WO2022227982A1 - Tws耳机和tws耳机的播放方法及装置 - Google Patents

Tws耳机和tws耳机的播放方法及装置 Download PDF

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Publication number
WO2022227982A1
WO2022227982A1 PCT/CN2022/083464 CN2022083464W WO2022227982A1 WO 2022227982 A1 WO2022227982 A1 WO 2022227982A1 CN 2022083464 W CN2022083464 W CN 2022083464W WO 2022227982 A1 WO2022227982 A1 WO 2022227982A1
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WIPO (PCT)
Prior art keywords
signal
filter
speaker
frequency bands
ear canal
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PCT/CN2022/083464
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English (en)
French (fr)
Chinese (zh)
Inventor
熊伟
仇存收
恽毅
许超
郭琴
李岩
田立生
Original Assignee
华为技术有限公司
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Priority to EP22794455.0A priority Critical patent/EP4297428A4/de
Publication of WO2022227982A1 publication Critical patent/WO2022227982A1/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
    • H04R1/00Details of transducers, loudspeakers or microphones
    • H04R1/10Earpieces; Attachments therefor ; Earphones; Monophonic headphones
    • H04R1/1083Reduction of ambient noise
    • 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
    • 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
    • G10K11/1787General system configurations
    • G10K11/17879General system configurations using both a reference signal and an error signal
    • G10K11/17881General system configurations using both a reference signal and an error signal the reference signal being an acoustic signal, e.g. recorded with a microphone
    • 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
    • G10K11/1787General system configurations
    • G10K11/17885General system configurations additionally using a desired external signal, e.g. pass-through audio such as music or speech
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R23/00Transducers other than those covered by groups H04R9/00 - H04R21/00
    • H04R23/02Transducers using more than one principle simultaneously
    • 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
    • H04R3/12Circuits for transducers, loudspeakers or microphones for distributing signals to two or more loudspeakers
    • 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
    • H04R3/12Circuits for transducers, loudspeakers or microphones for distributing signals to two or more loudspeakers
    • H04R3/14Cross-over networks
    • 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
    • G10K2210/00Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/10Applications
    • G10K2210/108Communication systems, e.g. where useful sound is kept and noise is cancelled
    • G10K2210/1081Earphones, e.g. for telephones, ear protectors or headsets
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00Details of transducers, loudspeakers or microphones
    • H04R1/10Earpieces; Attachments therefor ; Earphones; Monophonic headphones
    • H04R1/1016Earpieces of the intra-aural type
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00Details of transducers, loudspeakers or microphones
    • H04R1/20Arrangements for obtaining desired frequency or directional characteristics
    • H04R1/22Arrangements for obtaining desired frequency or directional characteristics for obtaining desired frequency characteristic only 
    • H04R1/24Structural combinations of separate transducers or of two parts of the same transducer and responsive respectively to two or more frequency ranges
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00Signal processing covered by H04R, not provided for in its groups
    • H04R2430/03Synergistic effects of band splitting and sub-band processing
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2460/00Details of hearing devices, i.e. of ear- or headphones covered by H04R1/10 or H04R5/033 but not provided for in any of their subgroups, or of hearing aids covered by H04R25/00 but not provided for in any of its subgroups
    • H04R2460/01Hearing devices using active noise cancellation
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R5/00Stereophonic arrangements
    • H04R5/033Headphones for stereophonic communication

Definitions

  • the present application relates to audio processing technology, and in particular, to a true wireless stereo (true wireless stereo, TWS) headset and a TWS headset playback method and device.
  • true wireless stereo true wireless stereo, TWS
  • the general demand is to achieve stable active noise cancellation (active noise cancellation or active noise control, ANC) or transparent transmission (hear through, HT) function while having high-quality music or broadband HD calls .
  • TWS earphones implement the ANC function, they will also eliminate some music or call sounds, which will affect the sound quality and call clarity.
  • the present application provides a TWS earphone and a TWS earphone playing method and device, which can not only reflect high sound quality in various frequency bands of an audio source, but also support ultra-bandwidth voice calls.
