WO2021170061A1 - 无线扩音系统及终端 - Google Patents

无线扩音系统及终端 Download PDF

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Publication number
WO2021170061A1
WO2021170061A1 PCT/CN2021/078028 CN2021078028W WO2021170061A1 WO 2021170061 A1 WO2021170061 A1 WO 2021170061A1 CN 2021078028 W CN2021078028 W CN 2021078028W WO 2021170061 A1 WO2021170061 A1 WO 2021170061A1
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WIPO (PCT)
Prior art keywords
voice signal
digital
signal
audio
microphone
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PCT/CN2021/078028
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English (en)
French (fr)
Inventor
冯建婷
杨枭
盛行
洪润琦
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华为技术有限公司
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Priority to EP21760045.1A priority Critical patent/EP4096202A4/en
Publication of WO2021170061A1 publication Critical patent/WO2021170061A1/zh

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00Circuits for transducers, loudspeakers or microphones
    • H04R3/02Circuits for transducers, loudspeakers or microphones for preventing acoustic reaction, i.e. acoustic oscillatory feedback
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/02Constructional features of telephone sets
    • H04M1/0202Portable telephone sets, e.g. cordless phones, mobile phones or bar type handsets
    • H04M1/026Details of the structure or mounting of specific components
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10KSOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1785Methods, e.g. algorithms; Devices
    • G10K11/17853Methods, e.g. algorithms; Devices of the filter
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10KSOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1785Methods, e.g. algorithms; Devices
    • G10K11/17853Methods, e.g. algorithms; Devices of the filter
    • G10K11/17854Methods, e.g. algorithms; Devices of the filter the filter being an adaptive filter
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10KSOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K15/00Acoustics not otherwise provided for
    • G10K15/02Synthesis of acoustic waves
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/02Constructional features of telephone sets
    • H04M1/03Constructional features of telephone transmitters or receivers, e.g. telephone hand-sets
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/72Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/724User interfaces specially adapted for cordless or mobile telephones
    • H04M1/72403User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality
    • H04M1/72409User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality by interfacing with external accessories
    • H04M1/72412User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality by interfacing with external accessories using two-way short-range wireless interfaces
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R27/00Public address systems
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/60Substation equipment, e.g. for use by subscribers including speech amplifiers
    • H04M1/6008Substation equipment, e.g. for use by subscribers including speech amplifiers in the transmitter circuit
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/72Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/724User interfaces specially adapted for cordless or mobile telephones
    • H04M1/72403User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality
    • H04M1/7243User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality with interactive means for internal management of messages
    • H04M1/72433User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality with interactive means for internal management of messages for voice messaging, e.g. dictaphones
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2227/00Details of public address [PA] systems covered by H04R27/00 but not provided for in any of its subgroups
    • H04R2227/001Adaptation of signal processing in PA systems in dependence of presence of noise
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2227/00Details of public address [PA] systems covered by H04R27/00 but not provided for in any of its subgroups
    • H04R2227/009Signal processing in [PA] systems to enhance the speech intelligibility
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2420/00Details of connection covered by H04R, not provided for in its groups
    • H04R2420/07Applications of wireless loudspeakers or wireless microphones

Definitions

  • the embodiments of the present application relate to the field of communications, and in particular to a wireless loudspeaker system and terminal.
  • the prior art loudspeaker solution usually requires users to carry loudspeaker equipment, and the loudspeaker equipment includes microphones and audio equipment, as shown in Figure 1.
  • the loudspeaker equipment includes microphones and audio equipment, as shown in Figure 1.
  • the present application provides a wireless sound amplification system, method, and terminal, which can effectively reduce the sound amplification time delay and is convenient to carry.
  • an embodiment of the present application provides a wireless loudspeaker system, which includes a terminal and a loudspeaker device, wherein the terminal is wirelessly connected to the loudspeaker device, and the terminal includes an audio module, a first microphone, and a second microphone; specifically Yes, the first microphone is used to collect the first voice signal corresponding to the user’s speaking, and transmits the first voice signal to the audio module; the second microphone is used to collect the second voice signal corresponding to the user’s speaking, and The second voice signal is transmitted to the audio module; it can be understood that the first microphone and the second microphone simultaneously collect the human voice of the user, but because the distance between the first microphone and the second microphone is different from the user, the collected voice signal Not the same.
  • the audio module is used to process the first voice signal and the second voice signal in response to the acquired first voice signal and the second voice signal to obtain the playback voice signal, and transmit the acquired playback voice signal to the PA Equipment: Amplifying equipment, used to amplify the broadcast voice signal in response to the acquired broadcast voice signal.
  • the terminal can process the collected voice signal at the processor layer, specifically the audio module, so as to realize a low-latency loopback path to reduce the impact of the amplification delay on the amplification effect, and,
  • the user can amplify the human voice by only using a mobile phone and a loudspeaker device, thus providing a portable loudspeaker system, so that the user can connect to the loudspeaker device through a mobile phone in a meeting, live broadcast, class, etc. Amplification can be achieved.
  • the audio module includes an audio analog-to-digital processing module and an audio digital signal processing module; specifically, the audio analog-to-digital processing module is used to respond to the acquired first voice signal and second voice signal, Perform analog-to-digital conversion on the first voice signal to obtain a first digital voice signal, and perform analog-to-digital conversion on the second voice signal to obtain a second digital voice signal, and transmit the first digital voice signal and the second digital voice signal To the audio digital signal processing module, where the first voice signal and the second voice signal are analog signals; the audio digital signal processing module is used to respond to the acquired first digital voice signal and the second digital voice signal, based on a preset The amplifying algorithm performs sound effect processing on the first digital voice signal and the second digital voice signal to obtain the third digital voice signal; performs mixing processing on the third digital voice signal to obtain the playback voice signal, and, the playback voice signal It is transmitted to the audio analog-digital processing module; the audio analog-digital processing module is also used to transmit the acquired playback voice signal to the
  • the hardware level of the terminal is realized, specifically the processing of the collected voice signal inside the audio module in the processor, so as to realize a low-latency exchange method to reduce the delay of the sound amplification and the sound amplification The effect of the effect.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the calling algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third Digital voice signal, wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the anti-howling processing during the amplification process is realized.
  • the voice signal collected by the first microphone is filtered out of the voice signal collected by the second microphone to obtain the voice signal corresponding to the user’s human voice. Thereby improving the accuracy of filtering interference signals.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm is specific Including: if the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter out the first voice signal Interfering signal.
  • the anti-microbial spray processing in the sound amplification process is realized, and the interference signal in the voice signal collected by the first microphone is filtered through the voice signal collected by the second microphone, so as to eliminate the wheat spray phenomenon.
  • an embodiment of the present application provides a wireless loudspeaker system.
  • the system includes an earphone and a terminal, the earphone is connected to the loudspeaker device in wired or wireless connection, the terminal includes an audio module and a speaker, and the earphone includes a first microphone and a second microphone.
  • the first microphone is used to collect the first voice signal and transmit the first voice signal to the audio module
  • the second microphone is used to collect the second voice signal and transmit the second voice signal to the audio module
  • the audio module Used to process the first voice signal and the second voice signal in response to the acquired first voice signal and the second voice signal to obtain the played voice signal, and transmit the played voice signal to the speaker
  • the speaker is used to respond Based on the acquired playback voice signal, the playback voice signal is amplified.
  • the terminal can process the collected voice signal at the processor layer, specifically the audio module, so as to realize a low-latency loopback path to reduce the impact of the amplification delay on the amplification effect, and,
  • the user can realize the amplification of the human voice only by connecting the mobile phone with the headset, thereby providing a portable sound reinforcement system, so that the user can use the headset and the mobile phone in scenes such as meetings, live broadcasts, and lectures. Amplification can be achieved.
  • the audio module includes an audio analog-to-digital processing module and an audio digital signal processing module; the audio analog-to-digital processing module is used to respond to the acquired first voice signal and second voice signal, Perform analog-to-digital conversion of the voice signal to obtain a first digital voice signal, and perform analog-to-digital conversion on the second voice signal to obtain a second digital voice signal, and transmit the first digital voice signal and the second digital voice signal to the audio digital A signal processing module, where the first voice signal and the second voice signal are analog signals; the audio digital signal processing module is used to respond to the acquired first digital voice signal and the second digital voice signal, based on a preset sound amplification algorithm , Perform sound effect processing on the first digital voice signal and the second digital voice signal to obtain a third digital voice signal; perform mixing processing on the third digital voice signal to obtain a fourth digital voice signal, and combine the fourth digital voice signal The signal is transmitted to the audio analog-to-digital processing module; the audio analog-to-digital processing module is also used to perform analog-to
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the calling algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third Digital voice signal, wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm is specific Including: if the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter out the first voice signal Interfering signal.
  • the embodiments of the present application provide a wireless loudspeaker system, which is characterized in that it includes a headset, a mobile phone, and a loudspeaker device.
  • the headset includes a first microphone and a second microphone; the first microphone is used to collect the first voice signal and transmit the first voice signal to the audio module; the second microphone is used to collect the second voice signal, and The second voice signal is transmitted to the audio module; the audio module is used to process the first voice signal and the second voice signal in response to the acquired first voice signal and the second voice signal to obtain the playback voice signal, and
  • the playback voice signal is transmitted to the amplification device; the amplification device is used to amplify the playback voice signal in response to the acquired playback voice signal.
  • the terminal can process the collected voice signal at the processor layer, specifically the audio module, so as to realize a low-latency loopback path to reduce the impact of the amplification delay on the amplification effect, and,
  • the user can realize the amplification of the human voice only by connecting the mobile phone with the headset, thereby providing a portable sound reinforcement system, so that the user can use the headset and the mobile phone in scenes such as meetings, live broadcasts, and lectures. Amplification can be achieved.
  • the audio module includes an audio analog-to-digital processing module and an audio digital signal processing module; the audio analog-to-digital processing module is used to respond to the acquired first voice signal and second voice signal, Perform analog-to-digital conversion on the voice signal to obtain a first digital voice signal, and perform analog-to-digital conversion on the second voice signal to obtain a second digital voice signal, and transmit the first digital voice signal and the second digital voice signal to the audio digital A signal processing module, where the first voice signal and the second voice signal are analog signals; the audio digital signal processing module is used to respond to the acquired first digital voice signal and the second digital voice signal, based on a preset sound amplification algorithm , Perform sound effect processing on the first digital voice signal and the second digital voice signal to obtain a third digital voice signal; perform mixing processing on the third digital voice signal to obtain a playback voice signal, and transmit the playback voice signal to audio
  • the analog-digital processing module; the audio analog-digital processing module is also used to transmit the acquired playback voice signal to the amplifying
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the calling algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third Digital voice signal, wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm is specific Including: if the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter out the first voice signal Interfering signal.
  • an embodiment of the present application provides a terminal.
  • the terminal includes an audio module and a pickup device; the pickup device is used to collect voice signals and transmit the voice signals to the audio module; and the audio module is used to respond to the acquisition
  • the received voice signal is processed to obtain the broadcast voice signal, and the broadcast voice signal is transmitted to the amplification equipment.
  • the terminal can process the collected voice signal at the processor layer, specifically the audio module, so as to realize a low-latency loopback path to reduce the impact of the amplification delay on the amplification effect, and,
  • the user can realize the amplification of the human voice only by connecting the mobile phone with the headset, thereby providing a portable sound reinforcement system, so that the user can use the headset and the mobile phone in scenes such as meetings, live broadcasts, and lectures. Amplification can be achieved.
  • the sound pickup device is a microphone on an earphone connected to the terminal in a wired or wireless connection, and includes a first microphone and a second microphone.
  • the sound pickup device is a microphone with the terminal, including a first microphone and a second microphone.
  • the voice signal includes a first voice signal and a second voice signal; the first microphone is used to collect the first voice signal and transmit the first voice signal to the audio module; the second microphone is used To collect the second voice signal and transmit the second voice signal to the audio module; the first voice signal and the second voice signal belong to the voice signal; the audio module is used to respond to the acquired first voice signal and the second voice signal , The first voice signal and the second voice signal are processed to obtain the broadcast voice signal, and the broadcast voice signal is transmitted to the amplification equipment.
  • the amplifying device is the speaker of the terminal;
  • the audio module includes an audio analog-to-digital processing module and an audio digital signal processing module;
  • the audio analog-to-digital processing module is used to respond to the acquired first voice signal and For the second voice signal, perform analog-to-digital conversion on the first voice signal to obtain a first digital voice signal, and perform analog-to-digital conversion on the second voice signal to obtain a second digital voice signal, and combine the first digital voice signal with the first digital voice signal.
  • Two digital voice signals are transmitted to the audio digital signal processing module, where the first voice signal and the second voice signal are analog signals; the audio digital signal processing module is used to respond to the acquired first digital voice signal and second digital voice Signal, based on a preset sound amplification algorithm, perform sound effect processing on the first digital voice signal and the second digital voice signal to obtain a third digital voice signal; perform mixing processing on the third digital voice signal to obtain a fourth digital voice signal , And, transmitting the fourth digital voice signal to the audio analog-to-digital processing module; the audio analog-to-digital processing module is also used to perform analog-to-digital conversion on the fourth digital voice signal in response to the acquired fourth digital voice signal to obtain playback The voice signal, and the playback voice signal is transmitted to the loudspeaker; among them, the playback voice signal is an analog signal.
  • the loudspeaker device is a loudspeaker device that is wirelessly connected to the terminal;
  • the audio analog-to-digital processing module is used to respond to the acquired first voice signal and second voice signal to respond to the first voice signal Perform analog-to-digital conversion to obtain the first digital voice signal, and perform analog-to-digital conversion on the second voice signal to obtain the second digital voice signal, and transmit the first digital voice signal and the second digital voice signal to audio digital signal processing Module, wherein the first voice signal and the second voice signal are analog signals;
  • the audio digital signal processing module is used to respond to the acquired first digital voice signal and the second digital voice signal, based on the preset sound amplification algorithm, Perform sound effect processing on the first digital voice signal and the second digital voice signal to obtain a third digital voice signal; perform mixing processing on the third digital voice signal to obtain a playback voice signal, and transmit the playback voice signal to the audio analog-to-digital Processing module; audio analog-to-digital processing module, which is also used to transmit the acquired playback voice signal to the
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the calling algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third Digital voice signal, wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm is specific Including: if the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter out the first voice signal Interfering signal.
  • an embodiment of the present application provides a wireless sound amplification method, which includes application to a terminal, and includes: acquiring a first voice signal collected by a first microphone of the terminal, and a second voice collected by a second microphone Signal; in response to the acquired first voice signal and second voice signal, the first voice signal and the second voice signal are processed to obtain the broadcast voice signal, and the broadcast voice signal is transmitted to the amplification device, the amplifying The device is an external device wirelessly connected to the terminal.
  • processing the first voice signal and the second voice signal includes: in response to the acquired first voice signal and the second voice signal, performing analog-to-digital conversion on the first voice signal to obtain The first digital voice signal and the second voice signal are converted from analog to digital to obtain the second digital voice signal, and the first digital voice signal and the second digital voice signal are combined, wherein the first voice signal and the second voice signal It is an analog signal; based on the preset sound amplification algorithm, the first digital voice signal and the second digital voice signal are processed by sound effects to obtain the third digital voice signal; the third digital voice signal is mixed and processed to obtain the playback voice signal , Transmit the acquired playback voice signal to the PA equipment.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the calling algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third Digital voice signal, wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm is specific Including: if the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter out the first voice signal Interfering signal.
  • an embodiment of the present application provides a wireless amplification method, which is applied to a terminal, and includes: acquiring a first voice signal collected by a first microphone of a headset, and a second voice signal collected by a second microphone of the headset Voice signal, the headset is connected to the terminal wired or wirelessly; in response to the first voice signal and the second voice signal acquired, the first voice signal and the second voice signal are processed to obtain the playback voice signal, and the speaker Play the voice signal for amplification.
  • processing the first voice signal and the second voice signal includes: in response to the acquired first voice signal and the second voice signal, performing analog-to-digital conversion on the first voice signal to obtain The first digital voice signal and the analog-to-digital conversion of the second voice signal are performed to obtain the second digital voice signal, wherein the first voice signal and the second voice signal are analog signals; Perform sound effect processing on the digital voice signal and the second digital voice signal to obtain a third digital voice signal; perform mixing processing on the third digital voice signal to obtain a fourth digital voice signal; perform analog-to-digital conversion on the fourth digital voice signal, Obtain the playback voice signal.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the calling algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third Digital voice signal, wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm is specific Including: if the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter out the first voice signal Interfering signal.
  • an embodiment of the present application provides a wireless amplification method, which is applied to a terminal, and includes: acquiring a first voice signal collected by a first microphone of a headset, and a second voice signal collected by a second microphone of the headset For voice signals, the headset is connected to the terminal in wired or wireless connection; in response to the acquired first voice signal and second voice signal, the first voice signal and the second voice signal are processed to obtain the playback voice signal, and the playback voice The signal is transmitted to a loudspeaker device, which is an external device wirelessly connected to the terminal.
  • processing the first voice signal and the second voice signal includes: in response to the acquired first voice signal and the second voice signal, performing analog-to-digital conversion on the first voice signal to obtain The first digital voice signal and the second voice signal are converted from analog to digital to obtain the second digital voice signal, and the first digital voice signal and the second digital voice signal are combined, wherein the first voice signal and the second voice signal It is an analog signal; based on the preset sound amplification algorithm, the first digital voice signal and the second digital voice signal are processed by sound effects to obtain the third digital voice signal; the third digital voice signal is mixed and processed to obtain the playback voice signal , Transmit the acquired playback voice signal to the PA equipment.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the calling algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third Digital voice signal, wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm is specific Including: if the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter out the first voice signal Interfering signal.
  • an embodiment of the present application provides a wireless amplification method, which is applied to a terminal, and includes: collecting a voice signal, in response to the acquired voice signal, processing the voice signal to obtain a playback voice signal, and The playback voice signal is transmitted to the amplifying device.
  • the sound pickup device is a microphone on an earphone connected to the terminal in a wired or wireless connection, and includes a first microphone and a second microphone.
  • the sound pickup device is a microphone with the terminal, including a first microphone and a second microphone.
  • the voice signal includes a first voice signal and a second voice signal; collecting the voice signal includes: collecting the first voice signal through the first microphone, and collecting the second voice signal through the second microphone; responding Based on the acquired first voice signal and second voice signal, the first voice signal and the second voice signal are processed to obtain the broadcast voice signal, and the broadcast voice signal is transmitted to the amplification device.
  • the loudspeaker device is the speaker of the terminal; processing the first voice signal and the second voice signal includes: responding to the acquired first voice signal and the second voice signal, processing the first voice signal and the second voice signal; Perform analog-to-digital conversion on the voice signal to obtain a first digital voice signal, and perform analog-to-digital conversion on the second voice signal to obtain a second digital voice signal, where the first voice signal and the second voice signal are analog signals; Set up a sound amplification algorithm to perform sound effect processing on the first digital voice signal and the second digital voice signal to obtain the third digital voice signal; perform mixing processing on the third digital voice signal to obtain the fourth digital voice signal; The digital voice signal undergoes analog-to-digital conversion to obtain the playback voice signal.
