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136 results about "Monaural" patented technology

Monaural or monophonic sound reproduction (often shortened to mono) is sound intended to be heard as if it were emanating from one position. This contrasts with stereophonic sound or stereo, which uses two separate audio channels to reproduce sound from two microphones on the right and left side, which is reproduced with two separate loudspeakers to give a sense of the direction of sound sources. In mono, only one loudspeaker is necessary, but, when played through multiple loudspeakers or headphones, identical signals are fed to each speaker, resulting in the perception of one-channel sound "imaging" in one sonic space between the speakers (provided that the speakers are set up in a proper symmetrical critical-listening placement). Monaural recordings, like stereo ones, typically use multiple microphones fed into multiple channels on a recording console, but each channel is "panned" to the center. In the final stage, the various center-panned signal paths are usually mixed down to two identical tracks, which, because they are identical, are perceived upon playback as representing a single unified signal at a single place in the soundstage. In some cases, multitrack sources are mixed to a one-track tape, thus becoming one signal. In the mastering stage, particularly in the days of mono records, the one- or two-track mono master tape was then transferred to a one-track lathe intended to be used in the pressing of a monophonic record. Today, however, monaural recordings are usually mastered to be played on stereo and multi-track formats, yet retain their center-panned mono soundstage characteristics.

Group beam selection and beam merging

A system that performs beam selection and beam merging using beam-specific signal quality metrics corresponding to a minimum noise floor. For example, a device may track a minimum noise floor for each beam, determine a highest minimum noise floor across the beams, and determine a noise floor ratio between the beam-specific minimum noise floor and the highest minimum noise floor. Using a combination of the noise floor ratio and signal-to-noise ratio (SNR) values, the device may perform beam selection by prioritizing low background noise as well as high SNR to select a pre-defined beam group. In addition, the device may use the noise floor ratio to perform beam merging and generate single-channel output audio data using the selected beam group. For example, the device may scale the beams based on a combination of the SNR value and the noise floor ratio.
Owner:AMAZON TECH INC

Smart audio power management for battery powered mobile speakers

In a wireless stereo system including first and second battery-powered speakers, if low power is sensed in the second speaker, the user may be given the option of shifting the first speaker to mono mode, or maintain the first speaker in stereo mode, playing its assigned channel, or alter a left / right stereo balance between the speakers. Speaker setting may also change depending on whether a speaker is on or off a charging dock.
Owner:SONY INTERACTIVE ENTERTAINMENT LLC

Adaptive inter-channel time difference estimation

A method to estimate an inter-channel time difference (ITD) in an encoder using a discontinuous transmission (DTX) is disclosed. The method includes receiving a time domain audio input including audio input signals and processing the audio input signals in frames to produce a mono mixdown signal and one or more stereo parameters. The method further includes encoding the mono mixdown signal on a frame-by-frame basis by: encoding of active content of the mono mixdown signal at a first bit rate until a pause period is detected; estimating ITD parameters during the encoding of active content; switching the encoding from the active encoding content to inactive encoding to encode background noise at a second bit rate during the pause period; and estimating ITD parameters during the pause period. The method further includes encoding the ITD estimated parameters and other stereo parameters periodically during the pause period.
Owner:TELEFONAKTIEBOLAGET LM ERICSSON (PUBL)

Safety aid

The present application is to provide a safety assistance device capable of appropriately and easily informing a driver of the danger of collision by outputting stereo sound based on the state of the driver. To solve the above problem, the safety assistance device (1) includes: an object detection camera (21) and a radar device (22) as detection units; and a control unit (10) that controls a dual-purpose speaker (31), a dual-purpose speaker (32), and a mono speaker (4) as sound output units based on detection information from a monitoring camera (5) as a monitoring unit. The control unit (10) selects outputting stereo sound from the dual-purpose speaker (31) and the dual-purpose speaker (32), or outputting sound from the dual-purpose speaker (31), the dual-purpose speaker (32), and the mono speaker (4), according to the position of the ear of the driver (101) detected by the monitoring camera (5).
Owner:HONDA MOTOR CO LTD

