) processing in order to avoid excessive attenuation of the local speech. To improve output speech quality, the device does not apply the DER gain values for nonreverberant signals, when a signal-to-noise ratio (SNR) value is too low, and/or when far-end talk (e.g., remote speech) is present. Dereverberation processing is further improved by using frequency dependent parameters to calculate the DER gain values and by adjusting other gain values when the DER gain values are applied.

Abstract: An apparatus includes a hybrid adaptive active noise control unit (HAANCU) configured to provide an anti-noise signal to an ear speaker from a reference noise signal of a reference microphone and an error signal of an error microphone, a decimator configured to decimate the reference noise signal and error signal, an adaptive hybrid ANC training unit (AHANCTU) including at least one noise cancellation filter and a filter configured to provide a feedback signal to the at least one noise cancellation, which trains parameters of the AHANCTU based on the decimated reference noise signal, the decimated error signal, and the feedback signal. The apparatus further includes a rate conversion unit configured to up-sample the parameters and update the HAANCU with the up-sampled parameters.

Abstract: A display apparatus includes a display panel configured to display an image by emitting light, a supporting member on a rear surface of the display panel, and a sound generator between the display panel and the supporting member. The sound generator includes a magnetic circuit unit and a voice coil unit on the rear surface of the display panel, and the magnetic circuit unit and the voice coil unit are configured to vibrate the display panel to generate sound.

Abstract: A rectangular microspeaker according to the present invention transfers vibration sound through the front surface or side surface of a diaphragm. In order to maximize an effective vibration area and to expand a volume, all parts including a frame are fabricated in a rectangular shape. The present invention is applied to a P type, an F type, and a composite type. The vibration sound generated on the bottom of the diaphragm as well as the vibration sound generated on the top of the diaphragm may be discharged through the side surface. A magnetic field part may be insert-molded separately or along with the frame.

Abstract: An electronic device includes a first surface which faces a first side and a second surface which faces a second side and including a first through hole which connects the first surface and the second surface to each other, a second housing component to the second side of the first housing component, and an acoustic component located on the first side of the first through hole. The first housing component and the second housing component as being combined with each other define a passage which communicates with outside from an end on the second side of the first through hole.

Abstract: The technology described in this document can be embodied in a method that includes receiving, at one or more processing devices of a headset that includes a sidetone generation circuit, an input signal representing ambient audio, and determining, by the one or more processing devices of the headset, that at least a portion of the input signal represents voice activity that satisfies a threshold condition. The method also includes, responsive to determining that the voice activity in the input signal satisfies the threshold condition, a control signal configured to cause the sidetone generation circuit to generate sidetone signals, and generating, by an acoustic transducer of the headset, an audio signal that represents, at least in part, the sidetone signals generated in accordance with the control signal.

Abstract: A wideband slot-loading loudspeaker includes a speaker driver configured to reproduce sound; an enclosure in which the speaker driver is disposed; a reflective member spaced apart from a front surface of the speaker driver, wherein a slot is formed in a space between the speaker driver and the reflective member; an acoustic discharge port provided at one end of the slot, the acoustic discharge port configured to discharge the sound reproduced by the speaker driver; at least one opening provided in the reflective member; and a sound resistance member disposed at the at least one opening.

Abstract: A portable electronic device including an enclosure having an enclosure wall that forms an interior chamber. A speaker module is positioned within the interior chamber and includes a speaker and a module wall forming a back volume chamber of the speaker. The back volume chamber includes an acoustic vent port formed through the module wall to acoustically couple the back volume chamber to the interior chamber. The device further including an electromechanical valve for regulating the acoustic coupling of the back volume chamber to the interior chamber. The electromechanical valve is operable to transition between an open configuration in which the acoustic vent port is open to the interior chamber and a closed configuration in which the acoustic vent port is closed off from the interior chamber.