  • the loudspeaker device is a loudspeaker device wirelessly connected to the terminal; processing the first voice signal and the second voice signal includes: responding to the acquired first voice signal and second voice signal Signal, perform analog-to-digital conversion on the first voice signal to obtain a first digital voice signal, and perform analog-to-digital conversion on the second voice signal to obtain a second digital voice signal, and combine the first digital voice signal and the second digital voice signal Signal, wherein the first voice signal and the second voice signal are analog signals; based on a preset amplification algorithm, perform sound effect processing on the first digital voice signal and the second digital voice signal to obtain a third digital voice signal; The three digital voice signals are mixed and processed to obtain the broadcast voice signal, and the obtained broadcast voice signal is transmitted to the amplification equipment.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the calling algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third Digital voice signal, wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm is specific Including: if the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter out the first voice signal Interfering signal.
  • an embodiment of the present application provides a terminal, which includes an acquisition module, a processing module, and a sending module.
  • the acquisition module is used to acquire the first voice signal collected by the first microphone of the terminal and the second voice signal collected by the second microphone;
  • the processing module is used to respond to the acquired first voice signal and the second voice signal.
  • the voice signal, the first voice signal and the second voice signal are processed to obtain the playback voice signal, and the sending module is used to transmit the playback voice signal to a loudspeaker device, and the loudspeaker device is an external wireless connection with the terminal equipment.
  • the processing module is specifically configured to perform analog-to-digital conversion on the first voice signal in response to the acquired first voice signal and second voice signal to obtain the first digital voice signal, and Perform analog-to-digital conversion of the second voice signal to obtain a second digital voice signal, and combine the first digital voice signal and the second digital voice signal, where the first voice signal and the second voice signal are analog signals; based on a preset sound amplification algorithm , Perform sound effect processing on the first digital voice signal and the second digital voice signal to obtain a third digital voice signal; perform mixing processing on the third digital voice signal to obtain a playback voice signal, and transmit the acquired playback voice signal to Loudspeaker equipment.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the calling algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third Digital voice signal, wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm is specific Including: if the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter out the first voice signal Interfering signal.
  • an embodiment of the present application provides a terminal, which includes an acquisition module, a processing module, and a speaker module.
  • the acquisition module is used to acquire the first voice signal collected by the first microphone of the earphone and the second voice signal collected by the second microphone of the earphone, and the earphone is connected to the terminal in wired or wireless connection;
  • the processing module uses In response to the acquired first voice signal and second voice signal, the first voice signal and the second voice signal are processed to obtain a played voice signal, and the speaker module is used to amplify the played voice signal.
  • the processing module is specifically configured to perform analog-to-digital conversion on the first voice signal in response to the acquired first voice signal and second voice signal to obtain the first digital voice signal, and
  • the second voice signal is converted from analog to digital to obtain the second digital voice signal, where the first voice signal and the second voice signal are analog signals; based on the preset sound amplification algorithm, the first digital voice signal and the second digital voice signal are performed Sound effect processing to obtain a third digital voice signal; perform mixing processing on the third digital voice signal to obtain a fourth digital voice signal; perform analog-to-digital conversion on the fourth digital voice signal to obtain a playback voice signal.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the calling algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third Digital voice signal, wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm is specific Including: if the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter out the first voice signal Interfering signal.
  • an embodiment of the present application provides a terminal, which includes: an acquisition module, a processing module, and a sending module.
  • the acquisition module is used to acquire the first voice signal collected by the first microphone of the earphone and the second voice signal collected by the second microphone of the earphone, and the earphone is connected to the terminal in wired or wireless connection;
  • the processing module uses In response to the acquired first voice signal and the second voice signal, the first voice signal and the second voice signal are processed to obtain the playback voice signal, and the sending module is used to transmit the playback voice signal to the amplification device, so
  • the loudspeaker device is an external device wirelessly connected to the terminal.
  • the processing module is specifically configured to perform analog-to-digital conversion on the first voice signal in response to the acquired first voice signal and second voice signal to obtain the first digital voice signal, and
  • the second voice signal undergoes analog-to-digital conversion to obtain a second digital voice signal, and combines the first digital voice signal and the second digital voice signal, where the first voice signal and the second voice signal are analog signals; based on preset amplification Algorithm, perform sound effect processing on the first digital voice signal and the second digital voice signal to obtain the third digital voice signal; perform mixing processing on the third digital voice signal to obtain the playback voice signal, and transmit the acquired playback voice signal To amplifying equipment.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the calling algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third Digital voice signal, wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm is specific Including: if the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter out the first voice signal Interfering signal.
  • an embodiment of the present application provides a terminal, including: an acquisition module, a processing module, and a sending module.
  • the acquisition module is used to collect voice signals
  • the processing module is used to respond to the acquired voice signals.
  • the voice signal is processed to obtain the playback voice signal
  • the sending module is used to transmit the playback voice signal to the amplifying device.
  • the sound pickup device is a microphone on an earphone connected to the terminal in a wired or wireless connection, and includes a first microphone and a second microphone.
  • the sound pickup device is a microphone with the terminal, including a first microphone and a second microphone.
  • the voice signal includes a first voice signal and a second voice signal
  • the acquisition module is configured to collect the first voice signal through the first microphone and collect the second voice signal through the second microphone
  • process The module is used to process the first voice signal and the second voice signal in response to the acquired first voice signal and the second voice signal to obtain the played voice signal, and transmit the played voice signal to the amplification device.
  • the loudspeaker device is the speaker of the terminal;
  • the processing module is specifically configured to perform analog-to-digital conversion on the first voice signal in response to the acquired first voice signal and the second voice signal to obtain the first voice signal.
  • Digital voice signal, and analog-to-digital conversion is performed on the second voice signal to obtain the second digital voice signal, wherein the first voice signal and the second voice signal are analog signals; based on the preset sound amplification algorithm, the first digital voice signal
  • the signal and the second digital voice signal are processed by sound effects to obtain the third digital voice signal; the third digital voice signal is mixed and processed to obtain the fourth digital voice signal; the fourth digital voice signal is converted from analog to digital to obtain the playback voice signal.
  • the loudspeaker device is a loudspeaker device wirelessly connected to the terminal;
  • the processing module is specifically configured to perform analog-to-digital processing on the first voice signal in response to the acquired first voice signal and second voice signal. Convert to obtain the first digital voice signal, and perform analog-to-digital conversion on the second voice signal to obtain the second digital voice signal, and combine the first digital voice signal and the second digital voice signal, where the first voice signal and the second voice signal Second, the voice signal is an analog signal; based on the preset sound amplification algorithm, perform sound effect processing on the first digital voice signal and the second digital voice signal to obtain a third digital voice signal; perform mixing processing on the third digital voice signal to obtain The voice signal is played, and the acquired voice signal is transmitted to the amplifying device.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the calling algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third Digital voice signal, wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm is specific Including: if the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter out the first voice signal Interfering signal.
  • an embodiment of the present application provides a computer-readable medium for storing a computer program, the computer program including instructions for executing the fifth aspect or any possible implementation of the fifth aspect.
  • an embodiment of the present application provides a computer-readable medium for storing a computer program, and the computer program includes instructions for executing the sixth aspect or any possible implementation of the sixth aspect.
  • the embodiments of the present application provide a computer-readable medium for storing a computer program, and the computer program includes instructions for executing the seventh aspect or any possible implementation of the seventh aspect.
  • the embodiments of the present application provide a computer-readable medium for storing a computer program, and the computer program includes instructions for executing the eighth aspect or any possible implementation of the eighth aspect.
  • an embodiment of the present application provides a computer program, the computer program including instructions for executing the fifth aspect or any possible implementation of the fifth aspect.
  • an embodiment of the present application provides a computer program, the computer program including instructions for executing the sixth aspect or any possible implementation of the sixth aspect.
  • embodiments of the present application provide a computer program, the computer program including instructions for executing the seventh aspect or any possible implementation of the seventh aspect.
  • an embodiment of the present application provides a computer program, which includes instructions for executing the eighth aspect or any possible implementation of the eighth aspect.
  • an embodiment of the present application provides a chip, which includes a processing circuit and transceiver pins.
  • the transceiver pin and the processing circuit communicate with each other through an internal connection path, and the processing circuit executes the method in the fifth aspect or any one of the possible implementation manners of the fifth aspect to control the receiving pin to receive the signal, and Control the sending pin to send signals.
  • an embodiment of the present application provides a chip, which includes a processing circuit and transceiver pins.
  • the transceiver pin and the processing circuit communicate with each other through an internal connection path, and the processing circuit executes the method in the sixth aspect or any one of the possible implementation manners of the sixth aspect to control the receiving pin to receive the signal, and Control the sending pin to send signals.
  • an embodiment of the present application provides a chip, which includes a processing circuit and transceiver pins.
  • the transceiver pin and the processing circuit communicate with each other through an internal connection path, and the processing circuit executes the method in the seventh aspect or any one of the possible implementation manners of the seventh aspect to control the receiving pin to receive the signal, and Control the sending pin to send signals.
  • an embodiment of the present application provides a chip, which includes a processing circuit and transceiver pins.
  • the transceiver pin and the processing circuit communicate with each other through an internal connection path, and the processing circuit executes the method in the eighth aspect or any one of the possible implementations of the eighth aspect to control the receiving pin to receive the signal, and Control the sending pin to send signals.
  • Fig. 1 is a schematic diagram of a loudspeaker device exemplarily shown
  • Fig. 2 is a schematic structural diagram of a terminal shown by way of example
  • Fig. 3a is one of the schematic diagrams of an exemplary wireless amplification function interface
  • Fig. 3b is one of the schematic diagrams of an exemplary wireless amplification function interface
  • Fig. 4 is one of schematic diagrams showing an exemplary application scenario
  • FIG. 5 is one of the schematic flowcharts of a wireless sound amplification method provided by the implementation of this application.
  • FIG. 6 is one of the schematic diagrams showing an exemplary internal processing flow of a mobile phone
  • FIG. 7 is one of schematic diagrams showing an exemplary application scenario
  • FIG. 8 is one of the schematic flowcharts of a wireless sound amplification method provided by the implementation of this application.
  • FIG. 9 is one of the schematic diagrams showing an exemplary internal processing flow of a mobile phone
  • Fig. 10 is a schematic diagram showing an exemplary echo generation principle
  • Fig. 11 is a schematic diagram of the principle of an exemplary anti-howling algorithm
  • FIG. 12 is a schematic flowchart of an anti-howling algorithm provided by the implementation of the present application.
  • FIG. 13 is a schematic flow chart of an algorithm for preventing wheat blowout provided by the implementation of the present application.
  • Fig. 14 is one of schematic diagrams showing an exemplary application scenario
  • FIG. 15 is one of the schematic flowcharts of a wireless sound amplification method provided by the implementation of this application.
  • FIG. 16 is one of the schematic diagrams showing an exemplary internal processing flow of a mobile phone
  • FIG. 17 is a schematic flowchart of an anti-howling algorithm provided by the implementation of the present application.
  • FIG. 18 is one of schematic diagrams of an application scenario shown by way of example.
  • FIG. 19 is one of the schematic flowcharts of a wireless sound amplification method provided by the implementation of this application.
  • FIG. 20 is one of the schematic diagrams showing an exemplary internal processing flow of a mobile phone
  • FIG. 21 is one of schematic diagrams showing an exemplary application scenario
  • FIG. 22 is one of the schematic flowcharts of a wireless sound amplification method provided by the implementation of this application.
  • FIG. 23 is one of schematic diagrams of an application scenario shown by way of example.
  • FIG. 24 is one of the schematic structural diagrams of a terminal provided by the implementation of the present application.
  • FIG. 25 is one of the schematic structural diagrams of a terminal provided by the implementation of the present application.
  • FIG. 26 is one of the schematic structural diagrams of a terminal provided by the implementation of the present application.
  • FIG. 27 is one of the schematic structural diagrams of a terminal provided by the implementation of the present application.
  • FIG. 28 is one of the schematic structural diagrams of a terminal provided by the embodiment of the present application.
  • first and second in the description and claims of the embodiments of the present application are used to distinguish different objects, rather than to describe a specific order of objects.
  • first target object and the second target object are used to distinguish different target objects, rather than to describe the specific order of the target objects.
  • words such as “exemplary” or “for example” are used as examples, illustrations, or illustrations. Any embodiment or design solution described as “exemplary” or “for example” in the embodiments of the present application should not be construed as being more preferable or advantageous than other embodiments or design solutions. To be precise, words such as “exemplary” or “for example” are used to present related concepts in a specific manner.
  • multiple processing units refer to two or more processing units; multiple systems refer to two or more systems.
  • the terminal involved in this application may be a terminal device equipped with a microphone, such as a mobile phone, a smart phone, a notebook computer, a tablet computer (personal computer, PC), and so on.
  • a microphone such as a mobile phone, a smart phone, a notebook computer, a tablet computer (personal computer, PC), and so on.
  • FIG. 2 shows a schematic structural diagram when the terminal is a mobile phone.
  • the mobile phone 100 may include a processor 110, an external memory interface 120, an internal memory 121, a universal serial bus (USB) interface 130, a charging management module 140, a power management module 141, a battery 142, an antenna 1, an antenna 2, Mobile communication module 150, wireless communication module 160, audio module 170, speaker 170A, receiver 170B, microphone 170C, earphone interface 170D, sensor module 180, buttons 190, motor 191, indicator 192, camera 193, display screen 194, and user An identification module (subscriber identification module, SIM) card interface 195, etc.
  • SIM subscriber identification module
  • the sensor module 180 may include a pressure sensor 180A, a gyroscope sensor 180B, an air pressure sensor 180C, a magnetic sensor 180D, an acceleration sensor 180E, a distance sensor 180F, a proximity sensor 180G, a fingerprint sensor 180H, a temperature sensor 180J, a touch sensor 180K, and ambient light Sensor 180L, bone conduction sensor 180M, etc.
  • the structure illustrated in the embodiment of the present application does not constitute a specific limitation on the mobile phone 100.
  • the mobile phone 100 may include more or fewer components than those shown in the figure, or combine certain components, or split certain components, or arrange different components.
  • the illustrated components can be implemented in hardware, software, or a combination of software and hardware.
  • the processor 110 may include one or more processing units.
  • the processor 110 may include an application processor (AP), a modem processor, a graphics processing unit (GPU), and an image signal processor. (image signal processor, ISP), controller, memory, video codec, digital signal processor (digital signal processor, DSP), baseband processor, and/or neural-network processing unit (NPU) Wait.
  • AP application processor
  • modem processor modem processor
  • GPU graphics processing unit
  • image signal processor image signal processor
  • ISP image signal processor
  • controller memory
  • video codec digital signal processor
  • DSP digital signal processor
  • NPU neural-network processing unit
  • the different processing units may be independent devices or integrated in one or more processors.
  • the controller may be the nerve center and command center of the mobile phone 100.
  • the controller can generate operation control signals according to the instruction operation code and timing signals to complete the control of fetching instructions and executing instructions.
  • a memory may also be provided in the processor 110 to store instructions and data.
  • the memory in the processor 110 is a cache memory.
  • the memory can store instructions or data that have just been used or recycled by the processor 110. If the processor 110 needs to use the instruction or data again, it can be directly called from the memory. Repeated accesses are avoided, the waiting time of the processor 110 is reduced, and the efficiency of the system is improved.
  • the processor 110 may include one or more interfaces.
  • the interface can include an integrated circuit (inter-integrated circuit, I2C) interface, an integrated circuit built-in audio (inter-integrated circuit sound, I2S) interface, a pulse code modulation (pulse code modulation, PCM) interface, and a universal asynchronous transmitter (universal asynchronous) interface.
  • I2C integrated circuit
  • I2S integrated circuit built-in audio
  • PCM pulse code modulation
  • UART universal asynchronous transmitter
  • MIPI mobile industry processor interface
  • GPIO general-purpose input/output
  • SIM subscriber identity module
  • USB Universal Serial Bus
  • the USB interface 130 is an interface that complies with the USB standard specification.
  • the USB interface 130 can be used to connect a charger to charge the mobile phone 100, and can also be used to transfer data between the mobile phone 100 and peripheral devices. It can also be used to connect earphones.
  • the wired earphone in the embodiment of the present application can be connected to a mobile phone through a USB interface.
  • the interface connection relationship between the modules illustrated in the embodiment of the present application is merely a schematic description, and does not constitute a structural limitation of the mobile phone 100.
  • the mobile phone 100 may also adopt different interface connection modes in the foregoing embodiments, or a combination of multiple interface connection modes.
  • the charging management module 140 is used to receive charging input from the charger.
  • the power management module 141 is used to connect the battery 142, the charging management module 140 and the processor 110.
  • the power management module 141 receives input from the battery 142 and/or the charge management module 140, and supplies power to internal devices.
  • the wireless communication function of the mobile phone 100 can be realized by the antenna 1, the antenna 2, the mobile communication module 150, the wireless communication module 160, the modem processor, and the baseband processor.
  • the mobile phone 100 can implement audio functions through the audio module 170, the speaker 170A, the receiver 170B, the microphone 170C, the earphone interface 170D, and the application processor. For example, music playback, recording, etc.
  • the audio module 170 is used to convert digital audio information into an analog audio signal for output, and is also used to convert an analog audio input into a digital audio signal.
  • the audio module 170 can also be used to encode and decode audio signals.
  • the audio module 170 may be provided in the processor 110, or part of the functional modules of the audio module 170 may be provided in the processor 110.
  • the speaker 170A also called “speaker” is used to convert audio electrical signals into sound signals.
  • the mobile phone 100 can listen to music through the speaker 170A, or listen to a hands-free call.
  • the receiver 170B also called “earpiece” is used to convert audio electrical signals into sound signals.
  • the mobile phone 100 answers a call or a voice message, it can receive the voice by bringing the receiver 170B close to the human ear.
  • the microphone 170C also called “microphone”, “microphone”, is used to convert sound signals into electrical signals.
  • the user can make a sound by approaching the microphone 170C through the human mouth, and input the sound signal into the microphone 170C.
  • the mobile phone 100 may be provided with one or more microphones 170C.
  • the mobile phone 100 may be provided with two microphones 170C, which can implement noise reduction functions in addition to collecting sound signals.
  • the mobile phone 100 may also be provided with three, four or more microphones 170C to collect sound signals, reduce noise, identify sound sources, and realize directional recording functions.
  • the earphone interface 170D is used to connect wired earphones.
  • the earphone interface 170D may be a USB interface 130, or a 3.5mm open mobile terminal platform (OMTP) standard interface, and a cellular telecommunications industry association (cellular telecommunications industry association of the USA, CTIA) standard interface.
  • OMTP open mobile terminal platform
  • CTIA cellular telecommunications industry association of the USA, CTIA
  • the mobile phone 100 implements a display function through a GPU, a display screen 194, and an application processor.
  • the display screen 194 is used to display images, videos, etc., for example, can be used to display the wireless amplification setting interface in the present application.
  • the mobile phone 100 can realize the shooting function through an ISP, a camera 193, a video codec, a GPU, a display screen 194, and an application processor.
  • the external memory interface 120 may be used to connect an external memory card, such as a Micro SD card, so as to expand the storage capacity of the mobile phone 100.