Optimization process for reducing stereo audio signal channels

The invention relates to a method for downmixing a stereo signal to obtain a monaural signal, the method comprising selecting (305), for a stereo signal frame, between: a first downmixing mode (T1), comprising filtering with phase correction in each channel of the stereo signal at an angle defined by half of a phase difference determined (303, 304) between two channels of the stereo signal, and-a second downmix mode (T2), comprising filtering with phase correction in one channel of the stereo signal according to an angle defined by the phase difference determined (303, 304) between two channels of the stereo signal, the selection of the first downmix mode or the second downmix mode depending on the value of the phase indicator determined (302) per frame. The invention also relates to a downmixing device for implementing the method.
Owner:ORANGE SA

Sound signal refinement method, sound signal decode method, apparatus thereof, program, and storage medium

There is provided a technology that improves, in a case where there is a sound signal obtained from a different code that is different from a code from which a decoded sound signal is obtained and that is derived from the same sound signal, the decoded sound signal by using the sound signal obtained from the different code. A signal (hereinafter, referred to as an upmixed common signal) obtained by upmixing a decoded sound common signal obtained by downmixing a decoded sound signal of each channel is subjected to signal purification using a signal (hereinafter, referred to as an upmixed monaural decoded sound signal) obtained by upmixing a monaural decoded sound signal to thereby generate a purified upmixed signal, and in each channel, the upmixed common signal is subtracted from the decoded sound signal and the purified upmixed signal is added thereto, to thereby generate a purified decoded sound signal.
Owner:NT T INC

Multi-channel call recording recognition method and device based on single-channel artificial intelligence model

The invention discloses a multichannel call recording recognition method based on a monaural artificial intelligence model. The method comprises the following steps: inputting multichannel call audio data; performing voice activity detection on each sound channel to obtain a voice segment in each sound channel and a corresponding original timestamp during detection; sorting the original timestamps based on the sequence of the original timestamps, constructing a monaural audio, and recording and synthesizing a timestamp; inputting the constructed monaural audio into a pre-trained monaural speech recognition model to generate a recognition text sequence and record an output timestamp; based on an overlapping interval between the synthetic timestamp and the output timestamp, backtracking and matching to the original timestamp; and constructing a triple containing the speaker, the timestamp and the recognition text according to a backtracking matching result. The invention further provides a multichannel call recording recognition device. According to the method provided by the invention, the speaker can be identified and the call logic sequence can be reconstructed while the semantic accuracy is kept.
Owner:ZHEJIANG UNIV +1

Apparatus, method and computer program for encoding, decoding, scene processing and other procedures related to DirAC based spatial audio coding using low-order, mid-order and high-order components generators

An apparatus for generating a sound field description using an input signal having a mono-signal or a multi-channel signal includes: an input signal analyzer for analyzing the input signal to derive direction data and diffuseness data; a low-order components generator for generating a low-order sound field description from the input signal up to a predetermined order and mode; a mid-order components generator for generating a mid-order sound field description above the predetermined order or at the predetermined order and above the predetermined mode and below or at a high order, wherein the mid-order sound field description comprises a direct contribution and a diffuse contribution; and a high-order components generator for generating a high-order sound field description comprising a sound field component above the high order using a synthesis of at least one direct portion, wherein the high-order sound field description comprises a direct contribution only.
Owner:FRAUNHOFER GESELLSCHAFT ZUR FORDERUNG DER ANGEWANDTEN FORSCHUNG EV

Method for generating a spatial voice signal and a device thereof, method for receiving a spatial voice signal and a device thereof

A method for generating a spatial voice signal performed by a voice pickup device, the method comprises: obtaining, by a plurality of microphones associated with the voice pickup device, a plurality of audio signals; determining a number of voice sources based on the plurality of audio signals; extracting, for each of the voice sources, a voice signal based on the plurality of audio signals, so as to generate a plurality of voice signals corresponding to the number of voice sources; mapping the plurality of voice signals into a first synthesis signal corresponding to a first channel of the spatial voice signal and a second synthesis signal corresponding to a second channel of the spatial voice signal, wherein for each of the voice sources, a time delay exists between the first synthesis signal and the second synthesis signal, and the time delay is associated with a relative location between the voice source and the plurality of microphones; transforming the first synthesis signal and the second synthesis signal into a mono channel signal associated with the spatial voice signal.
Owner:HARMAN INT IND INC +1