  • the present application provides a TWS earphone, comprising: an audio signal processing path, a frequency divider and at least two speakers; wherein the output end of the audio signal processing path is connected to the input end of the frequency divider; the frequency divider The output end of the frequency converter is connected to the at least two speakers; the audio signal processing path is configured to output a speaker drive signal after noise reduction or transparent transmission processing is performed on the audio source; the audio source is original music or call voice; or , the audio source includes the voice signal enhanced by the human voice and the original music or call voice; the frequency divider is configured to divide the speaker drive signal into sub-audio signals of at least two frequency bands, the at least two frequency bands The frequency bands correspond to the main working frequency bands of the at least two speakers; the adjacent frequency bands in the at least two frequency bands partially overlap, or the adjacent frequency bands in the at least two frequency bands do not overlap; the at least two speakers are configured with To play the corresponding sub audio signal.
  • the frequency band of the processed speaker driving signal corresponds to the frequency band of the audio source, and may include the whole frequency range of low, medium and high. However, because the main working frequency band of a single speaker may only cover part of the low, middle and high frequency bands, a single speaker cannot operate in the entire frequency band. reflect high-quality sound.
  • the present application controls the frequency divider to divide the frequency of the speaker drive signal in a preset manner by setting the parameters of the frequency divider.
  • the frequency divider may be configured to divide the frequency of the speaker drive signal based on the main working frequency bands of the at least two speakers to obtain sub-audio signals of at least two frequency bands corresponding to the main working frequency bands of the at least two speakers, and then let each speaker Play the sub audio signals of the corresponding frequency bands respectively, so that the speaker maintains the best frequency response when playing the sub audio signals transmitted to it.
  • At least two speakers are set in the TWS earphone of the present application, and the main working frequency bands of the at least two speakers are not identical.
  • the frequency divider can divide the speaker drive signal into sub-audio signals of at least two frequency bands, and the adjacent frequency bands in the at least two frequency bands partially overlap or do not overlap, so that each sub-audio signal is respectively transmitted to the speakers with matching frequency bands,
  • the aforementioned frequency band matching can mean that the main working frequency band of the speaker covers the frequency band of the sub-audio signal transmitted to it, so that the speaker maintains the best frequency response when playing the sub-audio signal transmitted to it. It reflects high sound quality and supports ultra-bandwidth voice calls.
  • the audio signal processing path includes: a secondary path SP filter configured to prevent the noise reduction or pass-through processing when the noise reduction or pass-through processing is concurrent with the audio source Cancellation of sound from this audio source.
  • the audio signal processing path further includes: a feedback FB microphone and a feedback filter; wherein the FB microphone is configured to pick up an ear canal signal, and the ear canal signal includes residual noise inside the ear canal signal and the music or the voice of the conversation; the SP filter is configured to input the audio source, process the audio source, and transmit the output signal and the ear canal signal to the feedback filter after superimposing; the feedback filter, is configured to generate a signal for the noise reduction or pass-through processing, the noise reduction or pass-through processed signal being one of the superimposed signals for generating the speaker drive signal.
  • the feedback FB microphone, the feedback filter and the SP filter are set in the codec CODEC.
  • the SP filter is configured to input the audio source, process the audio source, and the output signal is one of the superimposed signals of the speaker driving signal.
  • the SP filter is set in a digital signal processing DSP chip.
  • the SP filter (including fixed SP filter or adaptive SP filter) is determined by the above method, which can not only realize the function of noise reduction or transparent transmission, but also prevent the transparent transmission or noise reduction technology in the process of playing music or passing through. Also remove music or call sounds.
  • it further includes: a first digital-to-analog converter DAC; the input end of the first DAC is connected to the output end of the audio signal processing path, and the output end of the first DAC is connected to the frequency divider
  • the first DAC is configured to convert the speaker driving signal from a digital form to an analog form; correspondingly, the frequency divider is an analog frequency divider.
  • the speaker drive signal After the speaker drive signal is obtained, if the signal does not pass through the DAC, it is in digital form, but the signal played by the speaker needs to be in analog form, so the digital speaker drive signal can be converted into analog form through the first DAC
  • the speaker driving signal in the analog form is then divided into sub audio signals of at least two frequency bands through an analog frequency divider.
  • This embodiment adopts the structure of first conversion and then frequency division.
  • the method further includes: at least two second DACs; the input ends of the at least two second DACs are connected to the output ends of the frequency divider, and the output ends of the at least two second DACs are connected to each other. are respectively connected with one of the at least two speakers; the second DAC is configured to convert one of the sub-audio signals of the at least two frequency bands from a digital form to an analog form; correspondingly, the frequency divider is a digital divider.
  • This embodiment adopts the structure of frequency division first and then conversion.
  • the main working frequency bands of the at least two speakers are not identical.