  • an external memory card such as a Micro SD card
  • the internal memory 121 may be used to store computer executable program code, where the executable program code includes instructions.
  • the processor 110 executes various functional applications and data processing of the mobile phone 100 by running instructions stored in the internal memory 121.
  • the internal memory 121 may include a storage program area and a storage data area.
  • the storage program area can store an operating system, one or more application programs (such as a sound playback function, an image playback function, etc.) required by one or more functions.
  • the data storage area can store data (such as audio data, phone book, etc.) created during the use of the mobile phone 100.
  • the internal memory 121 may include a high-speed random access memory, and may also include a non-volatile memory, such as one or more magnetic disk storage devices, flash memory devices, universal flash storage (UFS), and the like.
  • the button 190 includes a power-on button, a volume button, and so on.
  • the button 190 may be a mechanical button. It can also be a touch button.
  • the mobile phone 100 can receive key input, and generate key signal input related to user settings and function control of the mobile phone 100.
  • the motor 191 can generate vibration prompts.
  • the indicator 192 may be an indicator light, which may be used to indicate the charging status, power change, or to indicate messages, missed calls, notifications, and so on.
  • the SIM card interface 195 is used to connect to the SIM card. The SIM card can be connected to and separated from the mobile phone 100 by inserting into the SIM card interface 195 or pulling out from the SIM card interface 195.
  • the function key of the wireless amplification function can also be in the drop-down interface, as shown in Fig. 3b.
  • the terminal can obtain the user's voice signal, and pass the obtained user's voice signal through the audio module of the terminal, for example, the audio module 170 in FIG.
  • the audio digital signal processing module in the module outputs the processed voice signal.
  • the audio digital signal processing module may be a high-fidelity (High-Fidelity, HIFI) module, which in the embodiment of the present application is mainly used to perform sound amplification algorithm processing and mixing processing on the signal obtained by the audio module , The specific functions will be described in detail in the following embodiments.
  • the device for collecting the user's voice signal may be a headset microphone, or a microphone built into the terminal.
  • the terminal can amplify the processed voice signal through its own speaker, or it can output the processed voice signal to an external device, and the external device can amplify the voice signal
  • the external device may be an audio device, for example, a Bluetooth audio device, or a TV audio device.
  • the device for collecting the user's voice signal may be a microphone in the terminal. After the terminal processes the collected voice signal, the processed voice signal is amplified through the speaker of the terminal. Specific implementation For example, please refer to scene one.
  • the device for collecting the user's voice signal may be a microphone in the terminal. After the terminal processes the collected voice signal, the processed voice signal may be output to the audio equipment connected to it. The sound is amplified by the audio equipment, and the specific embodiment can refer to scene two.
  • the device for collecting the user's voice signal may be a microphone of a headset connected to the terminal (hereinafter referred to as a headset microphone). After the terminal obtains the user's voice signal collected by the headset microphone, the voice signal is processed Process, and amplify the processed voice signal through the terminal speaker. For specific embodiments, refer to scenario three.
  • the device for collecting the user's voice signal may be a microphone of a headset connected to the terminal. After the terminal obtains the user's voice signal collected by the headset microphone, the voice signal is processed and processed. The voice signal of is output to the audio equipment connected to it, and the processed voice signal is amplified through the terminal speaker.
  • the terminal After the terminal obtains the user's voice signal collected by the headset microphone, the voice signal is processed and processed. The voice signal of is output to the audio equipment connected to it, and the processed voice signal is amplified through the terminal speaker.
  • scenario four please refer to scenario four.
  • the device for collecting the user's voice signal may be a headset connected to the terminal.
  • the headset can establish a connection with the audio equipment through the terminal, for example, a Bluetooth connection is established.
  • the processed voice signal is output to the audio device through the Bluetooth connection with the audio device, and the processed voice signal is amplified through the terminal speaker.
  • a Bluetooth connection is established.
  • the processed voice signal is output to the audio device through the Bluetooth connection with the audio device, and the processed voice signal is amplified through the terminal speaker.
  • refer to Scene Five refer to Scene Five.
  • connection may be a wired connection, for example, the headset and the mobile phone may be connected through a headset cord.
  • the “connection” mentioned in this application can also be a wireless connection.
  • the headset is a wireless headset that can be connected to a mobile phone via Bluetooth, or the mobile phone and audio equipment can establish a wireless connection via Bluetooth.
  • FIG 4 is a schematic diagram of an exemplary application scenario.
  • the application scenario includes a mobile phone.
  • the user can speak to the mobile phone, and the mobile phone can collect the user's voice signal through the mobile phone microphone, and Amplify the voice signal.
  • the scene may be a live broadcast scene or a speech scene, etc., that is, the user can amplify the speech through a mobile phone conversation while giving a speech, without having to carry a special loudspeaker device.
  • the mobile phones in FIG. 4, that is, the number of terminals described in this application, are merely illustrative examples. In actual applications, the number of mobile phones may be one or more, which is not limited in this application. It should be further noted that in this scenario and the following scenarios, the terminal is a mobile phone as an example for description. In other embodiments, the terminal may also be other devices such as a tablet, which is not limited in this application.
  • FIG. 5 is a schematic flowchart of a wireless sound amplification method in an embodiment of the application, and in FIG. 5:
  • Step 101 The mobile phone collects the user's voice signal through the mobile phone microphone.
  • the user can speak to the mobile phone, specifically to the microphone in the mobile phone.
  • the mobile phone can collect the user's voice signal through the microphone, which can also be called a human voice signal.
  • Step 102 The mobile phone processes the voice signal.
  • FIG. 6 is a schematic diagram of a process of processing a voice signal by a mobile phone.
  • the mobile phone has two microphones, microphone 1 and microphone 2, respectively. It should be noted that this application only takes microphone 1 and microphone 2 as an example for description. In fact, the number of microphones in the mobile phone can be two or more, which is not limited in this application.
  • the mobile phone can collect the user's voice signal through microphone 1 and microphone 2.
  • the mobile phone processes the user's voice signal as A loopback process, in which the mobile phone obtains the user’s voice signal from the microphone.
  • the mobile phone can be called uplink data.
  • the mobile phone processes the uplink data, that is, the user’s voice signal.
  • the processed voice signal output can be called downlink data or playback data.
  • the loopback process which specifically includes:
  • the audio module of the mobile phone may include an audio analog-to-digital processing module, which can perform analog-to-digital conversion on the user’s voice signals collected by microphone 1 and microphone 2, ie The analog voice signal corresponding to the human voice is converted into a digital voice signal, and the digital voice signal is transmitted to the audio digital signal processing module.
  • an audio analog-to-digital processing module which can perform analog-to-digital conversion on the user’s voice signals collected by microphone 1 and microphone 2, ie
  • the analog voice signal corresponding to the human voice is converted into a digital voice signal, and the digital voice signal is transmitted to the audio digital signal processing module.
  • the audio digital signal processing module can process the digital voice signal based on a preset sound amplification algorithm, and mix the processed digital voice signal with the playback link to obtain downlink data.
  • the mixing process can be understood as mixing useful information with a carrier to obtain a signal carrying useful information. That is to say, the audio digital signal processing module combines useful information, that is, the processed digital voice signal with the carrier, That is, the playback link is mixed and the downlink data is obtained.
  • the playback link may also be referred to as an audio stream, a carrier, or a playback stream, which is not limited in this application.
  • the audio digital signal processing module is based on the amplification algorithm, and the processing of the digital voice signal may include, but is not limited to: environmental noise reduction and enhancement processing, anti-howling processing, anti-blooming wheat processing, sampling rate and Bit width conversion processing.
  • environmental noise reduction and enhancement processing can be used to reduce noise signals entering the microphone, such as wind noise, environmental noise, and other interference signals, and can appropriately enhance the user’s voice signal to improve the clarity and clarity of the human voice.
  • Signal-to-noise ratio The processing procedure can be referred to the existing technology, which will not be repeated in this application.
  • sampling rate and bit width conversion processing can be used to convert the voice signal into a sampling rate and bit width compatible with the system, such as 48kHz/16bit.
  • the processing procedure can be referred to the existing technology, which will not be repeated in this application.
  • anti-howling processing can be used to eliminate the howling phenomenon.
  • the mobile phone can use the adaptive echo cancellation algorithm (Acoustic Echo Cancellation, AEC) in the prior art to eliminate howling.
  • AEC Acoustic Echo Cancellation
  • This method is mainly based on the downlink broadcast signal, that is, the downlink data described above.
  • Simulate the echo signal ref which can also be understood as an interference signal.
  • the signal (or sound) collected by the microphone of the mobile phone includes useful human voice, that is, the user’s voice signal and the echo signal ref, that is, the interference signal.
  • the mobile phone can collect The echo signal ref is removed from the received signal, so as to realize echo cancellation.
  • the specific method can refer to the prior art, and this application will not repeat it.
  • the anti-spraying treatment can be used to eliminate the spraying phenomenon, and the specific method will be described in detail in the second scenario.
  • Step 103 The mobile phone outputs the processed voice signal through the speaker.
  • the audio digital signal processing module transmits the processed digital voice signal, that is, downlink data, to the audio analog-to-digital processing module, and the audio analog-to-digital processing module performs digital-to-analog conversion, namely ,
  • the digital voice signal is converted into an analog voice signal, and played through the speaker of the mobile phone, so as to realize the amplification of the user's voice signal.
  • the user can adjust the volume of the mobile phone to adjust the sound reinforcement effect of the speaker.
  • FIG 7 is a schematic diagram of an application scenario of an embodiment of the application.
  • the application scenario includes a mobile phone and audio equipment.
  • the mobile phone and the audio equipment establish a communication connection, such as a Bluetooth connection.
  • the user It can speak to the mobile phone, and the mobile phone can collect the user's voice signal through the microphone of the mobile phone, and amplify the voice signal through the audio equipment.
  • the scene may also be a live broadcast scene or a speech scene, etc., that is, the user can speak to a mobile phone during a speech, and amplify the dialogue sound through the audio equipment.
  • the number of mobile phones and audio equipment in FIG. 7 is only a schematic example. In actual applications, the number of mobile phones and audio equipment may be one or more, which is not limited in this application.
  • FIG. 8 is a schematic flowchart of a wireless sound amplification method in an embodiment of the application, and in FIG. 8:
  • Step 201 The mobile phone collects the user's voice signal through the mobile phone microphone.
  • the user can speak to the mobile phone, specifically to the microphone in the mobile phone, and the mobile phone can collect the user's voice signal through the microphone.
  • the mobile phone can establish a connection with nearby audio equipment.
  • the connection between the mobile phone and the audio equipment refers to a wireless connection.
  • the mobile phone Bluetooth connection can be established with audio equipment.
  • the mobile phone can establish a WIFI connection with the audio device, which is not limited in this application.
  • the description is made by taking the connection between the mobile phone and the audio device through Bluetooth as an example.
  • the mobile phone can set the Bluetooth mode of the mobile phone to the Synchronous Connection Oritened (SCO) mode, which is used for synchronous voice transmission and has the characteristics of low transmission delay.
  • SCO Synchronous Connection Oritened
  • Step 202 The mobile phone processes the voice signal.
  • FIG. 9 is a schematic diagram of a process of processing a voice signal by a mobile phone.
  • the mobile phone has two microphones, microphone 1 and microphone 2, respectively.
  • the mobile phone collects the user's voice signal through the microphone 1 and the microphone 2, and processes the user's voice signal, that is, the uplink data in Figure 9 (the concept can be referred to above) through the audio digital signal processing module to obtain the downlink data .
  • the audio digital signal processing module may process the digital voice signal based on a preset amplification algorithm, and mix the processed digital voice signal with the playback link to obtain downlink data.
  • the processing process of the audio digital signal processing module on the uplink data includes but is not limited to: environmental noise reduction and enhancement processing, anti-howling processing, anti-blooming wheat processing, sampling rate and bit width conversion processing .
  • the audio digital signal processing module can perform anti-howling processing on the uplink data based on the anti-howling algorithm in this application.
  • the specific algorithm will be described in detail below.
  • the anti-howling processing of the audio digital signal processing module can also use the AEC algorithm in the prior art to eliminate the howling phenomenon.
  • step 102 please refer to step 102, which will not be repeated here.
  • Step 203 The mobile phone sends the processed voice signal to the audio device.
  • the audio digital signal processing module transmits the processed digital voice signal, that is, downlink data to the audio analog-to-digital processing module.
  • the audio analog-to-digital processing module can be based on the current Bluetooth mode, that is, the SCO mode,
  • the voice signal is transmitted to the Bluetooth chip, so that the digital voice signal is transmitted to the audio device through the Bluetooth channel between the mobile phone and the audio device through the Bluetooth chip.
  • Step 204 The audio equipment plays the voice signal.
  • the audio equipment can convert the received digital voice signal into an analog voice signal, and play the analog voice signal, so as to realize the amplification of the user's human voice.
  • the user can adjust the sound reinforcement effect by adjusting the volume of the audio equipment.
  • the essential reasons for howling include: the sound played by the audio equipment is re-collected by the microphone of the mobile phone, and then transmitted to the audio equipment for playback, and repeated back and forth; and the direct echo of the audio equipment directly transmitted to the microphone and the sound broadcast by the audio equipment passes through After the reflection from the outside world, it is transmitted to the microphone, which contains uncertain environmental information such as reflections, as shown in Figure 10.
  • an adaptive echo cancellation algorithm (AEC) is usually used to eliminate howling.
  • AEC adaptive echo cancellation algorithm
  • the mobile phone needs to accurately estimate the echo signal ref in order to accurately suppress the echo signal.
  • different audio equipment refers to audio equipment produced by different manufacturers, or different models of audio equipment produced by the same manufacturer, that is, different audio equipment has different delays, and Different audio equipment has different playback volume, resulting in different distortions that affect the playback of the equipment. Therefore, the existing AEC algorithm can only perform echo cancellation on connected devices with a specified delay, while for connected devices with non-specified delays, due to the limitations of the algorithm, the mobile phone cannot accurately estimate the echo signal ref, and thus cannot accurately eliminate it. Echo to suppress howling.
  • this application proposes an anti-howling algorithm that can ignore the influence of time delay, that is, the anti-howling algorithm in this application can be used for echo cancellation between connected devices with different delays.
  • the echo fed back to the microphone by the audio equipment belongs to the far-field sound, or becomes the far-field signal
  • the user’s voice collected by the microphone that is, the user’s voice signal belongs to the near-field sound, or becomes the near-field sound.
  • the feedback path can be interrupted, thereby suppressing the howling phenomenon caused by the far-field sound.
  • the anti-howling algorithm of this application uses spectral subtraction to eliminate far-field sound and preserve near-field sound.
  • Figure 12 shows the flow diagram of the anti-howling algorithm, which specifically includes:
  • the signals collected by microphone 1 and microphone 2 both include the user's voice signal and interference signal (the concept can be referred to above).
  • the near-field human voice is defined, that is, the signal from the user’s voice signal to the microphone 1 is s1, and the signal from the near-field human voice to the microphone 2 is s2.
  • the interference echo fed back by the audio equipment that is, the interference signal is e
  • the signals collected by the microphone 1 and the microphone 2 are respectively input to the equalizer module, which is used to process the high frequency resonance peaks of the signal x1 and the signal x2.
  • the signal will have a resonance peak of more than 15dB at high frequencies, which will cause distortion between the collected signal and the real signal.
  • the microphone frequency response can be made Smooth, that is, remove the distortion caused by the device in the collected signal.
  • the equalizer module outputs the processed signal, including the signal y1 processed on the signal x1 of the microphone 1, and the signal processed on the signal x2 of the microphone 2 is y2.
  • the equalizer module inputs y1 and y2 to the filter compensation module to filter out the signal in y2 from y1 to obtain the human voice, that is, the user’s voice signal.
  • the filter compensation module Compensate for near-field vocal damage. Specifically, after the signal y1 corresponding to microphone 1 is subtracted from the signal y2 corresponding to microphone 2, the energy of the obtained user's voice signal will be reduced. Therefore, the user's voice can be amplified to compensate for the damage.
  • the filter compensation module inputs the processed signal to the smooth output module, which is used to smooth the volume change caused by the above-mentioned processing process.
  • FIG 13 shows the flow diagram of the anti-blooming wheat algorithm, which specifically includes:
  • the signals collected by microphone 1 and microphone 2 both include the user's voice signal and interference signal (the concept can be referred to above).
  • the wheat spray detection threshold module can be preset with a wheat spray threshold. If the signal strength difference between microphone 1 and microphone 2 is greater than the wheat spray threshold, it is determined that there is a wheat spray phenomenon. On the contrary, if the signal strength of microphone 1 and microphone 2 are different Less than the wheat spraying threshold, there is no wheat spraying phenomenon. Exemplarily, if the wheat spray detection threshold module determines that there is a wheat spray phenomenon, the subsequent step 3) is executed.
  • the filtering module may perform a filtering operation on the main microphone, such as microphone 1, to filter out noise.
  • the signals acquired by the microphone 1 include useful signals, that is, the user's voice signal and interference signals.
  • the signal acquired by microphone 2 includes the user's voice signal and interference signal.
  • microphone 2 is a secondary microphone, and its distance from the user is slightly farther than the distance between microphone 1 and the user. When the microphone spray occurs, it can be considered as the main microphone.
  • the interference signal of the microphone causes this phenomenon. Therefore, the interference signal contained in the signal obtained by the microphone 2 is negligible, that is, the signal obtained by the microphone 2 can be regarded as a useful signal, that is, the user's voice signal.
  • the signal corresponding to the main microphone that is, the signal corresponding to the microphone 1 can be subtracted from the signal corresponding to the microphone 2 to obtain the interference signal of the microphone 1.
  • the filter compensation module can perform filter compensation on the filtered user's voice signal. Specifically, the filter compensation module can filter out the interference signal part of the signal corresponding to microphone 1 based on the interference signal of microphone 1 obtained in 2). The user’s voice signal, and the user’s voice is amplified to compensate for the damage.
  • the filter compensation module inputs the processed signal to the smooth output module, which is used to smooth the volume change caused by the above-mentioned processing process.
  • FIG. 14 is a schematic diagram of an application scenario of an embodiment of the application.
  • the application scenario includes a headset and a mobile phone, where the headset and the mobile phone device establish a communication connection.
  • the headset is a wired headset
  • the earphone and the mobile phone are connected through the earphone cable.
  • the headset is a wireless headset, such as a Bluetooth headset
  • the headset establishes a wireless connection with the mobile phone, such as a Bluetooth connection.
  • the user can speak into the headset microphone, and the mobile phone can collect the user's voice signal through the headset microphone and amplify the voice signal.
  • the scene may be a live broadcast, a lecture, a class, a meeting, or other scenes.
  • a wired headset is taken as an example for description in this embodiment. It should be noted that the numbers of earphones and terminals in FIG. 14 are only illustrative examples. In actual applications, the number of earphones and mobile phones may be one or more, which is not limited in this application.
  • FIG. 15 is a schematic flowchart of a wireless sound amplification method in an embodiment of the application, and in FIG. 15:
  • Step 301 The mobile phone collects the user's voice signal through the headset microphone.
  • the headset can collect the user's voice signal, and the phone acquires the voice signal collected by the headset.
  • the earphone is a wired earphone
  • the earphone and the mobile phone are connected through an earphone cable.