Sound adjusting device

PendingCN120825656AMicrophonesSignal processingEar trumpetSound sources
The invention provides a sound adjusting device which comprises a background sound source, an external sound source receiver, a frequency adjuster, a sound mixer and an output port. The background sound source is used for providing a background audio. The background audio is monaural audio. The external sound source receiver is used for receiving an external audio frequency from the outside. The frequency adjuster is used for receiving the background audio and adjusting the background audio into a left ear background audio and a right ear background audio. A first frequency of the left ear background audio and a second frequency of the right ear background audio have a frequency difference. The audio mixer is used for mixing the external audio and the left-ear background audio into a left-ear audio and mixing the external audio and the right-ear background audio into a right-ear audio. The output port is used for outputting the left-ear audio frequency to a left-ear horn and outputting the right-ear audio frequency to a right-ear horn.
Owner:COTRON CORPORATION

Audio and video acquisition equipment and audio processing chip

The embodiment of the invention provides an audio and video acquisition device and an audio processing chip, the audio and video acquisition device supports a local microphone and a wireless microphone at the same time, and one path of audio data is acquired through the local microphone built in the audio and video acquisition device. And meanwhile, the audio processing unit integrated in the audio and video acquisition equipment is wirelessly connected with the wireless microphone worn on the body of the user, another path of audio data acquired by the wireless microphone is acquired, and the audio processing unit synthesizes the two paths of audio data into two-channel stereo sound with real space sense. Meanwhile, a built-in image sensor and a built-in video processing unit can be used for collecting and processing videos. The audio and video acquisition equipment breaks through the limitation of the pickup distance of traditional equipment, solves the problem that a monaural track lacks stereoscopic impression, and realizes audio and video synchronous acquisition and high-quality output.
Owner:SHENZHEN HOLLYLAND TECH CO LTD

Audio signal coding method, audio signal coding device, program, and recording medium

The present invention provides embedded coding of a monaural audio signal and audio signals of multiple channels in which the algorithm delay in stereo coding / decoding is not greater than the algorithm delay in monaural coding / decoding. A coding device (100) codes the audio signals of multiple channels in units of frames. A stereo coding unit (110) obtains and outputs a stereo code representing a feature parameter which indicates a feature regarding inter-channel difference between the audio signals of multiple channels. A down-mixing unit (150) obtains, as a down-mixed signal, a signal in which the audio signals of multiple channels are mixed. A monaural coding unit (120) obtains a monaural code by coding the down-mixed signal using a coding scheme which includes a process of multiplying overlapped parts between frames by a window function, and outputs the monaural code. An additional coding unit (130) obtains an additional code by coding a signal in a overlapped section between a current frame and the immediately subsequent frame of the down-mixed signal, and outputs the additional code.
Owner:NIPPON TELEGRAPH & TELEPHONE CORP

Hand-operated earpiece for mobile phones

In one aspect, an earphone for use with a mobile phone or other device may include a low-profile, ergonomic head with a speaker and a flat front surface. The front surface may be be bounded by three sides and a curved bottom edge that is configured for placement on top of an incisura of an ear to hang on the incisura, thereby improving user comfort. Additionally, a trigger related to the earphone may be used so that, responsive to identifying the trigger, the mobile phone may facilitate a call and present mono audio from the call via the at least one speaker on the earphone. In certain examples, the trigger may include a threshold period of time elapsing from detachment of the earphone from the mobile device during call reception.
Owner:SPRAGUE GARY

UWB-based stereo synthesis method, system, device and storage medium

This invention provides a stereo synthesis method, system, device, and storage medium based on UWB positioning. The stereo synthesis method involves installing UWB positioning tags on a moving sound source and setting up a UWB wireless positioning system around the sound source. The system includes three or more anchor points, which communicate with the tags, calculating the distance between the tags and each anchor point in real time and uploading the data to a host computer. Simultaneously, the mono audio signal collected by a microphone moving with the sound source is uploaded to the host computer. The host computer uses image reproduction technology to fuse the mono sound collected by the microphone and the tag's location information using HRTF to generate a stereo audio signal. This stereo synthesis method enables stereo pickup and synthesis even when only mono audio is acquired from a moving sound source.
Owner:SHENZHEN LONGXINWEI SEMICON TECH CO LTD