  • the at least two speakers include a moving coil speaker and a moving iron speaker.
  • Two speakers are set in the TWS earphone of this embodiment, namely a moving coil speaker and a moving iron speaker.
  • the main working frequency band of the moving coil speaker is below 8.5kHz, and the main working frequency band of the moving iron speaker is above 8.5kHz.
  • the frequency divider divides the analog speaker drive signal into a sub audio signal below 8.5kHz and a sub audio signal above 8.5kHz.
  • the moving coil speaker can maintain the best frequency response by playing the sub audio signal below 8.5kHz.
  • the sub-audio signal above 8.5kHz can maintain the best frequency response, so that the TWS headset can not only reflect the high sound quality in each frequency band of the audio source, but also support ultra-bandwidth voice calls.
  • the at least two speakers include a moving coil speaker, a moving iron speaker, a MEMS speaker and a planar vibration diaphragm.
  • the main working frequency band of MEMS speakers depends on the application form.
  • the main working frequency band of in-ear headphones is the full frequency band
  • the main working frequency band of headphone is weaker below 7kHz
  • the main working frequency band is high frequency above 7kHz.
  • the main working frequency band of the flat vibrating diaphragm is 10kHz ⁇ 20kHz, in this way, the frequency divider can be set to divide the analog speaker driving signal into four sub-frequency bands, and the sub-audio signal below 8.5kHz can be played by the moving coil speaker to maintain the best frequency response.
  • the moving iron speaker can maintain the best frequency response when playing sub-audio signals above 8.5kHz
  • the MEMS speaker can maintain the best frequency response when playing sub-audio signals above 7kHz
  • the flat vibrating diaphragm can maintain the best frequency response when playing sub-audio signals above 10kHz frequency response, so that the TWS headset can not only reflect high sound quality in various frequency bands of the audio source, but also support ultra-bandwidth voice calls.
  • the present application provides a method for playing a TWS headset, which is applied to the TWS headset according to any one of the first aspects above; the method includes: acquiring an audio source, where the audio source is original music or call voice, or, the audio source includes a voice signal enhanced by human voice and the original music or call voice; noise reduction or transparent transmission processing is performed on the audio source to obtain a speaker drive signal; The driving signal is divided into sub-audio signals of at least two frequency bands, and the adjacent frequency bands in the at least two frequency bands partially overlap, or the adjacent frequency bands in the at least two frequency bands do not overlap; Playing one of the sub audio signals of the at least two frequency bands.
  • performing noise reduction or transparent transmission processing on the audio source to obtain the speaker drive signal includes: obtaining a fixed secondary path SP filter through a codec CODEC; filtering according to the fixed SP The device processes the audio source to obtain a filtered signal; performs noise reduction or transparent transmission processing on the filtered signal to obtain the speaker drive signal.
  • the obtaining the fixed secondary path SP filter through the codec CODEC includes: obtaining an estimated SP filter according to a preset speaker drive signal and the ear canal signal picked up by the feedback FB microphone , the ear canal signal includes the residual noise signal inside the ear canal; when the difference signal between the signal obtained by the estimated SP filter and the ear canal signal is within the set range, the estimated SP filter Determined as the fixed SP filter.
  • the method further includes: when the signal obtained by the estimated SP filter When the difference signal with the ear canal signal is within the set range, the parameters of the cascaded second-order filter are obtained according to the target frequency response of the estimated SP filter and the preset frequency division requirements; The parameters of the second-order filter are obtained from the SP cascaded second-order filter, and the SP cascaded second-order filter is used as the fixed SP filter.
  • performing noise reduction or transparent transmission processing on the audio source to obtain the speaker drive signal includes: obtaining an adaptive SP filter through a digital signal processing DSP chip; filtering according to the adaptive SP The device processes the audio source to obtain a filtered signal; performs noise reduction or transparent transmission processing on the filtered signal to obtain the speaker drive signal.
  • obtaining the adaptive SP filter through a digital signal processing DSP chip includes: obtaining a real-time noise signal; obtaining an estimated SP filter according to the audio source and the real-time noise signal; When the difference signal between the signal obtained by the estimated SP filter and the real-time noise signal is within a set range, the estimated SP filter is determined as the adaptive SP filter.
  • the acquiring a real-time noise signal includes: acquiring an external signal picked up by a feed-forward FF microphone and an ear canal signal picked up by a feedback FB microphone, where the external signal includes an external noise signal and the music Or call voice, the ear canal signal includes the residual noise signal inside the ear canal and the music or the call voice; obtain the voice signal picked up by the main microphone; subtract the external signal and the ear canal from the voice signal The signal obtains the real-time noise signal.