  • the headset is a wireless headset, a Bluetooth connection can be established between the headset and the mobile phone.
  • Step 302 The mobile phone processes the voice signal.
  • Figure 16 is a schematic flow chart of the mobile phone processing the voice signal
  • the mobile phone collects the user’s voice signal through the headset microphone, and combines the user’s voice signal, that is, the uplink data in Figure 16.
  • the audio digital signal processing module After processing by the audio digital signal processing module, the downlink data is obtained.
  • the audio digital signal processing module may process the digital voice signal based on a preset amplification algorithm, and mix the processed digital voice signal with the playback link to obtain downlink data.
  • the audio digital signal processing module is based on the amplification algorithm, and the processing of the digital voice signal may include, but is not limited to: environmental noise reduction and enhancement processing, anti-howling processing, anti-blooming wheat processing, sampling rate and Bit width conversion processing.
  • the headset includes a microphone.
  • the mobile phone microphone is in an on state.
  • the mobile phone microphone can collect the user’s voice signal and interference signal.
  • Figure 17 shows the flow diagram of the anti-howling algorithm in this scenario.
  • the signal collected by the headset microphone includes the user’s voice. Signals and interference signals.
  • the signals collected by the mobile phone microphone include the user’s voice signal and interference signals.
  • the distance between the mobile phone and the user is usually relatively long, so the user’s voice signal collected by the mobile phone microphone can be Negligible, that is to say, in the process of noise detection, the signal collected by the microphone of the mobile phone can be confirmed as an interference signal.
  • resonant peak processing and filtering are performed on the signals collected by the headset microphone, including the user's voice signal and interference signal, and the interference signal collected by the mobile phone microphone.
  • the related concepts of resonant peak processing can be referred to the above, and will not be repeated here.
  • filtering refers to removing the signal collected by the headset microphone from the signal collected by the mobile phone microphone to filter out interference signals and obtain the user's voice signal.
  • filtering compensation processing and smooth output processing can be performed on the obtained user's voice signal. For details, please refer to the above, and will not be repeated here.
  • the headset may include two or more microphones.
  • the mobile phone microphone can be turned on or off, which is not limited in this application.
  • the mobile phone acquires signals collected by multiple microphones of the headset, and the anti-howling processing process of the mobile phone on the acquired signals can refer to scenario 2, which will not be repeated here.
  • the anti-speech algorithm is similar to the anti-howling algorithm.
  • the mobile phone can perform anti-speech processing based on the signals collected by two or more headset microphones.
  • the mobile phone can do its best to prevent spraying wheat based on the signals collected by the headset microphone and the mobile phone microphone.
  • the specific processing process is the same as that of scenario 2, and will not be repeated here.
  • Step 303 The mobile phone outputs the processed voice signal through the speaker.
  • the audio digital signal processing module transmits the processed digital voice signal, that is, downlink data to the audio analog-to-digital processing module, and the audio analog-to-digital processing module performs digital-to-analog conversion, namely ,
  • the digital voice signal is converted into an analog voice signal, and played through the speaker of the mobile phone, so as to realize the amplification of the user's voice signal.
  • the user can adjust the volume of the mobile phone to adjust the sound reinforcement effect of the speaker.
  • FIG 18 is a schematic diagram of an application scenario of an embodiment of the application.
  • the application scenario includes earphones, mobile phones, and audio equipment.
  • the earphones, mobile phones, and audio equipment establish a communication connection.
  • the headset is a wireless headset, such as a Bluetooth headset
  • the headset establishes a wireless connection with the mobile phone, such as a Bluetooth connection.
  • the mobile phone and the audio equipment can establish a wireless connection, such as a Bluetooth connection.
  • the user can speak into the headset microphone, and the mobile phone can collect the user's voice signal through the headset microphone, and amplify the voice signal through the audio equipment.
  • the scene may be a live broadcast, a lecture, a class, a meeting, or other scenes.
  • a wired headset is taken as an example for description in this embodiment.
  • the numbers of earphones, mobile phones and audio equipment in FIG. 18 are only illustrative examples. In actual applications, the number of earphones, mobile phones and audio equipment may be one or more, which is not limited in this application.
  • FIG. 19 is a schematic flowchart of a wireless sound amplification method in an embodiment of the application, and in FIG. 19:
  • Step 401 The mobile phone collects the user's voice signal through the headset microphone.
  • the headset can collect the user's voice signal, and the phone can acquire the voice signal collected by the headset microphone.
  • the mobile phone after the mobile phone detects that the wireless amplification function is turned on, the mobile phone can establish a Bluetooth connection with nearby audio equipment, and the mobile phone sets the Bluetooth mode to the SCO mode.
  • the mobile phone sets the Bluetooth mode to the SCO mode.
  • Step 402 The mobile phone processes the voice signal.
  • Figure 20 is a schematic flow chart of the mobile phone processing voice signals, referring to Figure 20, specifically, the mobile phone collects the user’s voice signal through the headset microphone, and combines the user’s voice signal, that is, the uplink data in Figure 20. (The concept can be referred to above) After processing by the audio digital signal processing module, the downlink data is obtained.
  • the audio digital signal processing module may process the digital voice signal based on a preset sound amplification algorithm, and mix the processed digital voice signal with the playback link to obtain downlink data.
  • step 302 please refer to step 302, which will not be repeated here.
  • Step 403 The mobile phone sends the processed voice signal to the audio device.
  • the audio digital signal processing module transmits the processed digital voice signal, that is, downlink data, to the audio analog-to-digital processing module.
  • the audio analog-to-digital processing module can be based on the current Bluetooth mode, that is, the SCO mode,
  • the voice signal is transmitted to the Bluetooth chip, so that the digital voice signal is transmitted to the audio device through the Bluetooth channel between the mobile phone and the audio device through the Bluetooth chip.
  • Step 404 The audio equipment plays the voice signal.
  • the audio equipment can convert the received digital voice signal into an analog voice signal, and play the analog voice signal, so as to realize the amplification of the user's human voice.
  • the user can adjust the sound reinforcement effect by adjusting the volume of the audio equipment.
  • FIG. 21 is a schematic diagram of an application scenario of an embodiment of the application.
  • the application scenario includes a headset, a mobile phone, and an audio device, where the headset and the mobile phone device establish a communication connection.
  • the headset is a wireless headset, such as a Bluetooth headset, which can establish a wireless connection with a mobile phone, such as a Bluetooth connection.
  • the user can speak into the headset microphone, and the voice signal collected by the headset can be transmitted to the audio equipment to amplify the voice signal through the audio equipment.
  • the scene may be a live broadcast, a lecture, a class, a meeting, or other scenes.
  • the numbers of earphones, mobile phones, and audio equipment in FIG. 21 are only illustrative examples. In actual applications, the number of earphones, mobile phones, and audio equipment may be one or more, which is not limited in this application.
  • FIG. 22 is a schematic flowchart of a wireless sound amplification method in an embodiment of the application.
  • FIG. 22 is a schematic flowchart of a wireless sound amplification method in an embodiment of the application.
  • Step 501 The earphone microphone collects the user's voice signal.
  • the mobile phone can establish a Bluetooth connection with the Bluetooth headset. Specifically, in this embodiment, after the mobile phone turns on the wireless amplification function, the user can speak into the microphone of the Bluetooth headset, and the Bluetooth headset can collect the user's voice signal.
  • the mobile phone can establish a Bluetooth connection with a nearby audio device.
  • the Bluetooth headset establishes a Bluetooth connection with the audio device through the mobile phone, that is, Bluetooth Headphones and audio equipment can be connected through the Bluetooth, or can be called a Bluetooth channel to transmit data.
  • Step 502 the earphone processes the voice signal.
  • the Bluetooth headset may be equipped with the audio digital signal processing module and audio analog-to-digital processing module of the mobile phone in the above-mentioned embodiment, and implement the audio digital signal processing module and audio analog-to-digital processing module of the mobile phone.
  • the Bluetooth headset may be equipped with the audio digital signal processing module and audio analog-to-digital processing module of the mobile phone in the above-mentioned embodiment, and implement the audio digital signal processing module and audio analog-to-digital processing module of the mobile phone.
  • the Bluetooth headset can process the digital voice signal based on a preset amplification algorithm, and mix the processed digital voice signal with the playback link to obtain downlink data .
  • the processing procedure of the voice signal by the Bluetooth headset is the same as the internal processing procedure of the mobile phone in the above embodiment, and will not be repeated here.
  • Step 503 The earphone sends the processed voice signal to the audio device.
  • the earphone can transmit the processed voice signal to the audio device through the Bluetooth chip through the Bluetooth channel between the earphone and the audio device.
  • a Bluetooth headset is provided with a left and right ear transmission protocol, which stipulates that the main ear, such as the right ear of the headset, transmits sound to the secondary ear, such as the left ear of the headset, to realize binaural audio playback.
  • the specific content of the agreement can refer to the existing technology, which will not be repeated in this application.
  • the Bluetooth headset can use the stereo Bluetooth as the auxiliary ear based on the left and right ear protocol, and transmit the acquired downlink data, that is, the digital voice signal, to the audio device.
  • Step 504 The audio device plays the voice signal.
  • the audio equipment can convert the received digital voice signal into an analog voice signal, and play the analog voice signal, so as to realize the amplification of the user's human voice.
  • the user can adjust the sound reinforcement effect by adjusting the volume of the audio equipment.
  • the number of devices in each application scenario may include one or more.
  • FIG. 23 it is a schematic diagram of a multi-device application scenario.
  • These include mobile phone 1 and wired headsets, mobile phone 2 and Bluetooth headsets, and mobile phone 3 and audio equipment.
  • the number of mobile phones, earphones, and audio equipment are only illustrative examples, and this application is not limited.
  • the mobile phone 1, the mobile phone 2, and the mobile phone 3 may establish a wireless connection with the audio equipment through a wireless network to transmit data.
  • the mobile phone 1, the mobile phone 2, and/or the mobile phone 3 obtain the user's voice signal through the headset microphone and/or the mobile phone microphone, the user's voice signal can be processed, and the processed voice signal can be sent to the audio equipment. Amplify the user's voice signal through the audio equipment, so as to realize the multi-user voice amplification in the meeting or event application scenario.
  • the terminal includes hardware structures and/or software modules corresponding to each function.
  • the embodiments of the present application can be implemented in the form of hardware or a combination of hardware and computer software. Whether a certain function is executed by hardware or computer software-driven hardware depends on the specific application and design constraint conditions of the technical solution. Professionals and technicians can use different methods for each specific application to implement the described functions, but such implementation should not be considered beyond the scope of this application.
  • the embodiment of the present application may divide the terminal into functional modules according to the foregoing method examples.
  • each functional module may be divided corresponding to each function, or two or more functions may be integrated into one processing module.
  • the above-mentioned integrated modules can be implemented in the form of hardware or software functional modules. It should be noted that the division of modules in the embodiments of the present application is illustrative, and is only a logical function division, and there may be other division methods in actual implementation.
  • FIG. 24 is a schematic structural diagram of the terminal 200.
  • the terminal 200 includes an acquisition module 201, a processing module 202, and a sending module 203.
  • the acquiring module 201 is configured to acquire the first voice signal collected by the first microphone of the terminal and the second voice signal collected by the second microphone;
  • the processing module 202 is configured to respond to the acquired first voice signal and
  • the second voice signal is used to process the first voice signal and the second voice signal to obtain a playback voice signal, and the sending module 203 is used to transmit the playback voice signal to a loudspeaker device, and the loudspeaker device is wirelessly connected to the terminal. Connected external devices.
  • the processing module 202 is specifically configured to perform analog-to-digital conversion on the first voice signal in response to the acquired first voice signal and the second voice signal to obtain the first digital voice signal, and Perform analog-to-digital conversion of the second voice signal to obtain a second digital voice signal, and combine the first digital voice signal and the second digital voice signal, where the first voice signal and the second voice signal are analog signals; based on a preset sound amplification algorithm , Perform sound effect processing on the first digital voice signal and the second digital voice signal to obtain a third digital voice signal; perform mixing processing on the third digital voice signal to obtain a playback voice signal.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm, and an anti-howling algorithm.
  • the algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third digital voice signal.
  • a voice signal wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal
  • the preset sound reinforcement algorithm includes an anti-speech algorithm
  • the anti-speech algorithm specifically includes : If the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter the interference from the first voice signal Signal.
  • FIG. 25 is a schematic structural diagram of the terminal 300.
  • the terminal 300 includes an acquisition module 301, a processing module 302, and a speaker module 303.
  • the acquisition module 301 is configured to acquire the first voice signal collected by the first microphone of the earphone and the second voice signal collected by the second microphone of the earphone, and the earphone is connected to the terminal in a wired or wireless connection;
  • the processing module 302 Used to process the first voice signal and the second voice signal in response to the acquired first voice signal and the second voice signal to obtain the played voice signal, and the speaker module 303 is used to amplify the played voice signal.
  • the processing module 302 is specifically configured to perform analog-to-digital conversion on the first voice signal in response to the acquired first voice signal and second voice signal to obtain the first digital voice signal, and
  • the second voice signal is converted from analog to digital to obtain the second digital voice signal, where the first voice signal and the second voice signal are analog signals; based on the preset sound amplification algorithm, the first digital voice signal and the second digital voice signal are performed Sound effect processing to obtain a third digital voice signal; perform mixing processing on the third digital voice signal to obtain a fourth digital voice signal; perform analog-to-digital conversion on the fourth digital voice signal to obtain a playback voice signal.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm and an anti-howling algorithm.
  • the algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third digital voice signal.
  • a voice signal wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm specifically includes : If the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter the interference from the first voice signal Signal.
  • FIG. 26 is a schematic structural diagram of the terminal 400.
  • the terminal 400 includes: an acquisition module 401, a processing module 402, and a sending module 403.
  • the acquisition module 401 is configured to acquire the first voice signal collected by the first microphone of the headset and the second voice signal collected by the second microphone of the headset, and the headset is connected to the terminal by wire or wirelessly;
  • the processing module 402 Used to process the first voice signal and the second voice signal in response to the acquired first voice signal and the second voice signal to obtain the played voice signal, and the sending module 403 is used to transmit the played voice signal to the PA A device, the loudspeaker device is an external device wirelessly connected to the terminal.
  • the processing module 402 is specifically configured to perform analog-to-digital conversion on the first voice signal in response to the acquired first voice signal and the second voice signal to obtain the first digital voice signal, and
  • the second voice signal undergoes analog-to-digital conversion to obtain a second digital voice signal, and combines the first digital voice signal and the second digital voice signal, where the first voice signal and the second voice signal are analog signals; based on preset amplification
  • the algorithm performs sound effect processing on the first digital voice signal and the second digital voice signal to obtain a third digital voice signal; and performs mixing processing on the third digital voice signal to obtain a playback voice signal.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm, and an anti-howling algorithm.
  • the algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third digital voice signal.
  • a voice signal wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal
  • the preset sound reinforcement algorithm includes an anti-speech algorithm
  • the anti-speech algorithm specifically includes : If the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter the interference from the first voice signal Signal.
  • Fig. 27 is a schematic diagram of the terminal 500.
  • the terminal 500 includes: an acquisition module 501, a processing module 502, and a sending module 503.
  • the acquisition module 501 is used for collecting voice signals
  • the processing module 502 is used for In response to the acquired voice signal, the voice signal is processed to obtain the played voice signal
  • the sending module 503 is used to transmit the played voice signal to the amplifying device.
  • the sound pickup device is a microphone on an earphone connected to the terminal in a wired or wireless connection, and includes a first microphone and a second microphone.
  • the sound pickup device is a microphone with the terminal, including a first microphone and a second microphone.
  • the voice signal includes a first voice signal and a second voice signal
  • the acquisition module 501 is configured to collect the first voice signal through the first microphone and collect the second voice signal through the second microphone;
  • processing The module 502 is configured to process the first voice signal and the second voice signal in response to the acquired first voice signal and the second voice signal to obtain a played voice signal, and transmit the played voice signal to the amplification device.
  • the loudspeaker device is the speaker of the terminal;
  • the processing module 502 is specifically configured to perform analog-to-digital conversion on the first voice signal in response to the acquired first voice signal and the second voice signal to obtain the first voice signal.
  • Digital voice signal, and analog-to-digital conversion is performed on the second voice signal to obtain the second digital voice signal, wherein the first voice signal and the second voice signal are analog signals; based on the preset sound amplification algorithm, the first digital voice signal
  • the signal and the second digital voice signal are processed by sound effects to obtain the third digital voice signal; the third digital voice signal is mixed and processed to obtain the fourth digital voice signal; the fourth digital voice signal is converted from analog to digital to obtain the playback voice signal.
  • the loudspeaker device is a loudspeaker device wirelessly connected to the terminal; the processing module 502 is specifically configured to perform analog-to-digital processing on the first voice signal in response to the acquired first voice signal and second voice signal.
  • the voice signal is an analog signal; based on the preset sound amplification algorithm, perform sound effect processing on the first digital voice signal and the second digital voice signal to obtain a third digital voice signal; perform mixing processing on the third digital voice signal to obtain The voice signal is played, and the acquired voice signal is transmitted to the amplifying device.
  • the first voice signal includes a first human voice signal and an interference signal
  • the second voice signal includes a second human voice signal and an interference signal
  • the preset sound reinforcement algorithm includes an anti-howling algorithm, and an anti-howling algorithm.
  • the algorithm specifically includes: filtering the second digital voice signal corresponding to the second voice signal from the first digital voice signal corresponding to the first voice signal, and amplifying the filtered first digital voice signal to obtain the third digital voice signal.
  • a voice signal wherein the signal strength of the first voice signal is greater than the signal strength of the second voice signal.
  • the first voice signal includes the first human voice signal and the interference signal
  • the second voice signal includes the second human voice signal
  • the preset sound reinforcement algorithm includes the anti-speech algorithm
  • the anti-speech algorithm specifically includes : If the signal strength difference between the first voice signal and the second voice signal is greater than the spraying threshold, based on the second human voice signal, determine the interference signal in the first voice signal; filter the interference from the first voice signal Signal.
  • FIG. 28 shows a schematic block diagram of a terminal 600 according to an embodiment of the present application.
  • the terminal may include a processor 601 and a transceiver/transceiving pin 602, and optionally, a memory 603.
  • the processor 601 may be used to execute the steps performed by the terminal in each method of the foregoing embodiments, and control the receiving pin to receive signals, and control the sending pin to send signals.
  • bus 604. In addition to the data bus, the bus system 604 also includes a power bus, a control bus, and a status signal bus. However, for clear description, various buses are marked as the bus system 604 in the figure.
  • the memory 603 may be used for storing instructions in the foregoing method embodiments.
  • the terminal 600 may correspond to the terminal in the various methods of the foregoing embodiments, and the above-mentioned and other management operations and/or functions of the various elements in the terminal 600 are used to implement the corresponding methods of the foregoing various methods. The steps are not repeated here for the sake of brevity.
  • the embodiments of the present application also provide a computer-readable storage medium, the computer-readable storage medium stores a computer program, and the computer program contains at least a piece of code that can be executed by the terminal to control the terminal Used to implement the above method embodiments.
  • the embodiments of the present application also provide a computer program, which is used to implement the foregoing method embodiments when the computer program is executed by the terminal.