Abstract: A user terminal apparatus is provided. The user terminal apparatus includes: a flexible display including: a main area; and a sub area comprising a curved portion and extending from the main area to a side portion of the user terminal apparatus; a housing configured to enclose the flexible display; a sound output hole provided between the flexible display and the housing and configured to output a sound from a call function; and a sound input hole configured to receive an input sound.

Abstract: A display apparatus includes a display panel configured to display an image by emitting light, a supporting member on a rear surface of the display panel, and a sound generator between the display panel and the supporting member. The sound generator includes a magnetic circuit unit and a voice coil unit on the rear surface of the display panel, and the magnetic circuit unit and the voice coil unit are configured to vibrate the display panel to generate sound.

Abstract: A loudspeaker module comprises a housing and a loudspeaker unit. The housing is provided with an inner cavity for receiving and fixing the loudspeaker unit. A vibrating diaphragm of the loudspeaker unit partitions the inner cavity into a front sound cavity and a rear sound cavity. The front sound cavity is in communication with at least two sound uttering holes, and the at least two sound uttering holes are distributed on different sides of the housing. The electronic product comprises the loudspeaker module. By means of the technical solution of the present disclosure, the sound uttering holes are provided individually on different sides of a housing of a loudspeaker module.

Abstract: An electronic device includes a first housing component including a first surface and a second surface and including a first through hole which connects the first surface and the second surface to each other, a second housing component removably attached to the first housing component, and an acoustic component located on a first side of the first through hole. The first housing component and the second housing component as being combined with each other define a passage which communicates with the outside. The first housing component includes a third surface. The second housing component includes a fourth surface. On one of the third surface and the fourth surface, a rib which protrudes to abut on the other is located. The rib is located to separate a region including the first through hole from other portions.

Abstract: Example implementations disclosed herein can be used to generate a local sound signal corresponding to utterances of a user and other sounds detected by a microphone array coupled to a communication device and to condition the local sound signals to separate the utterances of the user from the other sounds to generate a conditioned sound signal. The conditioned sound signals are evaluated to generate a local quality score for the conditioned sound signals, and when the local quality score of the conditioned sound signals is below a threshold associated with the communication device, a local feedback message indicating a local user position change can be generated. The local feedback message can include instructions for the user to move to another location to improve the quality of the condition sound signals.

Abstract: In some examples, devices for detecting and measures a ghost echo current in an audio signal are described. For instance, the device may be configured to sample a current of an audio signal, compare with an expected current, and determine an impulse response of the ghost echo current based on the sampled current and the expected current. The device may also be configured to reduce the ghost echo current using the impulse response.

Abstract: Providing real-time interaction between a first player and a second player to collaborate for a musical performance over a network, includes maintaining a reference data item generated using a musical instrument performed by a user; receiving first user input data generated using a first musical instrument, the received first user input data associated with the first musical instrument and the musical performance; receiving second user input data generated using a second musical instrument, the received second user input data associated with the second musical instrument and the musical performance; detecting a missing data packet within the received second user input data when generating real-time collaboration data for the musical performance; replacing the missing data packet using the reference data item to correct the second user input data; and generating real-time collaboration data for the musical performance based on the corrected second user input data.

Abstract: An electronic device has a speaker housing secured within the device housing. The speaker housing has a cavity with a speaker at one end and a port at the other configured to communicate through an aperture in the housing of the electronic device. A panel of acoustic mesh is integrally formed within the cavity of the housing and is disposed between the port and the speaker. In other embodiments flexible structures are integrally molded onto a plate or the acoustic device and used to secure and acoustically seal the acoustic device within the device housing.

Abstract: A full duplex voice communication method constituted of: estimating an acoustic echo within a near-end signal; cancelling the estimated acoustic echo; detecting whether, or not, a change has occurred in a near-end acoustic echo path, the received near-end signal represents speech and the received far-end signal represents silence, wherein, responsive to the results thereof, the method is further constituted of: alternately attenuating frequency components of the echo cancelled near-end signal by a first frequency domain attenuation value and by a second greater frequency domain attenuation value; alternately attenuating a first function of the frequency component attenuated echo cancelled near-end signal by a first switchable attenuation value and by a second greater switchable attenuation value; and alternately attenuating a second function of the received far-end signal by a third switchable attenuation value and by a fourth greater switchable attenuation value.