  • the main working frequency bands of the at least two speakers are not identical.
  • the at least two speakers include a moving coil speaker and a moving iron speaker.
  • the at least two speakers include a moving coil speaker, a moving iron speaker, a microelectromechanical system MEMS speaker, and a planar vibrating diaphragm.
  • the present application provides a playback device for a TWS headset, which is applied to the TWS headset in the first aspect;
  • the device includes: an acquisition module for acquiring an audio source, where the audio source is original music or call voice, or the audio source includes the voice signal enhanced by human voice and the original music or call voice;
  • the processing module is used to perform noise reduction or transparent transmission processing on the audio source to obtain a speaker drive signal;
  • a frequency dividing module configured to divide the speaker drive signal into sub-audio signals of at least two frequency bands, and adjacent frequency bands in the at least two frequency bands partially overlap, or, adjacent frequency bands in the at least two frequency bands The frequency bands do not overlap;
  • a playing module is configured to play one of the sub audio signals of the at least two frequency bands through at least two speakers respectively.
  • the processing module is specifically configured to obtain a fixed secondary path SP filter through a codec CODEC; process the audio source according to the fixed SP filter to obtain a filtered signal; The filtered signal is subjected to noise reduction or transparent transmission processing to obtain the speaker driving signal.
  • the processing module is specifically configured to obtain an estimated SP filter according to a preset speaker drive signal and an ear canal signal picked up by the feedback FB microphone, and the ear canal signal includes an internal ear canal signal.
  • the processing module is further configured to, when the difference signal between the signal obtained by the estimated SP filter and the ear canal signal is within a set range, according to the estimated SP
  • the target frequency response of the filter and the preset frequency division requirements obtain the parameters of the cascaded second-order filter; according to the parameters of the cascaded second-order filter, the SP cascaded second-order filter is obtained, and the SP cascaded two order filter as the fixed SP filter.
  • the processing module is specifically configured to obtain an adaptive SP filter through a digital signal processing DSP chip; process the audio source according to the adaptive SP filter to obtain a filtered signal; The filtered signal is subjected to noise reduction or transparent transmission processing to obtain the speaker driving signal.
  • the processing module is specifically configured to obtain a real-time noise signal; obtain an estimated SP filter according to the audio source and the real-time noise signal; when the signal obtained through the estimated SP filter When the difference signal with the real-time noise signal is within a set range, the estimated SP filter is determined as the adaptive SP filter.
  • the processing module is specifically configured to acquire an external signal picked up by a feedforward FF microphone and an ear canal signal picked up by a feedback FB microphone, where the external signal includes an external noise signal and the music or Call voice, the ear canal signal includes the residual noise signal inside the ear canal and the music or the call voice; obtain the voice signal picked up by the main microphone; subtract the external signal and the ear canal signal from the voice signal Obtain a signal difference; obtain the estimated SP filter according to the audio source and the signal difference.
  • the main working frequency bands of the at least two speakers are not identical.
  • the at least two speakers include a moving coil speaker and a moving iron speaker.
  • the at least two speakers include a moving coil speaker, a moving iron speaker, a microelectromechanical system MEMS speaker, and a planar vibrating diaphragm.
  • the present application provides a computer-readable storage medium, comprising a computer program, which, when executed on a computer, causes the computer to execute the method of any one of the above-mentioned second aspects.
  • the present application provides a computer program, when the computer program is executed by a computer, for performing the method of any one of the above-mentioned second aspects.
  • FIG. 1 is an exemplary structural schematic diagram of a related art TWS headset
  • FIG. 2a is an exemplary structural schematic diagram of a related art TWS headset
  • FIG. 2b is an exemplary structural schematic diagram of a related art TWS headset
  • FIG. 3 is a schematic structural diagram of an exemplary TWS headset of the present application.
  • FIG. 4 is a schematic structural diagram of an exemplary TWS headset of the present application.
  • Fig. 5a is an exemplary acquisition flow chart of the fixed SP filter of the present application
  • Fig. 5b is an exemplary acquisition flow chart of the fixed SP filter of the present application
  • FIG. 6 is a schematic structural diagram of an exemplary TWS headset of the present application.
  • FIG. 7a is an exemplary schematic diagram of the signal frequency division of the present application.