  • the program may be stored in whole or in part on a storage medium packaged with the processor, and may also be stored in part or in a memory not packaged with the processor.
  • an embodiment of the present application further provides a processor, which is configured to implement the foregoing method embodiments.
  • the above-mentioned processor may be a chip.
  • the steps of the method or algorithm described in conjunction with the disclosure of the embodiments of the present application may be implemented in a hardware manner, or may be implemented in a manner in which a processor executes software instructions.
  • Software instructions can be composed of corresponding software modules, which can be stored in random access memory (Random Access Memory, RAM), flash memory, read-only memory (Read Only Memory, ROM), and erasable programmable read-only memory ( Erasable Programmable ROM (EPROM), Electrically Erasable Programmable Read-Only Memory (Electrically EPROM, EEPROM), register, hard disk, mobile hard disk, CD-ROM or any other form of storage medium known in the art.
  • RAM Random Access Memory
  • ROM read-only memory
  • EPROM Erasable Programmable ROM
  • EPROM Electrically Erasable Programmable Read-Only Memory
  • register hard disk, mobile hard disk, CD-ROM or any other form of storage medium known in the art.
  • An exemplary storage medium is coupled to the processor, so that the processor can read information from the storage medium and write information to the storage medium.
  • the storage medium may also be an integral part of the processor.
  • the processor and the storage medium may be located in the ASIC.
  • the ASIC may be located in a network device.
  • the processor and the storage medium may also exist as discrete components in the network device.
  • the functions described in the embodiments of the present application may be implemented by hardware, software, firmware, or any combination thereof. When implemented by software, these functions can be stored in a computer-readable medium or transmitted as one or more instructions or codes on the computer-readable medium.
  • the computer-readable medium includes a computer storage medium and a communication medium, where the communication medium includes any medium that facilitates the transfer of a computer program from one place to another.
  • the storage medium may be any available medium that can be accessed by a general-purpose or special-purpose computer.

Abstract

本申请实施例提供了一种无线扩音系统及终端,该系统包括终端与扩音设备,终端与扩音设备无线连接,终端包括音频模块、第一麦克风和第二麦克风;第一麦克风,用于采集第一语音信号,并将第一语音信号传输至音频模块;第二麦克风,用于采集第二语音信号,并将第二语音信号传输至音频模块;音频模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,并将播放语音信号传输至扩音设备;扩音设备,用于响应于获取到的播放语音信号,对播放语音信号进行扩音。本申请能够降低扩音时延对扩音效果的影响,并且,用户仅通过手机与扩音设备即可实现对人声的扩音,从而提供一种便于携带的扩音系统。

Description

无线扩音系统及终端
本申请要求于2020年2月28日提交中国专利局、申请号为202010131051.3、申请名称为“无线扩音系统及终端”的中国专利申请的优先权,其全部内容通过引用结合在本申请中。
技术领域
本申请实施例涉及通信领域,尤其涉及一种无线扩音系统及终端。
背景技术
在日常生活中,人们经常会遇到需要进行扩音的场景,例如:较为嘈杂场景、会议场景或上课场景等。
目前,已有技术的扩音方案通常是用户携带扩音设备,扩音设备包括麦克风和音响设备,如图1所示。但是,在一些偶发性的场景,例如会议场景等,对于偶尔有扩音需求的用户,随身携带一套扩音设备,不够便捷。
发明内容
本申请提供一种无线扩音系统、方法及终端,能够有效降低扩音时延,并方便携带。
为达到上述目的,本申请采用如下技术方案:
第一方面,本申请实施例提供一种无线扩音系统,该系统包括终端与扩音设备,其中,终端与扩音设备无线连接,该终端包括音频模块、第一麦克风和第二麦克风;具体的,第一麦克风,用于采集用户说话时对应的第一语音信号,并将第一语音信号传输至音频模块;第二麦克风,用于采集该用户说话时对应的第二语音信号,并将第二语音信号传输至音频模块;可以理解为,第一麦克风和第二麦克风同时采集用户说话的人声,但是,由于第一麦克风和第二麦克风与用户的距离不同,其采集到的语音信号也不相同。音频模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,并将获取到的播放语音信号传输至扩音设备;扩音设备,用于响应于获取到的播放语音信号,对播放语音信号进行扩音。
基于上述方式,实现了终端可在处理器层,具体是音频模块对采集到的语音信号进行处理,从而实现低时延环回路径,以降低扩音时延对扩音效果的影响,并且,用户仅通过手机与扩音设备即可实现对人声的扩音,从而提供一种便于携带的扩音系统,以使用户在会议、直播、上课等场景中,通过手机连接扩音设备,即可实现扩音。
在一种可能的实现方式中,音频模块包括音频模数处理模块和音频数字信号处理模块;具体的,音频模数处理模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号传输至音 频数字信号处理模块,其中,第一语音信号和第二语音信号为模拟信号;音频数字信号处理模块,用于响应于获取到的第一数字语音信号和第二数字语音信号,基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到播放语音信号,以及,将播放语音信号传输至音频模数处理模块;音频模数处理模块,还用于将获取到的播放语音信号传输至扩音设备。
基于上述方式,实现了终端的硬件层面,具体为处理器中的音频模块内部对采集到的语音信号的处理,从而实现一种低时延的换回方式,以降低扩音时延对扩音效果的影响。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
基于上述方式,实现了扩音过程中的抗啸叫处理,通过将第一麦克风采集到的语音信号滤除掉第二麦克风采集到的语音信号部分,以得到用户的人声对应的语音信号,从而提高滤除干扰信号的准确度。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
基于上述方式,实现了扩音过程中的防喷麦处理,通过第二麦克风采集到的语音信号对第一麦克风采集到的语音信号中的干扰信号进行滤除,从而消除喷麦现象。
第二方面,本申请实施例提供了一种无线扩音系统,该系统包括耳机与终端,耳机与扩音设备有线连接或无线连接,终端包括音频模块和扬声器,耳机包括第一麦克风和第二麦克风;第一麦克风,用于采集第一语音信号,并将第一语音信号传输至音频模块;第二麦克风,用于采集第二语音信号,并将第二语音信号传输至音频模块;音频模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,并将播放语音信号传输至扬声器;扬声器,用于响应于获取到的播放语音信号,对播放语音信号进行扩音。
基于上述方式,实现了终端可在处理器层,具体是音频模块对采集到的语音信号进行处理,从而实现低时延环回路径,以降低扩音时延对扩音效果的影响,并且,用户仅通过手机与耳机连接,即可实现对人声的扩音,从而提供一种便于携带的扩音系统,以使用户在会议、直播、上课等场景中,用户可通过耳机和手机,即可实现扩音。
在一种可能的实现方式中,音频模块包括音频模数处理模块和音频数字信号处理模块;音频模数处理模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号传输至音频数字信号处理模块,其中,第一语音信号和第二语音信号为模拟信号;音频数字信号处理模块,用于响应于获取到的第一数字语音信号和第二数字语音信号,基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到第四数字语音信号,以及,将第四数字语音信号传输至音频模数处理模块;音频模数处理模块,还用于响应于获取到的第四数字语音信号,对第四数字语音信号进行模数转换,获取播放语音信号,并将播放语音信号传输至扬声器;其中,播放语音信号为模拟信号。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
第三方面,本申请实施例提供了一种无线扩音系统,其特征在于,包括耳机、手机和扩音设备,耳机与扩音设备有线连接或无线连接,手机与扩音设备无线连接,终端包括音频模块,耳机包括第一麦克风和第二麦克风;第一麦克风,用于采集第一语音信号,并将第一语音信号传输至音频模块;第二麦克风,用于采集第二语音信号,并将第二语音信号传输至音频模块;音频模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,并将播放语音信号传输至扩音设备;扩音设备,用于响应于获取到的播放语音信号,对播放语音信号进行扩音。
基于上述方式,实现了终端可在处理器层,具体是音频模块对采集到的语音信号进行处理,从而实现低时延环回路径,以降低扩音时延对扩音效果的影响,并且,用户仅通过手机与耳机连接,即可实现对人声的扩音,从而提供一种便于携带的扩音系统,以使用户在会议、直播、上课等场景中,用户可通过耳机和手机,即可实现扩音。
在一种可能的实现方式中,音频模块包括音频模数处理模块和音频数字信号处理模块;音频模数处理模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换, 得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号传输至音频数字信号处理模块,其中,第一语音信号和第二语音信号为模拟信号;音频数字信号处理模块,用于响应于获取到的第一数字语音信号和第二数字语音信号,基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到播放语音信号,以及,将播放语音信号传输至音频模数处理模块;音频模数处理模块,还用于将获取到的播放语音信号传输至扩音设备。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
第四方面,本申请实施例提供了一种终端该终端包括音频模块和拾音设备;拾音设备,用于采集语音信号,并将语音信号传输至音频模块;音频模块,用于响应于获取到的语音信号,对语音信号进行处理,得到播放语音信号,并将播放语音信号传输至扩音设备。
基于上述方式,实现了终端可在处理器层,具体是音频模块对采集到的语音信号进行处理,从而实现低时延环回路径,以降低扩音时延对扩音效果的影响,并且,用户仅通过手机与耳机连接,即可实现对人声的扩音,从而提供一种便于携带的扩音系统,以使用户在会议、直播、上课等场景中,用户可通过耳机和手机,即可实现扩音。
在一种可能的实现方式中,拾音设备为与终端有线连接或无线连接的耳机上的麦克风,包括第一麦克风和第二麦克风。
在一种可能的实现方式中,拾音设备为与终端的麦克风,包括第一麦克风和第二麦克风。
在一种可能的实现方式中,语音信号包括第一语音信号与第二语音信号;第一麦克风,用于采集第一语音信号,并将第一语音信号传输至音频模块;第二麦克风,用于采集第二语音信号,并将第二语音信号传输至音频模块;第一语音信号和第二语音信号属于语音信号;音频模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语 音信号和第二语音信号进行处理,得到播放语音信号,并将播放语音信号传输至扩音设备。
在一种可能的实现方式中,扩音设备为终端的扬声器;音频模块包括音频模数处理模块和音频数字信号处理模块;音频模数处理模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号传输至音频数字信号处理模块,其中,第一语音信号和第二语音信号为模拟信号;音频数字信号处理模块,用于响应于获取到的第一数字语音信号和第二数字语音信号,基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到第四数字语音信号,以及,将第四数字语音信号传输至音频模数处理模块;音频模数处理模块,还用于响应于获取到的第四数字语音信号,对第四数字语音信号进行模数转换,获取播放语音信号,并将播放语音信号传输至扬声器;其中,播放语音信号为模拟信号。
在一种可能的实现方式中,扩音设备为与终端无线连接的扩音设备;音频模数处理模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号传输至音频数字信号处理模块,其中,第一语音信号和第二语音信号为模拟信号;音频数字信号处理模块,用于响应于获取到的第一数字语音信号和第二数字语音信号,基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到播放语音信号,以及,将播放语音信号传输至音频模数处理模块;音频模数处理模块,还用于将获取到的播放语音信号传输至扩音设备。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
第五方面,本申请实施例提供了一种无线扩音方法,该方法包括应用于终端,包括:获取终端的第一麦克风采集到的第一语音信号,以及第二麦克风采集到的第二语音信号; 响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,并将播放语音信号传输至扩音设备,所述扩音设备为与所述终端无线连接的外接设备。
在一种可能的实现方式中,对第一语音信号和第二语音信号进行处理,包括:响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到播放语音信号,将获取到的播放语音信号传输至扩音设备。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
第六方面,本申请实施例提供了一种无线扩音方法,该方法应用于终端,包括:获取耳机的第一麦克风采集到的第一语音信号,以及耳机的第二麦克风采集到的第二语音信号,该耳机与终端有线连接或无线连接;响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,并通过扬声器对播放语音信号进行扩音。
在一种可能的实现方式中,对第一语音信号和第二语音信号进行处理,包括:响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到第四数字语音信号;对第四数字语音信号进行模数转换,获取播放语音信号。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音 信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
第七方面,本申请实施例提供了一种无线扩音方法,该方法应用于终端,包括:获取耳机的第一麦克风采集到的第一语音信号,以及耳机的第二麦克风采集到的第二语音信号,该耳机与终端有线连接或无线连接;响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,并将播放语音信号传输至扩音设备,所述扩音设备为与所述终端无线连接的外接设备。
在一种可能的实现方式中,对第一语音信号和第二语音信号进行处理,包括:响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到播放语音信号,将获取到的播放语音信号传输至扩音设备。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
第八方面,本申请实施例提供了一种无线扩音方法,该方法应用于终端,包括:采集语音信号,响应于获取到的语音信号,对语音信号进行处理,得到播放语音信号,并将播放语音信号传输至扩音设备。
在一种可能的实现方式中,拾音设备为与终端有线连接或无线连接的耳机上的麦克风,包括第一麦克风和第二麦克风。
在一种可能的实现方式中,拾音设备为与终端的麦克风,包括第一麦克风和第二麦克风。
在一种可能的实现方式中,语音信号包括第一语音信号与第二语音信号;采集语音信号包括:通过第一麦克风采集第一语音信号,以及,通过第二麦克风采集第二语音信号;响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,并将播放语音信号传输至扩音设备。
在一种可能的实现方式中,扩音设备为终端的扬声器;对第一语音信号和第二语音信号进行处理,包括:响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到第四数字语音信号;对第四数字语音信号进行模数转换,获取播放语音信号。
在一种可能的实现方式中,扩音设备为与终端无线连接的扩音设备;对第一语音信号和第二语音信号进行处理,包括:响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到播放语音信号,将获取到的播放语音信号传输至扩音设备。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
第九方面,本申请实施例提供了一种终端,包括获取模块、处理模块和发送模块。具体的,获取模块用于获取终端的第一麦克风采集到的第一语音信号,以及第二麦克风采集到的第二语音信号;处理模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,发送模块,用于将播放语音信号传输至扩音设备,所述扩音设备为与所述终端无线连接的外接设备。