Wearable device and method for acquiring voice signal based on wearing state

An earbud includes a speaker, a communication circuit, multiple microphones, an acceleration sensor, and at least one processor. The earbud is configured to receive, from an external electronic device, a first data signal for identifying at least one filter among multiple filters for audio data output from the multiple microphones. The earbud is configured to identify at least one microphone for mono recording among the multiple microphones based on reception of the first data signal. The earbud is configured to obtain voice data by applying a filter for the mono recording among the multiple filters to a voice signal received from the at least one microphone. The earbud is configured to transmit a second data signal indicating the voice data to the external electronic device.
Owner:SAMSUNG ELECTRONICS CO LTD

Wireless earphone

ActiveCN309639111SSound sourcesMonaural
1. The name of the design product: wireless earphone. 2. The use of the design product: used for wireless connection of sound source equipment and playing audio. 3. The design points of the design product: in shape. 4. The picture or photo that best shows the design points: component 1 perspective view 1. 5. Other circumstances that need to be explained: the design product is sold and used in sets, single use is monaural, and combined use is stereo.
Owner:REALME MOBILE TELECOMM SHENZHEN CO LTD

Multichannel audio coding apparatus and method of operation therefor

PCT designated stageWO2026027356A1Speech analysisBiological modelsMonauralData signal
An encoder comprises a transient extractor (103) generating a residual multichannel audio signal by extracting single channel transient signal components. A combiner circuit (105) generates a set of mono transient signal components by combining single channel transient signal components. A downmix circuit (107) generates downmix audio data by downmixing the residual multichannel audio signal and the mono channel transient signal components, and generates signal upmix parameters for upmixing the residual multichannel audio signal and transient upmix parameters for upmixing the mono channel transient signal components. An output circuit (109) generates an encoded audio data signal comprising the generated data. A decoder has upmixers that upmix the received downmixed residual signal and the mono transient signal components using the received parameters, including generating a plurality of transient signal components for one received mono transient signal component.
Owner:KONINKLIJKE PHILIPS NV

Audio encoding apparatus and method, and audio decoding apparatus and method

An audio signal processing apparatus is configured to: transform a first audio signal includes n channels to generate a first audio data in a frequency domain, generate a frequency feature signal for each channel from the first audio data in the frequency domain, based on a first deep neural network (DNN), generate a second audio signal includes m channels from the first audio signal, based on a second DNN, and generate an output audio signal by encoding the second audio signal and the frequency feature signal. The first audio signal is a high order ambisonic signal includes a zeroth order signal and a plurality of first order signals. The second audio signal includes a mono signal or a stereo signal. m is smaller than n.
Owner:SAMSUNG ELECTRONICS CO LTD

Sound signal refinement method, sound signal decode method, apparatus thereof, program, and storage medium

There is provided a technology that improves, in a case where there is a sound signal obtained from a different code that is different from a code from which a decoded sound signal is obtained and that is derived from the same sound signal, the decoded sound signal by using the sound signal obtained from the different code. A signal (hereinafter, referred to as a decoded sound common signal) obtained by downmixing a decoded sound signal of each channel is subjected to signal purification using a monaural decoded sound signal to thereby generate a purified common signal, and in each channel, a signal obtained by upmixing the decoded sound common signal is subtracted from the decoded sound signal and a signal obtained by upmixing the purified common signal is added thereto, to thereby generate a purified decoded sound signal.
Owner:NT T INC

Multi-scene stage sound effect management and control system and method based on intelligent frequency division