Abstract: According to an embodiment, a power spectrum iteration factor is determined according to a noisy speech and a background noise, and a moving average power spectrum of the speech is obtained according to the power spectrum iteration factor. A server is able to trace the noisy speech according to the power spectrum iteration factor.

Abstract: An electronic device has a speaker. The speaker has a main body and a cover. The main body has a speaker body and a tube. The tube is situated at a side of the speaker body. The cover is situated above the main body for covering the tube and part of the speaker body, such that the speaker has a covered portion and an uncovered portion. The cover has a hole. The sound produced by the covered portion is output via the tube along a first direction. The sound produced by the uncovered portion is output via the hole along a second direction.

Abstract: A method, and one or more non-transitory computer-readable media storing instructions, and a device for removal of unwanted components in an audio signal, the device comprising a processor, coupled to memory, configured to receive reference and processed inputs into memory where the processed input is a result of a reduction process of unwanted components of the audio signal, estimate envelope values for processed and reference inputs at a plurality of time and frequency instances, for each time and frequency instance: compute a first gain in relation to a ratio of the estimated envelope value of the processed input to the estimated envelope value of the reference input, apply a nonlinear process to said first gain to produce a second gain, compute an output gain as the ratio between second gain and first gain and, apply the output gain to processed input, thereby producing a filtered output with unwanted components suppressed.

Abstract: An electronic device may include a speaker, and audio circuitry coupled to the speaker. The audio circuitry may generate digitized samples of an audio waveform signal, and compare each digitized sample of the audio waveform signal to a threshold. The audio circuitry may when a given digitized sample is above the threshold, then apply a compression operation to the given digitized sample and successive digitized samples for a set time period, and when the given digitized sample is below the threshold and not within the set time period, then not apply the compression operation.

Abstract: An accessory device having multiple speakers and/or microphones to perform a number of audio functions, for use with mobile devices, are provided. The audio transducers (e.g., microphones and/or speakers) may be housed in one or more extendable and/or rotationally adjustable arms. To compensate for the unwanted signal feedback between the speakers and microphones, acoustic echo cancellation may be implemented to determine the proper distance and relative location between the speakers and microphones. Acoustic echo cancellation removes the echo from voice communications to improve the quality of the sound. The removal of the unwanted signals captured by the microphones may be accomplished by characterizing the audio signal paths from the speakers to the microphones (speaker-to-microphone path distance profile), including the distance and relative location between the speakers and microphones.

Abstract: An audio processing system includes an audio receiving module, a sound source separation module and a noise suppression module. The audio receiving module receives at least two audio signals. The sound source separation module receives a plurality of space features of the audio signals and obtains a main sound source signal separated from the audio signals based on the space features. The noise suppression module processes the main sound source signal based on an averaged amplitude value of noise in the main sound source signal so as to suppress noise in the main sound source signal. Each audio signal of the at least two audio signals includes signals from a plurality of sound sources.

Abstract: Acoustic echo cancellation is improved by receiving a speaker signal that is used to produce audio in a room, and receiving audio signals that capture audio from an array of microphones in the room, including an acoustic echo from the speakers. To cancel the acoustic echo, one adaptive filter is associated with a corresponding subspace in the room. Each of the audio signals is assigned to at least one of the adaptive filters, and a set of coefficients is iteratively determined for each of the adaptive filters. The coefficients for an adaptive filter are determined by selecting each of the audio signals assigned to that adaptive filter and adapting the filter to remove an acoustic echo from each of the selected audio signals. At each iteration, a different audio signal is selected from the audio signals assigned to the adaptive filter in order to determine the set of coefficients.