在一种可能的实现方式中,处理模块具体用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到播放语音信号,将获取到的播放语音信号传输至扩音设备。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
第十方面,本申请实施例提供了一种终端,包括获取模块、处理模块、扬声器模块。具体的,获取模块,用于获取耳机的第一麦克风采集到的第一语音信号,以及耳机的第二麦克风采集到的第二语音信号,该耳机与终端有线连接或无线连接;处理模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,扬声器模块,用于对播放语音信号进行扩音。
在一种可能的实现方式中,处理模块具体用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到第四数字语音信号;对第四数字语音信号进行模数转换,获取播放语音信号。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
第十一方面,本申请实施例提供了终端,包括:获取模块、处理模块和发送模块。具体的,获取模块,用于获取耳机的第一麦克风采集到的第一语音信号,以及耳机的第二麦克风采集到的第二语音信号,该耳机与终端有线连接或无线连接;处理模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,发送模块,用于将播放语音信号传输至扩音设备,所述扩音设备为与所述终端无线连接的外接设备。
在一种可能的实现方式中,处理模块,具体用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到播放语音信号,将获取到的播放语音信号传输至扩音设备。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
第十二方面,本申请实施例提供了一种终端,包括:获取模块、处理模块和发送模块,具体的,获取模块,用于采集语音信号,处理模块,用于响应于获取到的语音信号, 对语音信号进行处理,得到播放语音信号,发送模块,用于将播放语音信号传输至扩音设备。
在一种可能的实现方式中,拾音设备为与终端有线连接或无线连接的耳机上的麦克风,包括第一麦克风和第二麦克风。
在一种可能的实现方式中,拾音设备为与终端的麦克风,包括第一麦克风和第二麦克风。
在一种可能的实现方式中,语音信号包括第一语音信号与第二语音信号;获取模块,用于通过第一麦克风采集第一语音信号,以及,通过第二麦克风采集第二语音信号;处理模块,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,并将播放语音信号传输至扩音设备。
在一种可能的实现方式中,扩音设备为终端的扬声器;处理模块具体用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到第四数字语音信号;对第四数字语音信号进行模数转换,获取播放语音信号。
在一种可能的实现方式中,扩音设备为与终端无线连接的扩音设备;处理模块具体用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到播放语音信号,将获取到的播放语音信号传输至扩音设备。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在一种可能的实现方式中,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确 定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
第十三方面,本申请实施例提供了一种计算机可读介质,用于存储计算机程序,该计算机程序包括用于执行第五方面或第五方面的任意可能的实现方式中的方法的指令。
第十四方面,本申请实施例提供了一种计算机可读介质,用于存储计算机程序,该计算机程序包括用于执行第六方面或第六方面的任意可能的实现方式中的方法的指令。
第十五方面,本申请实施例提供了一种计算机可读介质,用于存储计算机程序,该计算机程序包括用于执行第七方面或第七方面的任意可能的实现方式中的方法的指令。
第十六方面,本申请实施例提供了一种计算机可读介质,用于存储计算机程序,该计算机程序包括用于执行第八方面或第八方面的任意可能的实现方式中的方法的指令。
第十七方面,本申请实施例提供了一种计算机程序,该计算机程序包括用于执行第五方面或第五方面的任意可能的实现方式中的方法的指令。
第十八方面,本申请实施例提供了一种计算机程序,该计算机程序包括用于执行第六方面或第六方面的任意可能的实现方式中的方法的指令。
第十九方面,本申请实施例提供了一种计算机程序,该计算机程序包括用于执行第七方面或第七方面的任意可能的实现方式中的方法的指令。
第二十方面,本申请实施例提供了一种计算机程序,该计算机程序包括用于执行第八方面或第八方面的任意可能的实现方式中的方法的指令。
第二十一方面,本申请实施例提供了一种芯片,该芯片包括处理电路、收发管脚。其中,该收发管脚、和该处理电路通过内部连接通路互相通信,该处理电路执行第五方面或第五方面的任一种可能的实现方式中的方法,以控制接收管脚接收信号,以控制发送管脚发送信号。
第二十二方面,本申请实施例提供了一种芯片,该芯片包括处理电路、收发管脚。其中,该收发管脚、和该处理电路通过内部连接通路互相通信,该处理电路执行第六方面或第六方面的任一种可能的实现方式中的方法,以控制接收管脚接收信号,以控制发送管脚发送信号。
第二十三方面,本申请实施例提供了一种芯片,该芯片包括处理电路、收发管脚。其中,该收发管脚、和该处理电路通过内部连接通路互相通信,该处理电路执行第七方面或第七方面的任一种可能的实现方式中的方法,以控制接收管脚接收信号,以控制发送管脚发送信号。
第二十四方面,本申请实施例提供了一种芯片,该芯片包括处理电路、收发管脚。其中,该收发管脚、和该处理电路通过内部连接通路互相通信,该处理电路执行第八方面或第八方面的任一种可能的实现方式中的方法,以控制接收管脚接收信号,以控制发送管脚发送信号。
附图说明
为了更清楚地说明本申请实施例的技术方案,下面将对本申请实施例的描述中所需要使用的附图作简单地介绍,显而易见地,下面描述中的附图仅仅是本申请的一些实施例,对于本领域普通技术人员来讲,在不付出创造性劳动性的前提下,还可以根据这些附图获得其他的附图。
图1是示例性示出的一种扩音设备的示意图;
图2是示例性示出的一种终端的结构示意图;
图3a是示例性示出的无线扩音功能界面示意图之一;
图3b是示例性示出的无线扩音功能界面示意图之一;
图4是示例性示出的一种应用场景示意图之一;
图5是本申请实施利提供的一种无线扩音方法的流程示意图之一;
图6是示例性示出的一种手机内部处理流程示意图之一;
图7是示例性示出的一种应用场景示意图之一;
图8是本申请实施利提供的一种无线扩音方法的流程示意图之一;
图9是示例性示出的一种手机内部处理流程示意图之一;
图10是示例性示出的回声产生原理的示意图;
图11是示例性示出的抗啸叫算法原理示意图;
图12是本申请实施利提供的一种抗啸叫算法的流程示意图;
图13是本申请实施利提供的一种防喷麦算法的流程示意图;
图14是示例性示出的一种应用场景示意图之一;
图15是本申请实施利提供的一种无线扩音方法的流程示意图之一;
图16是示例性示出的一种手机内部处理流程示意图之一;
图17是本申请实施利提供的一种抗啸叫算法的流程示意图;
图18是示例性示出的一种应用场景示意图之一;
图19是本申请实施利提供的一种无线扩音方法的流程示意图之一;
图20是示例性示出的一种手机内部处理流程示意图之一;
图21是示例性示出的一种应用场景示意图之一;
图22是本申请实施利提供的一种无线扩音方法的流程示意图之一;
图23是示例性示出的一种应用场景示意图之一;
图24是本申请实施利提供的一种终端的结构示意图之一;
图25是本申请实施利提供的一种终端的结构示意图之一;
图26是本申请实施利提供的一种终端的结构示意图之一;
图27是本申请实施利提供的一种终端的结构示意图之一;
图28是本申请实施利提供的一种终端的结构示意图之一。
具体实施方式
下面将结合本申请实施例中的附图,对本申请实施例中的技术方案进行清楚、完整地描述,显然,所描述的实施例是本申请一部分实施例,而不是全部的实施例。基于本申请中的实施例,本领域普通技术人员在没有作出创造性劳动前提下所获得的所有其他 实施例,都属于本申请保护的范围。
本文中术语“和/或”,仅仅是一种描述关联对象的关联关系,表示可以存在三种关系,例如,A和/或B,可以表示:单独存在A,同时存在A和B,单独存在B这三种情况。
本申请实施例的说明书和权利要求书中的术语“第一”和“第二”等是用于区别不同的对象,而不是用于描述对象的特定顺序。例如,第一目标对象和第二目标对象等是用于区别不同的目标对象,而不是用于描述目标对象的特定顺序。
在本申请实施例中,“示例性的”或者“例如”等词用于表示作例子、例证或说明。本申请实施例中被描述为“示例性的”或者“例如”的任何实施例或设计方案不应被解释为比其它实施例或设计方案更优选或更具优势。确切而言,使用“示例性的”或者“例如”等词旨在以具体方式呈现相关概念。
在本申请实施例的描述中,除非另有说明,“多个”的含义是指两个或两个以上。例如,多个处理单元是指两个或两个以上的处理单元;多个系统是指两个或两个以上的系统。
具体的,在本申请中所涉及的终端可以为如移动电话、智能电话、笔记本电脑、平板电脑(personal computer,PC)等等配置有麦克风的终端设备。
示例性的,图2示出了终端为手机时的结构示意图。手机100可以包括处理器110,外部存储器接口120,内部存储器121,通用串行总线(universal serial bus,USB)接口130,充电管理模块140,电源管理模块141,电池142,天线1,天线2,移动通信模块150,无线通信模块160,音频模块170,扬声器170A,受话器170B,麦克风170C,耳机接口170D,传感器模块180,按键190,马达191,指示器192,摄像头193,显示屏194,以及用户标识模块(subscriber identification module,SIM)卡接口195等。其中传感器模块180可以包括压力传感器180A,陀螺仪传感器180B,气压传感器180C,磁传感器180D,加速度传感器180E,距离传感器180F,接近光传感器180G,指纹传感器180H,温度传感器180J,触摸传感器180K,环境光传感器180L,骨传导传感器180M等。
可以理解的是,本申请实施例示意的结构并不构成对手机100的具体限定。在本申请另一些实施例中,手机100可以包括比图示更多或更少的部件,或者组合某些部件,或者拆分某些部件,或者不同的部件布置。图示的部件可以以硬件,软件或软件和硬件的组合实现。
处理器110可以包括一个或多个处理单元,例如:处理器110可以包括应用处理器(application processor,AP),调制解调处理器,图形处理器(graphics processing unit,GPU),图像信号处理器(image signal processor,ISP),控制器,存储器,视频编解码器,数字信号处理器(digital signal processor,DSP),基带处理器,和/或神经网络处理器(neural-network processing unit,NPU)等。其中,不同的处理单元可以是独立的器件,也可以集成在一个或多个处理器中。
其中,控制器可以是手机100的神经中枢和指挥中心。控制器可以根据指令操作码和时序信号,产生操作控制信号,完成取指令和执行指令的控制。
处理器110中还可以设置存储器,用于存储指令和数据。在一些实施例中,处理器 110中的存储器为高速缓冲存储器。该存储器可以保存处理器110刚用过或循环使用的指令或数据。如果处理器110需要再次使用该指令或数据,可从所述存储器中直接调用。避免了重复存取,减少了处理器110的等待时间,因而提高了系统的效率。
在一些实施例中,处理器110可以包括一个或多个接口。接口可以包括集成电路(inter-integrated circuit,I2C)接口,集成电路内置音频(inter-integrated circuit sound,I2S)接口,脉冲编码调制(pulse code modulation,PCM)接口,通用异步收发传输器(universal asynchronous receiver/transmitter,UART)接口,移动产业处理器接口(mobile industry processor interface,MIPI),通用输入输出(general-purpose input/output,GPIO)接口,用户标识模块(subscriber identity module,SIM)接口,和/或通用串行总线(universal serial bus,USB)接口等。
USB接口130是符合USB标准规范的接口,USB接口130可以用于连接充电器为手机100充电,也可以用于手机100与外围设备之间传输数据。也可以用于连接耳机,例如本申请实施例中的有线耳机可通过USB接口连接手机。
可以理解的是,本申请实施例示意的各模块间的接口连接关系,只是示意性说明,并不构成对手机100的结构限定。在本申请另一些实施例中,手机100也可以采用上述实施例中不同的接口连接方式,或多种接口连接方式的组合。
充电管理模块140用于从充电器接收充电输入。电源管理模块141用于连接电池142,充电管理模块140与处理器110。电源管理模块141接收电池142和/或充电管理模块140的输入,为内部器件供电。
手机100的无线通信功能可以通过天线1,天线2,移动通信模块150,无线通信模块160,调制解调处理器以及基带处理器等实现。
手机100可以通过音频模块170,扬声器170A,受话器170B,麦克风170C,耳机接口170D,以及应用处理器等实现音频功能。例如音乐播放,录音等。
音频模块170用于将数字音频信息转换成模拟音频信号输出,也用于将模拟音频输入转换为数字音频信号。音频模块170还可以用于对音频信号编码和解码。在一些实施例中,音频模块170可以设置于处理器110中,或将音频模块170的部分功能模块设置于处理器110中。
扬声器170A,也称“喇叭”,用于将音频电信号转换为声音信号。手机100可以通过扬声器170A收听音乐,或收听免提通话。
受话器170B,也称“听筒”,用于将音频电信号转换成声音信号。当手机100接听电话或语音信息时,可以通过将受话器170B靠近人耳接听语音。
麦克风170C,也称“话筒”,“传声器”,用于将声音信号转换为电信号。当拨打电话或发送语音信息时,用户可以通过人嘴靠近麦克风170C发声,将声音信号输入到麦克风170C。手机100可以设置一个或者多个麦克风170C。在另一些实施例中,手机100可以设置两个麦克风170C,除了采集声音信号,还可以实现降噪功能。在另一些实施例中,手机100还可以设置三个,四个或更多麦克风170C,实现采集声音信号,降噪,还可以识别声音来源,实现定向录音功能等。
耳机接口170D用于连接有线耳机。耳机接口170D可以是USB接口130,也可以是 3.5mm的开放移动手机平台(open mobile terminal platform,OMTP)标准接口,美国蜂窝电信工业协会(cellular telecommunications industry association of the USA,CTIA)标准接口。
手机100通过GPU,显示屏194,以及应用处理器等实现显示功能。显示屏194用于显示图像,视频等,例如可用于显示本申请中的无线扩音设置界面。
手机100可以通过ISP,摄像头193,视频编解码器,GPU,显示屏194以及应用处理器等实现拍摄功能。
外部存储器接口120可以用于连接外部存储卡,例如Micro SD卡,实现扩展手机100的存储能力。
内部存储器121可以用于存储计算机可执行程序代码,所述可执行程序代码包括指令。处理器110通过运行存储在内部存储器121的指令,从而执行手机100的各种功能应用以及数据处理。内部存储器121可以包括存储程序区和存储数据区。其中,存储程序区可存储操作系统,一个或者多个功能所需的应用程序(比如声音播放功能,图像播放功能等)等。存储数据区可存储手机100使用过程中所创建的数据(比如音频数据,电话本等)等。此外,内部存储器121可以包括高速随机存取存储器,还可以包括非易失性存储器,例如一个或者多个磁盘存储器件,闪存器件,通用闪存存储器(universal flash storage,UFS)等。
按键190包括开机键,音量键等。按键190可以是机械按键。也可以是触摸式按键。手机100可以接收按键输入,产生与手机100的用户设置以及功能控制有关的键信号输入。
马达191可以产生振动提示。指示器192可以是指示灯,可以用于指示充电状态,电量变化,也可以用于指示消息,未接来电,通知等。SIM卡接口195用于连接SIM卡。SIM卡可以通过插入SIM卡接口195,或从SIM卡接口195拔出,实现和手机100的接触和分离。
至此,已将终端各结构及其功能介绍完毕,下面结合附图,对本申请的无线扩音方法进行详细说明。
具体的,在本申请中,用户需要使用终端进行扩音时,可在终端的设置界面中,开启终端的无线扩音功能,如图3a所示,以使终端执行本申请中实施例中终端侧所执行的各步骤。示例性的,无线扩音功能的功能键还可以在下拉界面中,如图3b所示。
在本申请中,终端开启无线扩音功能后,终端可获取到用户的语音信号,并将获取到的用户的语音信号通过终端的音频模块,例如图2中的音频模块170,具体可以为音频模块中的音频数字信号处理模块进行处理后,输出处理后的语音信号。示例性的,音频数字信号处理模块可以为高保真(High-Fidelity,HIFI)模块,该模块在本申请的实施例中主要用于对音频模块获取到的信号进行扩音算法处理以及混音处理,具体功能将在下面的实施例中详细说明。可选地,在本申请中,采集用户的语音信号的设备可以为耳机麦克风,也可以为终端自带的麦克风。可选地,在本申请中,终端可将处理后的语音信号,通过自身的扬声器进行扩音,或者,也可以将处理后的语音信号输出到外接设备,通过外接设备对语音信号进行扩音,示例性的,外接设备可以为音响设备,例如,蓝牙 音响设备,或者,电视的音响设备等。
下面结合具体应用场景,对上述方案进行说明。在一种可能的实现方式中,采集用户的语音信号的装置可以为终端中的麦克风,终端将采集到的语音信号进行处理后,通过终端的扬声器对处理后的语音信号进行扩音,具体实施例可参照场景一。
在另一种可能的实现方式中,采集用户的语音信号的装置可以为终端中的麦克风,终端对采集到的语音信号进行处理后,可将处理后的语音信号输出至与其连接的音响设备,并由音响设备进行扩音,具体实施例可参照场景二。
在又一种可能的实现方式中,采集用户的语音信号的装置可以为与终端连接的耳机的麦克风(以下简称耳机麦克风),终端获取耳机麦克风采集到的用户的语音信号后,对语音信号进行处理,并通过终端扬声器对处理后的语音信号进行扩音,具体实施例可参照场景三。
在又一种可能的实现方式中,采集用户的语音信号的装置可以为与终端连接的耳机的麦克风,终端获取耳机麦克风采集到的用户的语音信号后,对语音信号进行处理,并将处理后的语音信号输出至与其连接的音响设备,并通过终端扬声器对处理后的语音信号进行扩音,具体实施例可参照场景四。
可选地,在本申请中,采集用户的语音信号的装置可以为与终端连接的耳机,耳机可通过终端与音响设备建立连接,例如建立蓝牙连接,耳机可将采集到的用户的语音信号进行处理后,通过与音响设备之间的蓝牙连接,将处理后的语音信号输出至音响设备,并通过终端扬声器对处理后的语音信号进行扩音,具体实施例可参照场景五。
需要说明的是,本申请所述的“连接”可以为有线连接,例如,耳机与手机可以通过耳机线相连。本申请所述的“连接”还可以为无线连接,例如,耳机为无线耳机,可通过蓝牙与手机连接,或者,手机与音响设备可通过蓝牙建立无线连接。
场景一
如图4所示为示例性示出的应用场景的示意图,参照图4,该应用场景中包括手机,在该场景中,用户可对手机说话,手机可通过手机麦克风采集用户的语音信号,并对语音信号进行扩音。示例性的,该场景可以为直播场景或演讲场景等,即,用户可在演讲时,通过手机对话音进行扩音,而无需携带特殊的扩音装置。
需要说明的是,图4中的手机,即本申请所述的终端的数量仅为示意性举例,在实际应用中,手机个数可以为一个或多个,本申请不做限定。进一步需要说明的是,在本场景以及下面的场景中,均以终端为手机为例进行说明,在其它实施例中,终端也可以为平板等其他设备,本申请不做限定。
结合图4,如图5所示为本申请实施例中的无线扩音方法的流程示意图,在图5中:
步骤101,手机通过手机麦克风采集用户的语音信号。
具体的,在本实施例中,手机开启无线扩音功能后,用户可对手机,具体是对手机中的麦克风说话,手机可通过麦克风采集到用户的语音信号,也可以称为人声信号。
步骤102,手机对语音信号进行处理。
具体的,如图6所示为手机对语音信号进行处理的流程示意图。参照图6,示例性 的,在本实施例中,手机具有两个麦克风,分别为麦克风1和麦克风2。需要说明的是,本申请仅以麦克风1和麦克风2为例进行说明,实际上,手机中的麦克风数量可以为两个或两个以上,本申请不做限定。