The invention discloses a multi-scene stage sound effect management and control system and method based on intelligent frequency division, and relates to the technical field of stage sound systems, the system comprises an audio input module, an intelligent frequency division module, a dynamic sound effect enhancement module, a multi-channel power amplifier module and a user interface module, the audio input module is used for receiving human voice and accompaniment signals; the intelligent frequency division module comprises a DSP processing unit and a scene identification unit, the DSP processing unit divides a signal into four frequency bands through an FFT algorithm, and the scene identification unit automatically matches a frequency division strategy according to a performance type; the dynamic sound effect enhancement module processes signals of all frequency bands; the multi-channel power amplifier module comprises a monaural subwoofer power amplifier unit and a dual-channel mid-high pitch power amplifier unit, and a power amplifier output end is connected with a corresponding loudspeaker unit; and the user interface module provides a scene preset button and a parameter real-time display screen unit, so that sound effect self-adaption of multiple performance types is realized, the ultrahigh audio frequency segment reduction capability is improved, and meanwhile, the human voice and accompaniment signal separation degree is improved.
Owner:HANGZHOU CHINASOFT INTELLIGENT OPTOELECTRONICS CO LTD

Audio channel metadata generation method and device, equipment and storage medium

The invention relates to an audio channel metadata generation method and device, equipment and a storage medium, and belongs to the technical field of audio processing, and the method comprises the steps: determining whether a received audio file belongs to a monaural; if yes, generating an audio channel format element corresponding to the audio file; dividing the audio channel into a plurality of audio blocks according to the audio channel format elements; identifying universal attributes and type attributes of the audio blocks; metadata corresponding to the universal attributes and metadata corresponding to the type attributes are generated. According to the method, whether the loaded audio file is the monaural file or not is judged, if the loaded audio file is the monaural file, the elements of the audio channel required for transmitting the audio file are generated, and the metadata of the general attribute and the type attribute are generated in the form of the audio block, so that the metadata related to the audio channel is established; therefore, the rendering and playing requirements of different types of audio files are met.
Owner:SINE MICRO (BEIJING) ELECTRONIC TECH CO LTD

Cross-platform compatible self-adaptive single / double-channel identification method and system for IC (Integrated Circuit) of Yiai power amplifier

The invention relates to the technical field of audio power amplifier driving, and provides a method and a system for IC cross-platform compatibility and self-adaptive single-channel and double-channel identification of an Eixian power amplifier. I2C and a reset interface are unified through a hardware abstraction layer, a bitmap channel mark is formed through address detection and state verification, a monaural parameter set or a left and right channel parameter set is selected for mode judgment, each IC is independently started, and an audio channel is configured; and fault-tolerant output in an operation period monitors abnormity and is degraded from stereophonic sound to monophonic sound to keep sounding. According to the scheme, one-time development of multi-platform multiplexing is realized, single-machine firmware is compatible with single and double channels, the transplantation and mass production cost is reduced, and the reliability and the user experience are improved.
Owner:SHENZHEN YITIAN TECH CO LTD

Method of processing audio for playback of immersive audio

A method (200) of processing audio in an immersive audio format comprising at least one height audio channel, comprising: obtaining (250) two height audio signals from at least a portion of the at least one height audio channel: modifying (270) a relative phase between the two height audio signals in frequency bands in which phase differences are predominantly out of phase to obtain two phase modified height audio signals: and playing back (290) the processed audio comprising the two phase modified height audio signals with at least two audio loudspeakers. The phase differences occur as a result of having monaural signals emanated from two audio loudspeakers at one or more listening positions symmetrically off-center with respect to the at least two loudspeakers, laterally spaced with respect to each of said one or more listening positions. The method allows perception of sound height / elevation without using overhead speakers.
Owner:DOLBY LABORATORIES LICENSING CORP

A sound detection method

The application discloses a sound detection method and relates to the technical field of measuring electric variables; the method comprises the following steps: S1, audio signals are segmented by using a sliding wake-up time window, and multi-channel root mean square values are calculated; when the multi-channel root mean square values are greater than a preset wake-up threshold, it is determined that wake-up is triggered; S2, the audio signals are segmented by using a sliding interaction time window to obtain audio segments of each channel; each audio segment is further divided to obtain multiple equal-length sub-segments; single-channel root mean square values are calculated; when the single-channel root mean square values are greater than an activation threshold, it is determined that valid signals exist in the sub-segments; the wake-up detection in step S1 and the formation of a lightweight signal processing method in step S2 take into account real-time performance and deployment cost, and the efficiency of recognizing touch sequences based on a microphone array is improved; in step S2, the multiple equal-length sub-segments are further divided, the single-channel root mean square values of the sub-segments are compared with the activation threshold, and the recognition accuracy is improved.
Owner:SUZHOU ACOUSTIC IND TECH RES INST CO LTD