Abstract: An apparatus having a first circuit and a second circuit is disclosed. The first circuit is configured to generate a first intermediate signal by expanding a first input signal subjected to a quantization. The second circuit is configured to generate a second intermediate signal based on a second input signal. The second intermediate signal approximates an echo in the first input signal caused by the second input signal. The second circuit is also configured to generate a third intermediate signal by companding the second intermediate signal to compensate the quantization of the first input signal.

Abstract: A system and method to perform signal processing using a loudspeaker output are described. The system includes an enclosure configured to define a back cavity of the loudspeaker and components to obtain a representation of the loudspeaker acoustic output at the back cavity. The system also includes a processor to process the representation to perform the signal processing.

Abstract: Traditionally, echo cancellation has employed linear adaptive filters to cancel echoes in a two way communication system. The rate of adaptation is often dynamic and varies over time. Disclosed are novel rates of adaptation that perform well in the presence of background noise, during double talk and with echo path changes. Additionally, the echo or residual echo can further be suppressed with non-linear processing performed using joint frequency-time domain processing.

Abstract: An electronic device includes a case, a speaker, a supporting stand, and a waterproof breathable membrane. The case includes a first inner space and a through hole connected to the first inner space. The speaker is disposed in the first inner space, and produces sound waves outwards from the through hole. The supporting stand connects the case to cover the through hole. The supporting stand includes a sound hole and a second inner space connected to the sound hole and the through hole. The waterproof breathable membrane covers one of the sound hole and the through hole.

Abstract: An electronic device is provided. The electronic device includes a case, an acoustic boot, a first microphone and a second microphone. The case includes a first acoustic opening, a second acoustic opening. The acoustic boot comprises a first duct and a second duct, the first duct is connected to the first acoustic opening, and the second duct is connected to the second acoustic opening. The first microphone is connected to the first duct. The second microphone is connected to the second duct.

Abstract: An acoustic processing apparatus is provided. The apparatus includes a pre-processing component, a filter and a first signal processing component. The pre-processing component compensates a non-linearity of a reference signal to generate an input signal. The filter coupled to the pre-processing component, the filter executes filtering on the input signal to generate an output signal. The first signal processing component, coupled to the pre-processing component, the reference signal obtains a gain from the first signal processing component to generate a first signal, and the first signal processing component passes the gain to the pre-processing component.

Abstract: Features are disclosed for performing acoustic echo cancellation using random noise. The output may be used to perform speech recognition. Random noise may be introduced into a reference signal path and into a microphone signal path. The random noise introduced into the microphone signal path may be transformed based on an estimated echo path and then combined with microphone output. The random noise introduced into the reference signal path may be combined with a reference signal and then transformed. In some embodiments, the random noise in the reference signal path may be used in the absence of another reference signal, allowing the acoustic echo canceler to be continuously trained.

Abstract: An acoustic generator includes a vibrating body. The acoustic generator includes a frame member attached by a bonding material to an external peripheral portion of the vibrating body. The acoustic generator has an exciter provided on the vibrating body inside of the frame member. In this case, the acoustic generator further includes an extended portion of the bonding material provided to be extended on the vibrating body inside of the frame from the frame member.

Abstract: A method is disclosed for acoustic feedback attenuation at a telecommunications terminal. A speakerphone equipped with a loudspeaker and two microphones is featured. Signals from the two microphones are subjected to a calibration stage and then to a runtime stage. The purpose of the calibration stage is to match the microphones to each other by advantageously using both magnitude and phase equalization across the frequency spectrum of the microphones. During the runtime stage, the microphones monitor the ambient sounds received from sound sources, such as the speakerphone's users and the loudspeaker itself, during a conference call. The speakerphone applies the generated set of filter coefficients to the optimized microphone's signals. By combining the signal from the reference microphone with the filtered signal from the optimized microphone, the speakerphone is able to attenuate the sounds from the loudspeaker that would otherwise be transmitted back to other conference call participants.