具体的,用户对手机说话,手机可通过麦克风1和麦克风2采集到用户的语音信号,需要说明的是,如图6所示,在手机侧,手机对用户的语音信号的处理过程可认为是一个环回过程,在该环回过程中,手机从麦克风处获取到的用户的语音信号,对于手机而言,可称为上行数据,相应的,手机对上行数据,即用户的语音信号进行处理后,输出的处理后的语音信号,可称为下行数据或者播放数据。
下面对上行数据的处理过程,即所述环回过程进行详细说明,具体包括:
手机的音频模块,如图2中的音频模块170中可包括音频模数处理模块,该模块可将麦克风1和麦克风2采集到的用户的语音信号进行模数转换,即,将采集到的用户的人声对应的模拟语音信号转换为数字语音信号,并将数字语音信号传输至音频数字信号处理模块。
仍参照图6,音频数字信号处理模块可基于预设的扩音算法,对数字语音信号进行处理,并将处理后的数字语音信号与播放链路进行混音处理后,得到下行数据。需要说明的是,混音过程可理解为将有用信息与载体混合,以获取承载有有用信息的信号,也就是说,音频数字信号处理模块将有用信息,即处理后的数字语音信号与载体,即播放链路进行混音,得到下行数据。示例性的,所述播放链路也可以称为音频流、载体或播放流,本申请不做限定。
具体的,在本申请中,音频数字信号处理模块基于扩音算法,对数字语音信号的处理可包括但不限于:环境降噪和增强处理、抗啸叫处理、防喷麦处理、采样率和位宽转换处理。
示例性的,环境降噪和增强处理可用于降低进入麦克风的噪声信号,如风噪、环境噪声以及其他干扰信号等,并可对用户的语音信号进行适度增强,以提升人声的清晰度和信噪比。该处理过程可参照已有技术,本申请不再赘述。
示例性的,采样率和位宽转换处理可用于将语音信号转换成与系统兼容的采样率及位宽,如48kHz/16bit。该处理过程可参照已有技术,本申请不再赘述。
示例性的,抗啸叫处理可用于消除啸叫现象。在本实施例中,手机可采用已有技术中的自适应回声消除算法(Acoustic Echo Cancellation,AEC)以消除啸叫,该方法主要是根据下行播放的信号,即上文中所述的下行数据,模拟出回声信号ref,也可以理解为干扰信号,手机麦克风采集到的信号(或声音)包括有用人声,即用户的语音信号和回声信号ref,即干扰信号,根据该算法,手机可从采集到的信号中去除回声信号ref,从而实现回声抵消,具体方法可参照已有技术,本申请不做赘述。
示例性的,防喷麦处理可用于消除喷麦现象,具体方式将在场景二中详细说明。
步骤103,手机通过扬声器输出处理后的语音信号。
具体的,仍参照图6,在本实施例中,音频数字信号处理模块将处理后的数字语音信号,即下行数据传输至音频模数处理模块,由音频模数处理模块进行数模转换,即,将数字语音信号转换为模拟语音信号,并通过手机的扬声器进行播放,从而实现对用户 的语音信号的扩音。可选地,用户可通过调节手机的音量的大小,以调节扬声器的扩音效果。
场景二
如图7所示为本申请实施例的应用场景的示意图,参照图7,该应用场景中包括手机和音响设备,其中,手机和音响设备建立通信连接,例如蓝牙连接,在该场景中,用户可对手机说话,手机可通过手机麦克风采集用户的语音信号,并通过音响设备对语音信号进行扩音。示例性的,该场景同样可以为直播场景或演讲场景等,即,用户可在演讲时,对手机说话,并通过音响设备实现对话音进行扩音。
需要说明的是,图7中的手机和音响设备的数量仅为示意性举例,在实际应用中,手机和音响设备的个数可以为一个或多个,本申请不做限定。
结合图7,如图8所示为本申请实施例中的无线扩音方法的流程示意图,在图8中:
步骤201,手机通过手机麦克风采集用户的语音信号。
具体的,在本实施例中,手机开启无线扩音功能后,用户可对手机,具体是对手机中的麦克风说话,手机可通过麦克风采集到用户的语音信号。
具体的,在本实施例中,手机检测到无线扩音功能开启后,手机可与附近的音响设备建立连接,需要说明的是,手机与音响设备的连接是指无线连接,一个示例中,手机可与音响设备建立蓝牙连接。另一个示例中,手机可与音响设备建立WIFI连接,本申请不做限定。
示例性的,在本实施例中,以手机与音响设备通过蓝牙连接为例进行说明。具体的,手机与音响设备建立蓝牙连接后,手机可将手机的蓝牙模式设置为同步定向链接(Synchronous Connection Oritened,SCO)模式,该模式用于同步话音传送,具有传输时延低的特性。
步骤202,手机对语音信号进行处理。
具体的,如图9所示为手机对语音信号进行处理的流程示意图,参照图9,示例性的,在本实施例中,手机具有两个麦克风,分别为麦克风1和麦克风2。
具体的,手机通过麦克风1和麦克风2采集用户的语音信号,并将用户的语音信号,即图9中的上行数据(概念可参照上文)经由音频数字信号处理模块进行处理后,得到下行数据。
仍参照图9,具体的,音频数字信号处理模块可基于预设的扩音算法,对数字语音信号进行处理,并将处理后的数字语音信号与播放链路进行混音后,得到下行数据。
示例性的,在本实施例中,音频数字信号处理模块对上行数据的处理过程包括但不限于:环境降噪和增强处理、抗啸叫处理、防喷麦处理、采样率和位宽转换处理。
可选地,在本实施例中,音频数字信号处理模块可基于本申请中的抗啸叫算法,对上行数据进行抗啸叫处理,具体算法将在下文中详细说明。可选地,音频数字信号处理模块的抗啸叫处理也可以采用已有技术中的AEC算法,以消除啸叫现象。
其它细节可参照步骤102,此处不赘述。
步骤203,手机将处理后的语音信号发送给音响设备。
具体的,仍参照图9,音频数字信号处理模块将处理后的数字语音信号,即下行数据传输至音频模数处理模块,音频模数处理模块可基于当前的蓝牙模式,即SCO模式,将数字语音信号传输给给蓝牙芯片,以通过蓝牙芯片将数字语音信号通过手机与音响设备之间的蓝牙通道传输给音响设备。
步骤204,音响设备播放语音信号。
具体的,音响设备可将接收到的数字语音信号转换为模拟语音信号,并播放模拟语音信号,以实现对用户的人声进行扩音。可选地,用户可通过调节音响设备的音量的大小,以调节扩音效果。
下面对于本申请所采用的抗啸叫处理以及防喷麦处理进行详细说明:
首先,为使本领域人员理解抗啸叫处理过程,首先对关于抗啸叫的背景技术进行简单介绍。啸叫产生的本质原因包括:音响设备播放的声音被手机麦克风重新采集后,再次传到音响设备播放,且来回反复;以及,音响设备直接传递到麦克风的直接回声和音响设备播出的声音经过外界反射后,再传输到麦克风,包含反射等环境不确定信息,如图10所示。
如上文所述,已有技术中通常采用自适应回声消除算法(Acoustic Echo Cancellation,AEC)以消除啸叫,相关概念可参照上文。
但是,对于AEC算法,手机需要准确估计出回声信号ref,才能准确抑制回声信号。但是,在已有技术中,不同的音响设备是指不同的生产厂商生产的音响设备,或者是同一个生产厂商生产的不同型号的音响设备,也就是说,不同的音响设备时延不同,且不同音响设备播放音量不同,导致影响设备播放时产生的失真也不一样。因此,现有的AEC算法只能对具有指定时延的连接设备进行回声消除,而对于非指定时延的连接设备,由于算法的局限性,手机无法准确地估计回声信号ref,进而无法准确消除回声,以抑制啸叫。
针对上述问题,本申请提出一种抗啸叫算法,可忽略时延的影响,也就是说,本申请中的抗啸叫算法可用于不同时延的连接设备之间的回声消除。具体的,如图11所示,音响设备反馈到麦克风端的回声属于远场声音,或成为远场信号,麦克风采集的用户的人声,即用户的语音信号属于近场声音,,或成为近场信号。如果能消除远场声音,就可以打断反馈通路,从而抑制远场声音造成的啸叫现象。由于近场声音到手机两个麦克风之间距离不同,手机两个麦克风采集到的声强差较大,而对远场声音,手机尺寸可以忽略不计,声音到手机的两个麦克风的信号强度相当,根据该特征,本申请的抗啸叫算法采用谱减法,以将远场声音消除,并保留近场声音。
如图12所示为抗啸叫算法的流程示意图,具体包括:
1)麦克风1和麦克风2采集到的信号均包括用户的语音信号和干扰信号(概念可参照上文)。
2)谐振峰处理。具体的,根据两个麦克风采集到的远场声音和近场声音差异,定义近场人声,即用户的语音信号到麦克风1的信号为s1,近场人声到麦克风2的信号为s2,音响设备反馈回来的干扰回声,即干扰信号为e,则麦克风1的采集到的信号为: x1=s1+e;麦克风2的采集到的信号为:x2=s2+e。麦克风1和麦克风2采集到的信号分别输入到均衡器模块,该模块用于对信号x1和信号x2的高频谐振峰进行处理。具体的,由于麦克风器件与管道的原因,信号在高频会存在15dB以上的谐振峰,使得采集到的信号和真实信号之间存在失真,对信号进行高频谢振峰处理后,可使麦克风频响平滑,即去除采集的信号中由器件带来的失真。均衡器模块输出处理后的信号,包括对麦克风1的信号x1处理后的信号y1,以及,对麦克风2的信号x2处理后的信号为y2。
3)滤波补偿,具体的,均衡器模块把y1和y2输入至滤波补偿模块,用于从y1中滤除掉y2中的信号,以获取人声,即用户的语音信号,并对滤波过程中对近场人声的损伤进行补偿。具体的,麦克风1对应的信号y1减去麦克风2对应的信号y2后,会使得获取到的用户的语音信号的能量减少,因此可通过对用户的语音进行放大,以补偿损伤。
4)平滑输出。具体的,滤波补偿模块将处理后的信号输入至平滑输出模块,该模块用于平滑上述处理过程造成的音量变化。
如图13所示为防喷麦算法的流程示意图,具体包括:
1)麦克风1和麦克风2采集到的信号均包括用户的语音信号和干扰信号(概念可参照上文)。
2)喷麦检测阈值。具体的,喷麦检测阈值模块可预先设置有喷麦阈值,若麦克风1与麦克风2的信号强度差大于喷麦阈值,则确定存在喷麦现象,反之,若麦克风1与麦克风2的信号强度差小于喷麦阈值,则不存在喷麦现象。示例性的,若喷麦检测阈值模块确定存在喷麦现象,则执行后续的步骤3)。
3)滤波。具体的,滤波模块可对主麦克风,例如麦克风1执行滤波操作,以滤除噪声。示例性的,如上文所述,麦克风1获取到的信号包括有用信号,即用户的语音信号和干扰信号。麦克风2获取到的信号包括用户的语音信号和干扰信号,其中,麦克风2为副麦克风,其与用户的距离较之麦克风1与用户的距离稍远,在发生喷麦现象时,可认为是主麦克风的干扰信号造成该现象,因此,麦克风2获取到的信号中所包含的干扰信号可忽略不计,也就是说,可将麦克风2获取到的信号认为是有用信号,即用户的语音信号。在滤波过程中,可将主麦克风,即麦克风1对应的信号减去麦克风2对应的信号,得到麦克风1的干扰信号。
3)滤波补偿。滤波补偿模块可对滤波后的用户的语音信号进行滤波补偿,具体的,滤波补偿模块可基于2)中获取到的麦克风1的干扰信号,滤除麦克风1对应的信号中的干扰信号部分,得到用户的语音信号,并对用户的语音进行放大,以补偿损伤。
6)平滑输出。具体的,滤波补偿模块将处理后的信号输入至平滑输出模块,该模块用于平滑上述处理过程造成的音量变化。
场景三
如图14所示为本申请实施例的应用场景的示意图,参照图14,该应用场景中包括耳机和手机,其中,耳机和手机设备建立通信连接,一个示例中,若耳机为有线耳机, 则耳机和手机通过耳机线连接。另一个示例中,若耳机为无线耳机,例如蓝牙耳机,则耳机与手机建立无线连接,例如蓝牙连接。在该场景中,用户可对耳机麦克风说话,手机可通过耳机麦克风采集用户的语音信号,并对语音信号进行扩音。示例性的,该场景可以为直播、演讲或者上课、会议等场景中。示例性的,本实施例中以有线耳机为例进行说明。需要说明的是,图14中的耳机和终端的数量仅为示意性举例,在实际应用中,耳机和手机的个数可以为一个或多个,本申请不做限定。
结合图14,如图15所示为本申请实施例中的无线扩音方法的流程示意图,在图15中:
步骤301,手机通过耳机麦克风采集用户的语音信号。
具体的,在本实施例中,手机开启无线扩音功能后,用户可对耳机的麦克风说话,耳机可采集到用户的语音信号,手机获取耳机采集到的语音信号。
可选地,若耳机为有线耳机,则耳机与手机通过耳机线连接。可选地,若耳机为无线耳机,耳机与手机可建立蓝牙连接。
步骤302,手机对语音信号进行处理。
具体的,如图16所示为手机对语音信号进行处理的流程示意图,参照图16,具体的,手机通过耳机麦克风采集用户的语音信号,并将用户的语音信号,即图16中的上行数据(概念可参照上文)经由音频数字信号处理模块进行处理后,得到下行数据。参照图16,具体的,音频数字信号处理模块可基于预设的扩音算法,对数字语音信号进行处理,并将处理后的数字语音信号与播放链路进行混音后,得到下行数据。
具体的,在本申请中,音频数字信号处理模块基于扩音算法,对数字语音信号的处理可包括但不限于:环境降噪和增强处理、抗啸叫处理、防喷麦处理、采样率和位宽转换处理。
在一种可能的实现方式中,耳机包括一个麦克风,在该实施例中,手机麦克风处于开启状态。相应的,手机麦克风可采集到用户的语音信号和干扰信号如图17所示为该场景下的抗啸叫算法的流程示意图,参照图17,具体的,耳机麦克风采集到的信号包括用户的语音信号和干扰信号,手机麦克风采集到的信号包括用户的语音信号和干扰信号,其中,由于在该应用场景中,手机与用户的距离通常较远,因此,手机麦克风采集到的用户的语音信号可忽略不计,也就是说,在噪声检测过程中,可以将手机麦克风采集到的信号确认为干扰信号。接着,对耳机麦克风采集到的信号,包括用户的语音信号和干扰信号以及手机麦克风采集到的干扰信号进行谐振峰处理以及滤波,谐振峰处理的相关概念可参照上文,此处不赘述。示例性的,滤波是指将耳机麦克风采集到的信号去除手机麦克风采集到的信号,即可滤除干扰信号,得到用户的语音信号。随后,可对得到的用户的语音信号进行滤波补偿处理和平滑输出处理,具体细节可参照上文,此处不赘述。
在另一种可能的实现方式中,耳机可包括两个或两个以上麦克风,在该实施例中,手机麦克风可开启,也可以关闭,本申请对此不做限定。示例性的,手机获取到耳机的多个麦克风采集到的信号,手机对获取到的信号进行抗啸叫的处理过程可参照场景二,此处不赘述。
需要说明的是,在本实施例中,防喷麦算法与抗啸叫算法类似,一个示例中,手机可基于两个或两个以上耳机麦克风采集到的信号,进行防喷麦处理。另一个示例中,手机可基于耳机麦克风和手机麦克风采集到的信号,尽心防喷麦处理,具体处理过程与场景二相同,此处不赘述。
其它细节可参照场景二,此处不赘述。
步骤303,手机通过扬声器输出处理后的语音信号。
具体的,仍参照图16,在本实施例中,音频数字信号处理模块将处理后的数字语音信号,即下行数据传输至音频模数处理模块,由音频模数处理模块进行数模转换,即,将数字语音信号转换为模拟语音信号,并通过手机的扬声器进行播放,从而实现对用户的语音信号的扩音。可选地,用户可通过调节手机的音量的大小,以调节扬声器的扩音效果。
场景四
如图18所示为本申请实施例的应用场景的示意图,参照图18,该应用场景中包括耳机、手机和音响设备,其中,耳机、手机和音响设备设备建立通信连接,一个示例中,若耳机为有线耳机,则耳机和手机通过耳机线连接。另一个示例中,若耳机为无线耳机,例如蓝牙耳机,则耳机与手机建立无线连接,例如蓝牙连接。以及,手机与音响设备可建立无线连接,例如蓝牙连接。
在该场景中,用户可对耳机麦克风说话,手机可通过耳机麦克风采集用户的语音信号,并通过音响设备对语音信号进行扩音。示例性的,该场景可以为直播、演讲或者上课、会议等场景中。示例性的,本实施例中以有线耳机为例进行说明。需要说明的是,图18中的耳机、手机和音响设备的数量仅为示意性举例,在实际应用中,耳机、手机和音响设备的个数可以为一个或多个,本申请不做限定。
结合图18,如图19所示为本申请实施例中的无线扩音方法的流程示意图,在图19中:
步骤401,手机通过耳机麦克风采集用户的语音信号。
具体的,在本实施例中,手机开启无线扩音功能后,用户可对耳机的麦克风说话,耳机可采集到用户的语音信号,手机获取耳机麦克风采集到的语音信号。
示例性的,在本实施例中,手机检测到无线扩音功能开启后,手机可与附近的音响设备建立蓝牙连接,并且,手机将蓝牙模式设置为SCO模式。其它细节可参照步骤201,此处不赘述。
步骤402,手机对语音信号进行处理。
具体的,如图20所示为手机对语音信号进行处理的流程示意图,参照图20,具体的,手机通过耳机麦克风采集用户的语音信号,并将用户的语音信号,即图20中的上行数据(概念可参照上文)经由音频数字信号处理模块进行处理后,得到下行数据。参照图20,具体的,音频数字信号处理模块可基于预设的扩音算法,对数字语音信号进行处理,并将处理后的数字语音信号与播放链路进行混音后,得到下行数据。
其它细节可参照步骤302,此处不赘述。
步骤403,手机将处理后的语音信号发送给音响设备。
具体的,仍参照图20,音频数字信号处理模块将处理后的数字语音信号,即下行数据传输至音频模数处理模块,音频模数处理模块可基于当前的蓝牙模式,即SCO模式,将数字语音信号传输给给蓝牙芯片,以通过蓝牙芯片将数字语音信号通过手机与音响设备之间的蓝牙通道传输给音响设备。
步骤404,音响设备播放语音信号。
具体的,音响设备可将接收到的数字语音信号转换为模拟语音信号,并播放模拟语音信号,以实现对用户的人声进行扩音。可选地,用户可通过调节音响设备的音量的大小,以调节扩音效果。
场景五
如图21所示为本申请实施例的应用场景的示意图,参照图21,该应用场景中包括耳机、手机和音响设备,其中,耳机和手机设备建立通信连接,示例性的,本实施例中的耳机为无线耳机,例如蓝牙耳机,耳机可与手机建立无线连接,例如蓝牙连接。在该场景中,用户可对耳机麦克风说话,耳机可采集到的语音信号传输到音响设备,以通过音响设备对语音信号进行扩音。示例性的,该场景可以为直播、演讲或者上课、会议等场景中。需要说明的是,图21中的耳机、手机和音响设备的数量仅为示意性举例,实际应用中,耳机、手机和音响设备的个数可以为一个或多个,本申请不做限定。
结合图21,如图22所示为本申请实施例中的无线扩音方法的流程示意图,在图22中:
步骤501,耳机麦克风采集用户的语音信号。
示例性的,在本实施例中,手机可与蓝牙耳机建立蓝牙连接。具体的,在本实施例中,手机开启无线扩音功能后,用户可对蓝牙耳机的麦克风说话,蓝牙耳机可采集到用户的语音信号。
示例性的,在本实施例中,手机检测到无线扩音功能开启后,手机可与附近的音响设备建立蓝牙连接,也可以理解为,蓝牙耳机通过手机与音响设备建立蓝牙连接,即,蓝牙耳机与音响设备可通过该蓝牙连接,或可称为蓝牙通道以传输数据。
步骤502,耳机对语音信号进行处理。
具体的,在本实施例中,蓝牙耳机中可配置有上述实施例中的手机端的音频数字信号处理模块以及音频模数处理模块,并实现上述手机端的音频数字信号处理模块以及音频模数处理模块的各项功能。
示例性的,蓝牙耳机获取到用户的语音信号后,可基于预设的扩音算法,对数字语音信号进行处理,并将处理后的数字语音信号与播放链路进行混音后,得到下行数据。蓝牙耳机对语音信号的处理过程与上述实施例中的手机内部处理过程相同,此处不赘述。
步骤503,耳机将处理后的语音信号发送给音响设备。
示例性的,在本实施例中,耳机可将处理后的语音信号,通过蓝牙芯片经耳机与音响设备之间的蓝牙通道传输给音响设备。
具体的,在已有技术中,蓝牙耳机设置有左右耳传输协议,该协议规定主耳,例如耳机的右耳,将声音传输给副耳,例如耳机的左耳,以实现双耳播放音频,具体协议内容可参照已有技术,本申请不做赘述。在本实施例中,蓝牙耳机可基于左右耳协议,将音响蓝牙作为副耳,并将获取到的下行数据,即数字语音信号传输给音响设备。
步骤504,音响设备播放语音信号。
具体的,音响设备可将接收到的数字语音信号转换为模拟语音信号,并播放模拟语音信号,以实现对用户的人声进行扩音。可选地,用户可通过调节音响设备的音量的大小,以调节扩音效果。
在一种可能的实现方式中,如上文所述,各应用场景中的设备的数量可包括一个或多个,一个示例中,如图23所示为多设备的应用场景示意图,参照图23,其中包括手机1和有线耳机、手机2和蓝牙耳机以及手机3以及音响设备。需要说明的是,手机、耳机和音响设备数量仅为示意性举例,本申请不做限定。具体的,在该场景下,手机1、手机2和手机3可通过无线网络与音响设备建立无线连接,以传输数据。具体的,手机1、手机2和/或手机3通过耳机麦克风和/或手机麦克风获取到用户的语音信号后,可对用户的语音信号进行处理,并将处理后的语音信号发送给音响设备,以通过音响设备对用户的语音信号进行扩音,从而实现在会议或者活动应用场景中,对多用户的声音进行扩音。
上述主要从各个网元之间交互的角度对本申请实施例提供的方案进行了介绍。可以理解的是,终端为了实现上述功能,其包含了执行各个功能相应的硬件结构和/或软件模块。本领域技术人员应该很容易意识到,结合本文中所公开的实施例描述的各示例的单元及算法步骤,本申请实施例能够以硬件或硬件和计算机软件的结合形式来实现。某个功能究竟以硬件还是计算机软件驱动硬件的方式来执行,取决于技术方案的特定应用和设计约束条件。专业技术人员可以对每个特定的应用来使用不同方法来实现所描述的功能,但是这种实现不应认为超出本申请的范围。
本申请实施例可以根据上述方法示例对终端进行功能模块的划分,例如,可以对应各个功能划分各个功能模块,也可以将两个或两个以上的功能集成在一个处理模块中。上述集成的模块既可以采用硬件的形式实现,也可以采用软件功能模块的形式实现。需要说明的是,本申请实施例中对模块的划分是示意性的,仅仅为一种逻辑功能划分,实际实现时可以有另外的划分方式。