Audio channel metadata generation method and device, equipment and storage medium

The invention relates to an audio channel metadata generation method and device, equipment and a storage medium, and belongs to the technical field of audio processing, and the method comprises the steps: determining whether a received audio file belongs to a monaural; if yes, generating an audio channel format element corresponding to the audio file; dividing the audio channel into a plurality of audio blocks according to the audio channel format elements; identifying universal attributes and type attributes of the audio blocks; metadata corresponding to the universal attributes and metadata corresponding to the type attributes are generated. According to the method, whether the loaded audio file is the monaural file or not is judged, if the loaded audio file is the monaural file, the elements of the audio channel required for transmitting the audio file are generated, and the metadata of the general attribute and the type attribute are generated in the form of the audio block, so that the metadata related to the audio channel is established; therefore, the rendering and playing requirements of different types of audio files are met.
Owner:SINE MICRO (BEIJING) ELECTRONIC TECH CO LTD

Lightweight binaural audio synthesis method based on implicit neural network

The invention discloses a lightweight binaural audio synthesis method based on an implicit neural network, which efficiently generates high-fidelity binaural audio from monaural audio and sound source pose information through a two-stage framework. An initial binaural signal containing a main time clue is roughly generated by using a neural time domain distortion module, a frequency spectrum of the signal is finely corrected by using an implicit binaural correction module, and a frequency spectrum clue is introduced. The implicit correction module models a complex frequency spectrum correction process into a continuous mapping function which is represented by a small multilayer perceptron and is from space-time-frequency coordinates to complex correction values. According to the method, the parameter quantity and the calculation requirement of the model are reduced, so that the model can be efficiently deployed on virtual reality, augmented reality and other edge devices with limited calculation resources, the problem of efficiency and quality balancing in real-time high-fidelity synthesis is solved, the method is suitable for virtual reality (VR), augmented reality (AR) and other resource-limited scenes needing real-time 3D audio rendering, and the real-time 3D audio rendering efficiency is improved. Such as mobile games, online conferences and smart headphones.
Owner:EAST CHINA NORMAL UNIV

300W gallium nitride belt DSP amplification circuit

A 300W gallium nitride belt DSP amplification circuit comprises an input selection circuit, an ADC circuit and a power amplification circuit. The input selection circuit comprises an analog switch chip U02 and an analog switch chip U03, and a line and a DANTE signal convert control voltage through a twelfth pin and a thirteenth pin of the analog switch chip U02 and the analog switch chip U03 to form a truth table to meet switching; the ADC circuit comprises an audio converter chip U2, and the switched analog audio signal is converted into a digital signal through the audio converter chip U2; the power amplifier circuit comprises a monaural gallium nitride D-type power amplifier chip U3, a digital signal is subjected to power amplification after I2S communication through a third pin, a fourth pin and a fifth pin of the monaural gallium nitride D-type power amplifier chip U3, and a ninth pin and a tenth pin of the monaural gallium nitride D-type power amplifier chip U3 are respectively communicated with a PC I2C for DSP tuning.
Owner:FIRST AUDIO MFG CO LTD

Vehicle sound production control method, device and equipment, vehicle and medium

The embodiment of the invention provides a vehicle sound production control method, device and equipment, a vehicle and a medium, and is applied to the technical field of vehicles, the method comprises the steps that at least two types of audio signals are transmitted to at least two motor modules of the vehicle respectively, and the audio signals are used for making the motor modules emit sounds corresponding to the audio signals; controlling the at least two motor modules to start playing simultaneously in response to a playing instruction; wherein the audio signal is a monaural signal. According to the embodiment of the invention, the at least two motor modules in the vehicle are used for sounding, so that the audio signal can be transmitted to the outside of the vehicle, the function of the motor in the vehicle is expanded, additional sound equipment facing the outside of the vehicle does not need to be arranged, and the cost is reduced.
Owner:BYD CO LTD