Abstract: A speakerphone having transmit path with a microphone and a first amplifier having a gain determined by a controller and an input coupled to the microphone, a receive path with a second amplifier having a gain determined by the controller and a speaker coupled to an output of the second amplifier, and operable in a half-duplex mode using an adaptive echo canceller to at least partially remove from the transmit path acoustically-coupled signals from the receive path. The controller increases the gain of the first amplifier and decreases the gain of the second amplifier when a level of signals in the transmit path exceed a level of signals in the receive path, and decreases the gain of the first amplifier and increases the gain of the second amplifier when the level of signals in the receive path exceed the level of signals in the transmit path.

Abstract: A system for preventing noise transmission at the end of a telephone call includes a telephone having a handset and a base unit. The telephone has a proximity sensor configured to detect when the handset is in close proximity to the base unit prior to the handset contacting the base unit. A control element communicates with the proximity sensor and prevents transmission of sound received from the handset when the proximity sensor indicates that the handset is in close proximity to the base unit thereby preventing transmission of noise caused by the handset contacting the base unit.

Abstract: A facility for conducting a real-time conversation in which the selected one of a number of participants utilizes a silent mode is described. Remark spoken by participants other than the selected one are transformed into text and displayed for the selected participant. Remarks entered textually by the selected participant are transformed into speech and played audibly for participants other than the selected one.

Abstract: The present invention provides a method for exchanging information in a voice communication system, which includes a first communication device establishing a communication connection with a second communication device, and the first communication device exchanging digital data with the second communication device for achieving a specific function in a specific interval; wherein the specific interval is an interval that the first communication device does not exchange voice signals with the second communication device.

Abstract: A telephone includes a transmit channel and a receive channel, each including a bank of sub-band filters having a VAD coupled one to each sub-band filter. Each VAD measures the spectral energy in a sub-band, compares the spectral energy to a first threshold, and produces an output signal representative of whether or not the first threshold is exceeded. The voice activity detector also includes a threshold circuit for calculating a dynamically adjustable noise threshold based upon averaged measured spectral energy. A wide band or system VAD monitors echo canceling circuitry to detect voice activity and double talk.

Abstract: Systems and methods for managing the volume of multiple VoIP streams are disclosed. The system includes a VoIP server configured to receive an input audio stream from a first VoIP handset, create separate output audio streams from the input audio stream for transmission to second and third VoIP handsets, and to connect to a communications network. The system also includes a volume control table coupled to the VoIP server, the volume control table including records of volume adjustments made during prior conversations between the two or more VoIP handsets. The VoIP server is further configured create the separate output audio streams such that one or more of the output streams has a volume that is different than input audio stream based on the records.

Abstract: Embodiments of the present invention provide systems, methods, and computer-readable media for presenting a mute indicator to a caller when the caller is attempting to provide input to a call that has been muted. In particular, a determination is made that a mute function is engaged during a call. An area surrounding a calling system, such as a phone, is then monitored by a monitoring system to determine whether a caller is attempting to provide input to the call. A determination is then made that a caller is attempting to provide input to the call. The determination may be based on a caller providing a threshold volume of voice input within a threshold area of proximity near the calling system. A mute indicator is then presented to the caller to notify the caller that the call has a mute function engaged.

Abstract: A method and apparatus is disclosed that controls voice call quality by networks and maintains the optimal voice call quality in a portable terminal. The voice call quality control method includes: identifying a network, connected to the portable terminal when the portable terminal enters a voice call mode, and a value for applying voice call quality of the network; changing a value for setting voice call quality, performed in the network, by the value for applying voice call quality; and controlling voice call quality of a voice call mode in the network by the altered value for setting voice call quality.

Abstract: A voice band extender includes a component separator for receiving a band-limited speech signal over a telecommunications line to separate the speech signal into a noise-suppressed signal and an extracted-noise signal, a noise-suppressed signal component extender for adding a signal having its frequency band higher than that of the noise-suppressed signal to thereby produce an extended noise-suppressed signal, an extracted-noise signal component extender for producing an extended extracted-noise signal, a signal intensity adjuster for adjusting signal intensity of either or both of the extended noise-suppressed signal and the extended extracted-noise signal, and a synthesizer for combining the extended noise-suppressed signal and the extended extracted-noise signal obtained by the intensity adjustment. The respective frequency bands of the extracted-noise signal and the noise-suppressed signal are thus extended.