如图24所示为终端200的结构示意图,参照图24,终端200包括获取模块201、处理模块202和发送模块203。具体的,获取模块201用于获取终端的第一麦克风采集到的第一语音信号,以及第二麦克风采集到的第二语音信号;处理模块202,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,发送模块203,用于将播放语音信号传输至扩音设备,所述扩音设备为与所述终端无线连接的外接设备。
在上述技术方案的基础上,处理模块202具体用于响应于获取到的第一语音信号和 第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到播放语音信号。
在上述技术方案的基础上,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在上述技术方案的基础上,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
如图25所示为终端300的结构示意图,参照图25,终端300包括获取模块301、处理模块302、扬声器模块303。具体的,获取模块301,用于获取耳机的第一麦克风采集到的第一语音信号,以及耳机的第二麦克风采集到的第二语音信号,该耳机与终端有线连接或无线连接;处理模块302,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,扬声器模块303,用于对播放语音信号进行扩音。
在上述技术方案的基础上,处理模块302具体用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到第四数字语音信号;对第四数字语音信号进行模数转换,获取播放语音信号。
在上述技术方案的基础上,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在上述技术方案的基础上,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
如图26所示为终端400的结构示意图,参照图26,终端400包括:获取模块401、处理模块402和发送模块403。具体的,获取模块401,用于获取耳机的第一麦克风采集到的第一语音信号,以及耳机的第二麦克风采集到的第二语音信号,该耳机与终端有线 连接或无线连接;处理模块402,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,发送模块403,用于将播放语音信号传输至扩音设备,所述扩音设备为与所述终端无线连接的外接设备。
在上述技术方案的基础上,处理模块402,具体用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到播放语音信号。
在上述技术方案的基础上,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在上述技术方案的基础上,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
如图27所示为终端500的示意图,参照图27,终端500包括:获取模块501、处理模块502和发送模块503,具体的,获取模块501,用于采集语音信号,处理模块502,用于响应于获取到的语音信号,对语音信号进行处理,得到播放语音信号,发送模块503,用于将播放语音信号传输至扩音设备。
在上述技术方案的基础上,拾音设备为与终端有线连接或无线连接的耳机上的麦克风,包括第一麦克风和第二麦克风。
在上述技术方案的基础上,拾音设备为与终端的麦克风,包括第一麦克风和第二麦克风。
在上述技术方案的基础上,语音信号包括第一语音信号与第二语音信号;获取模块501,用于通过第一麦克风采集第一语音信号,以及,通过第二麦克风采集第二语音信号;处理模块502,用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号和第二语音信号进行处理,得到播放语音信号,并将播放语音信号传输至扩音设备。
在上述技术方案的基础上,扩音设备为终端的扬声器;处理模块502具体用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到第四数字语音信号;对第四数字语音信号进行模数转换,获取播放语音信号。
在上述技术方案的基础上,扩音设备为与终端无线连接的扩音设备;处理模块502 具体用于响应于获取到的第一语音信号和第二语音信号,对第一语音信号进行模数转换,得到第一数字语音信号,以及,对第二语音信号进行模数转换,得到第二数字语音信号,并将第一数字语音信号和第二数字语音信号,其中,第一语音信号和第二语音信号为模拟信号;基于预设扩音算法,对第一数字语音信号和第二数字语音信号进行音效处理,得到第三数字语音信号;并对第三数字语音信号进行混音处理,得到播放语音信号,将获取到的播放语音信号传输至扩音设备。
在上述技术方案的基础上,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号与干扰信号;预设扩音算法包括抗啸叫算法,抗啸叫算法具体包括:从第一语音信号对应的第一数字语音信号中滤除第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到第三数字语音信号,其中,第一语音信号的信号强度大于第二语音信号的信号强度。
在上述技术方案的基础上,第一语音信号包括第一人声信号与干扰信号,第二语音信号包括第二人声信号;预设扩音算法包括防喷麦算法,防喷麦算法具体包括:若第一语音信号与第二语音信号之间的信号强度差值大于喷麦阈值,则基于第二人声信号,确定第一语音信号中的干扰信号;从第一语音信号中滤除干扰信号。
在另一个示例中,图28示出了本申请实施例的一种终端600的示意性框图,终端可以包括:处理器601和收发器/收发管脚602,可选地,还包括存储器603。该处理器601可用于执行前述的实施例的各方法中的终端所执行的步骤,并控制接收管脚接收信号,以及控制发送管脚发送信号。
终端600的各个组件通过总线604耦合在一起,其中总线系统604除包括数据总线之外,还包括电源总线、控制总线和状态信号总线。但是为了清楚说明起见,在图中将各种总线都标为总线系统604。
可选地,存储器603可以用于前述方法实施例中的存储指令。
应理解,根据本申请实施例的终端600可对应于前述的实施例的各方法中的终端,并且终端600中的各个元件的上述和其它管理操作和/或功能分别为了实现前述各个方法的相应步骤,为了简洁,在此不再赘述。
其中,上述方法实施例涉及的各步骤的所有相关内容均可以援引到对应功能模块的功能描述,在此不再赘述。
基于相同的技术构思,本申请实施例还提供一种计算机可读存储介质,该计算机可读存储介质存储有计算机程序,该计算机程序包含至少一段代码,该至少一段代码可由终端执行,以控制终端用以实现上述方法实施例。
基于相同的技术构思,本申请实施例还提供一种计算机程序,当该计算机程序被终端执行时,用以实现上述方法实施例。
所述程序可以全部或者部分存储在与处理器封装在一起的存储介质上,也可以部分或者全部存储在不与处理器封装在一起的存储器上。
基于相同的技术构思,本申请实施例还提供一种处理器,该处理器用以实现上述方法实施例。上述处理器可以为芯片。
结合本申请实施例公开内容所描述的方法或者算法的步骤可以硬件的方式来实现, 也可以是由处理器执行软件指令的方式来实现。软件指令可以由相应的软件模块组成,软件模块可以被存放于随机存取存储器(Random Access Memory,RAM)、闪存、只读存储器(Read Only Memory,ROM)、可擦除可编程只读存储器(Erasable Programmable ROM,EPROM)、电可擦可编程只读存储器(Electrically EPROM,EEPROM)、寄存器、硬盘、移动硬盘、只读光盘(CD-ROM)或者本领域熟知的任何其它形式的存储介质中。一种示例性的存储介质耦合至处理器,从而使处理器能够从该存储介质读取信息,且可向该存储介质写入信息。当然,存储介质也可以是处理器的组成部分。处理器和存储介质可以位于ASIC中。另外,该ASIC可以位于网络设备中。当然,处理器和存储介质也可以作为分立组件存在于网络设备中。
本领域技术人员应该可以意识到,在上述一个或多个示例中,本申请实施例所描述的功能可以用硬件、软件、固件或它们的任意组合来实现。当使用软件实现时,可以将这些功能存储在计算机可读介质中或者作为计算机可读介质上的一个或多个指令或代码进行传输。计算机可读介质包括计算机存储介质和通信介质,其中通信介质包括便于从一个地方向另一个地方传送计算机程序的任何介质。存储介质可以是通用或专用计算机能够存取的任何可用介质。
上面结合附图对本申请的实施例进行了描述,但是本申请并不局限于上述的具体实施方式,上述的具体实施方式仅仅是示意性的,而不是限制性的,本领域的普通技术人员在本申请的启示下,在不脱离本申请宗旨和权利要求所保护的范围情况下,还可做出很多形式,均属于本申请的保护之内。

Claims (20)

  1. 一种无线扩音系统,其特征在于,包括终端与扩音设备,所述终端与所述扩音设备无线连接,所述终端包括音频模块、第一麦克风和第二麦克风;
    所述第一麦克风,用于采集第一语音信号,并将所述第一语音信号传输至所述音频模块;
    所述第二麦克风,用于采集第二语音信号,并将所述第二语音信号传输至所述音频模块;
    所述音频模块,用于响应于获取到的所述第一语音信号和所述第二语音信号,对所述第一语音信号和所述第二语音信号进行处理,得到播放语音信号,并将所述播放语音信号传输至所述扩音设备;
    所述扩音设备,用于响应于获取到的所述播放语音信号,对所述播放语音信号进行扩音。
  2. 根据权利要求1所述的系统,其特征在于,所述音频模块包括音频模数处理模块和音频数字信号处理模块;
    音频模数处理模块,用于响应于获取到的所述第一语音信号和所述第二语音信号,对所述第一语音信号进行模数转换,得到第一数字语音信号,以及,对所述第二语音信号进行模数转换,得到第二数字语音信号,并将所述第一数字语音信号和所述第二数字语音信号传输至所述音频数字信号处理模块,其中,所述第一语音信号和所述第二语音信号为模拟信号;
    所述音频数字信号处理模块,用于响应于获取到的所述第一数字语音信号和所述第二数字语音信号,基于预设扩音算法,对所述第一数字语音信号和所述第二数字语音信号进行音效处理,得到第三数字语音信号;并对所述第三数字语音信号进行混音处理,得到所述播放语音信号,以及,将所述播放语音信号传输至所述音频模数处理模块;
    所述音频模数处理模块,还用于将获取到的所述播放语音信号传输至所述扩音设备。
  3. 根据权利要求1所述的系统,其特征在于,所述第一语音信号包括第一人声信号与干扰信号,所述第二语音信号包括第二人声信号与所述干扰信号;所述预设扩音算法包括抗啸叫算法,所述抗啸叫算法具体包括:
    从所述第一语音信号对应的第一数字语音信号中滤除所述第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到所述第三数字语音信号,其中,所述第一语音信号的信号强度大于所述第二语音信号的信号强度。
  4. 根据权利要求1所述的系统,其特征在于,所述第一语音信号包括第一人声信号与干扰信号,所述第二语音信号包括第二人声信号;所述预设扩音算法包括防喷麦算法,所述防喷麦算法具体包括:
    若所述第一语音信号与所述第二语音信号之间的信号强度差值大于喷麦阈值,则基 于所述第二人声信号,确定所述第一语音信号中的干扰信号;
    从所述第一语音信号中滤除所述干扰信号。
  5. 一种无线扩音系统,其特征在于,包括耳机与终端,所述耳机与所述扩音设备有线连接或无线连接,所述终端包括音频模块和扬声器,所述耳机包括第一麦克风和第二麦克风;
    所述第一麦克风,用于采集第一语音信号,并将所述第一语音信号传输至所述音频模块;
    所述第二麦克风,用于采集第二语音信号,并将所述第二语音信号传输至所述音频模块;
    所述音频模块,用于响应于获取到的所述第一语音信号和所述第二语音信号,对所述第一语音信号和所述第二语音信号进行处理,得到播放语音信号,并将所述播放语音信号传输至所述扬声器;
    所述扬声器,用于响应于获取到的所述播放语音信号,对所述播放语音信号进行扩音。
  6. 根据权利要求5所述的系统,其特征在于,所述音频模块包括音频模数处理模块和音频数字信号处理模块;
    所述音频模数处理模块,用于响应于获取到的所述第一语音信号和所述第二语音信号,对所述第一语音信号进行模数转换,得到第一数字语音信号,以及,对所述第二语音信号进行模数转换,得到第二数字语音信号,并将所述第一数字语音信号和所述第二数字语音信号传输至所述音频数字信号处理模块,其中,所述第一语音信号和所述第二语音信号为模拟信号;
    所述音频数字信号处理模块,用于响应于获取到的所述第一数字语音信号和所述第二数字语音信号,基于预设扩音算法,对所述第一数字语音信号和所述第二数字语音信号进行音效处理,得到第三数字语音信号;并对所述第三数字语音信号进行混音处理,得到第四数字语音信号,以及,将所述第四数字语音信号传输至所述音频模数处理模块;
    所述音频模数处理模块,还用于响应于获取到的所述第四数字语音信号,对所述第四数字语音信号进行模数转换,获取所述播放语音信号,并将所述播放语音信号传输至所述扬声器;其中,所述播放语音信号为模拟信号。
  7. 根据权利要求5所述的系统,其特征在于,所述第一语音信号包括第一人声信号与干扰信号,所述第二语音信号包括第二人声信号与所述干扰信号;所述预设扩音算法包括抗啸叫算法,所述抗啸叫算法具体包括:
    从所述第一语音信号对应的第一数字语音信号中滤除所述第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到所述第三数字语音信号,其中,所述第一语音信号的信号强度大于所述第二语音信号的信号强度。
  8. 根据权利要求5所述的系统,其特征在于,所述第一语音信号包括第一人声信号 与干扰信号,所述第二语音信号包括第二人声信号;所述预设扩音算法包括防喷麦算法,所述防喷麦算法具体包括:
    若所述第一语音信号与所述第二语音信号之间的信号强度差值大于喷麦阈值,则基于所述第二人声信号,确定所述第一语音信号中的干扰信号;
    从所述第一语音信号中滤除所述干扰信号。
  9. 一种无线扩音系统,其特征在于,包括耳机、手机和扩音设备,所述耳机与所述扩音设备有线连接或无线连接,所述手机与所述扩音设备无线连接,所述终端包括音频模块,所述耳机包括第一麦克风和第二麦克风;
    所述第一麦克风,用于采集第一语音信号,并将所述第一语音信号传输至所述音频模块;
    所述第二麦克风,用于采集第二语音信号,并将所述第二语音信号传输至所述音频模块;
    所述音频模块,用于响应于获取到的所述第一语音信号和所述第二语音信号,对所述第一语音信号和所述第二语音信号进行处理,得到播放语音信号,并将所述播放语音信号传输至所述扩音设备;
    所述扩音设备,用于响应于获取到的所述播放语音信号,对所述播放语音信号进行扩音。
  10. 根据权利要求9所述的系统,其特征在于,所述音频模块包括音频模数处理模块和音频数字信号处理模块;
    所述音频模数处理模块,用于响应于获取到的所述第一语音信号和所述第二语音信号,对所述第一语音信号进行模数转换,得到第一数字语音信号,以及,对所述第二语音信号进行模数转换,得到第二数字语音信号,并将所述第一数字语音信号和所述第二数字语音信号传输至所述音频数字信号处理模块,其中,所述第一语音信号和所述第二语音信号为模拟信号;
    所述音频数字信号处理模块,用于响应于获取到的所述第一数字语音信号和所述第二数字语音信号,基于预设扩音算法,对所述第一数字语音信号和所述第二数字语音信号进行音效处理,得到第三数字语音信号;并对所述第三数字语音信号进行混音处理,得到所述播放语音信号,以及,将所述播放语音信号传输至所述音频模数处理模块;
    所述音频模数处理模块,还用于将获取到的所述播放语音信号传输至所述扩音设备。
  11. 根据权利要求9所述的系统,其特征在于,所述第一语音信号包括第一人声信号与干扰信号,所述第二语音信号包括第二人声信号与所述干扰信号;所述预设扩音算法包括抗啸叫算法,所述抗啸叫算法具体包括:
    从所述第一语音信号对应的第一数字语音信号中滤除所述第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到所述第三数字语音信号,其中,所述第一语音信号的信号强度大于所述第二语音信号的信号强度。
  12. 根据权利要求9所述的系统,其特征在于,所述第一语音信号包括第一人声信号与干扰信号,所述第二语音信号包括第二人声信号;所述预设扩音算法包括防喷麦算法,所述防喷麦算法具体包括:
    若所述第一语音信号与所述第二语音信号之间的信号强度差值大于喷麦阈值,则基于所述第二人声信号,确定所述第一语音信号中的干扰信号;
    从所述第一语音信号中滤除所述干扰信号。
  13. 一种终端,其特征在于,包括音频模块和拾音设备;
    所述拾音设备,用于采集语音信号,并将所述语音信号传输至所述音频模块;
    所述音频模块,用于响应于获取到的所述语音信号,对所述语音信号进行处理,得到播放语音信号,并将所述播放语音信号传输至扩音设备。
  14. 根据权利要求13所述的终端,其特征在于,所述拾音设备为与所述终端有线连接或无线连接的耳机上的麦克风,包括第一麦克风和第二麦克风。
  15. 根据权利要求13所述的终端,其特征在于,所述拾音设备为与所述终端的麦克风,包括第一麦克风和第二麦克风。
  16. 根据权利要求14或15所述的终端,其特征在于,所述语音信号包括第一语音信号和第二语音信号;
    所述第一麦克风,用于采集所述第一语音信号,并将所述第一语音信号传输至所述音频模块;
    所述第二麦克风,用于采集所述第二语音信号,并将所述第二语音信号传输至所述音频模块;
    所述音频模块,用于响应于获取到的所述第一语音信号和所述第二语音信号,对所述第一语音信号和所述第二语音信号进行处理,得到播放语音信号,并将所述播放语音信号传输至所述扩音设备。
  17. 根据权利要求16所述的终端,其特征在于,所述扩音设备为所述终端的扬声器;所述音频模块包括音频模数处理模块和音频数字信号处理模块;
    所述音频模数处理模块,用于响应于获取到的所述第一语音信号和所述第二语音信号,对所述第一语音信号进行模数转换,得到第一数字语音信号,以及,对所述第二语音信号进行模数转换,得到第二数字语音信号,并将所述第一数字语音信号和所述第二数字语音信号传输至所述音频数字信号处理模块,其中,所述第一语音信号和所述第二语音信号为模拟信号;
    所述音频数字信号处理模块,用于响应于获取到的所述第一数字语音信号和所述第二数字语音信号,基于预设扩音算法,对所述第一数字语音信号和所述第二数字语音信号进行音效处理,得到第三数字语音信号;并对所述第三数字语音信号进行混音处理, 得到第四数字语音信号,以及,将所述第四数字语音信号传输至所述音频模数处理模块;
    所述音频模数处理模块,还用于响应于获取到的所述第四数字语音信号,对所述第四数字语音信号进行模数转换,获取所述播放语音信号,并将所述播放语音信号传输至所述扬声器;其中,所述播放语音信号为模拟信号。
  18. 根据权利要求16所述的终端,其特征在于,所述扩音设备为与所述终端无线连接的扩音设备;
    所述音频模数处理模块,用于响应于获取到的所述第一语音信号和所述第二语音信号,对所述第一语音信号进行模数转换,得到第一数字语音信号,以及,对所述第二语音信号进行模数转换,得到第二数字语音信号,并将所述第一数字语音信号和所述第二数字语音信号传输至所述音频数字信号处理模块,其中,所述第一语音信号和所述第二语音信号为模拟信号;
    所述音频数字信号处理模块,用于响应于获取到的所述第一数字语音信号和所述第二数字语音信号,基于预设扩音算法,对所述第一数字语音信号和所述第二数字语音信号进行音效处理,得到第三数字语音信号;并对所述第三数字语音信号进行混音处理,得到所述播放语音信号,以及,将所述播放语音信号传输至所述音频模数处理模块;
    所述音频模数处理模块,还用于将获取到的所述播放语音信号传输至所述扩音设备。
  19. 根据权利要求17或18所述的终端,其特征在于,所述第一语音信号包括第一人声信号与干扰信号,所述第二语音信号包括第二人声信号与所述干扰信号;所述预设扩音算法包括抗啸叫算法,所述抗啸叫算法具体包括:
    从所述第一语音信号对应的第一数字语音信号中滤除所述第二语音信号对应的第二数字语音信号,并对滤除后的第一数字语音信号进行放大处理,得到所述第三数字语音信号,其中,所述第一语音信号的信号强度大于所述第二语音信号的信号强度。
  20. 根据权利要求17或18所述的终端,其特征在于,所述第一语音信号包括第一人声信号与干扰信号,所述第二语音信号包括第二人声信号;所述预设扩音算法包括防喷麦算法,所述防喷麦算法具体包括:
    若所述第一语音信号与所述第二语音信号之间的信号强度差值大于喷麦阈值,则基于所述第二人声信号,确定所述第一语音信号中的干扰信号;
    从所述第一语音信号中滤除所述干扰信号。
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