Abstract: The invention relates to a method for detecting the status of a telephone receiver (H) for a communication terminal (EG), wherein the telephone receiver (H) is equipped with a microphone (M) and with a loudspeaker (LS). The loudspeaker (LS) of the telephone receiver (H) is used to output a defined tone signal, and this tone signal is simultaneously picked up again by means of the microphone (M) of the telephone receiver (H). The tone signal picked up is analysed, with at least one value for at least one parameter of the tone signal picked up being ascertained, and the at least one value is used to determine the status of the telephone receiver (H).

Abstract: The invention provides an apparatus capable of automatic volume adjustment. In one embodiment, the apparatus comprises a speaker, an array microphone located in the vicinity of the speaker, a beamforming module, a signal-to-noise ratio calculation module, and a volume adjustment module. The speaker first broadcasts a first audio signal. The array microphone converts a sound into a plurality of second audio signals. The beamforming module derives a speaker sound signal and an ambient noise signal from the second audio signals, wherein the speaker sound signal mainly comprises speaker sound components generated by the speaker and the ambient noise signal mainly comprises noises other than the speaker sound components. The signal-to-noise ratio calculation module calculates a ratio of a first power of the speaker sound signal to a second power of the ambient noise signal to obtain a signal-to-noise ratio.

Abstract: A degree of seal of an ear about a speaker port may be estimated by detecting touch contact between the ear and at least one touch sensor in fixed relation to the speaker port. The degree of seal is estimated based on the detected touch contact. Based upon the estimated degree of seal, the acoustic output of the speaker may be adjusted. The adjustment may compensate for perceived changes to the quality of the acoustic output resulting from the degree of seal. The at least one touch sensor may be a plurality of touch sensors spaced around the speaker port. Each sensor may have a truncated wedge shape, with a narrow end closest to the speaker port. Upon receipt of user input indicative of a high degree of ear seal, a sample of the sensor(s) may be taken and stored for using during future estimation of the degree of seal.

Abstract: In various embodiments, a speakerphone may comprise multiple (e.g., 16) microphones placed in a circular array around a central speaker. The microphones may be mounted vertically in the speakerphone with their respective diaphragms substantially parallel to the top surface of the speakerphone. The centrally mounted speaker may be coupled to a stiff internal speaker enclosure. The speaker enclosure may be made of a stiff, heavy material (e.g., a dense plastic) to prevent the speaker vibrations from excessively vibrating the speakerphone enclosure (which may affect the microphones). The speaker enclosure may also include a raised rim and ridges to increase its stiffness.

Abstract: The present invention is a system and method that improves upon voice activity detection by packetizing actual noise signals, typically background noise. In accordance with the present invention an access network receives an input voice signal (including noise) and converts the input voice signal into a packetized voice signal. The packetized voice signal is transmitted via a network to an egress network. The egress network receives the packetized voice signal, converts the packetized voice signal into an output voice signal, and outputs the output voice signal. The egress network also extracts and stores noise packets from the received packetized voice signal and converts the packetized noise signal into an output noise signal. When the access network ceases to receive the input voice signal while the call is still ongoing, the access network instructs the egress network to continually output the output noise signal.

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

Abstract: In a speakerphone device identification of signals (i.e., voice input or speaker output) in a process for reducing acoustic feedback, in a communication device, is accomplished by adding a signature noise (i.e., an identification mark) to output signals radiated by the speaker to enable these signals to be separated from speech input to the microphone. Having identified the signal (i.e., speech output) likely to cause a “singing” phenomenon, appropriate insertion loss to reduce the feedback may be added to the appropriate speech path within the communication device, to reduce a probability of singing.