  • FIG. 7b is an exemplary schematic diagram of signal frequency division of the present application.
  • FIG. 7c is an exemplary schematic diagram of signal frequency division of the present application.
  • FIG. 7d is an exemplary schematic diagram of signal frequency division of the present application.
  • FIG. 8a is an exemplary structural schematic diagram of the TWS headset of the present application.
  • FIG. 8b is an exemplary structural schematic diagram of the TWS headset of the present application.
  • FIG. 8c is an exemplary structural schematic diagram of the TWS headset of the present application.
  • FIG. 8d is an exemplary structural schematic diagram of the TWS headset of the present application.
  • FIG. 8e is an exemplary structural schematic diagram of the TWS headset of the present application.
  • FIG. 9a is an exemplary structural schematic diagram of the TWS headset of the present application.
  • FIG. 9b is an exemplary structural schematic diagram of the TWS headset of the present application.
  • FIG. 9c is an exemplary structural schematic diagram of the TWS headset of the present application.
  • FIG. 9d is an exemplary structural schematic diagram of the TWS headset of the present application.
  • FIG. 9e is an exemplary structural schematic diagram of the TWS headset of the present application.
  • FIG. 10 is an exemplary flowchart of the playback method of the TWS headset of the present application.
  • FIG. 11 is an exemplary structural diagram of the playback device of the TWS earphone of the present application.
  • At least one (item) refers to one or more, and "a plurality” refers to two or more.
  • “And/or” is used to describe the relationship between related objects, indicating that there can be three kinds of relationships, for example, “A and/or B” can mean: only A, only B, and both A and B exist , where A and B can be singular or plural.
  • the character “/” generally indicates that the associated objects are an “or” relationship.
  • At least one item(s) below” or similar expressions thereof refer to any combination of these items, including any combination of single item(s) or plural items(s).
  • At least one (a) of a, b or c can mean: a, b, c, "a and b", “a and c", “b and c", or "a and b and c" ", where a, b, c can be single or multiple.
  • FIG. 1 is a schematic diagram of an exemplary structure of a related art TWS headset.
  • the TWS headset includes three types of microphones, namely a main microphone, a feedforward (FF) microphone and a feedback (FB) microphone, wherein the main microphone is used to pick up the human voice in the call, and the FF microphone It is used to pick up the external noise signal, and the FB microphone is used to pick up the residual noise signal inside the ear canal; the TWS headset also includes a moving coil speaker, which is used to play the processed music or call voice.
  • FF feedforward
  • FB feedback
  • FIG. 2 a is a schematic structural diagram of an exemplary TWS headset in the related art.
  • the main microphone and the FF microphone are respectively connected to the input end of the vocal enhancement filter, and the output end of the vocal enhancement filter and the audio source (including music and call voice) are superimposed by the stacker 1 to obtain a downlink link signal; the downlink signal is transmitted to one input of the stacker 2; the downlink signal is also transmitted to the input of the secondary path (SP) filter; the input of the FF microphone and the feedforward filter
  • SP secondary path
  • the output end of the feedforward filter is connected to the other input end of the stacker 2;
  • the FB microphone is connected to one input end of the stacker 3, the output end of the SP filter is connected to the other input end of the stacker 3,
  • the output end of the stacker 3 is connected to the input end of the feedback filter, and the output end of the feedback filter is connected to the third input end of the stacker 2; the output end of the stacker 2 is connected to the digital to analog
  • AHA joint controllers are respectively connected with vocal enhancement filter, feedforward filter, feedback filter and SP filter.
  • the role of the AHA joint controller In order to ensure the normal and stable operation of the TWS headset, some abnormal situations need to be handled by the TWS headset itself, so an AHA joint controller is required.
  • the AHA joint controller analyzes multiple signals to determine the current state of ANC, HT or augmented hearing (AH), and then determines whether abnormal conditions such as whistling and clipping occur, so as to implement corresponding processing.
  • the aforementioned filter parameter values are used to realize the control of the system.
  • the vocal enhancement filter, the audio source and the AHA joint controller are arranged in the digital signal processing (digital signal process, DSP) chip, and the feedforward filter, the feedback filter, the SP filter and the DAC are arranged in the codec. in the encoder (coder-decoder, CODEC).
  • DSP digital signal processing
  • CODEC encoder-decoder
  • FIG. 2 b is a schematic structural diagram of an exemplary TWS headset of the related art. As shown in Figure 2b, the difference from the structure shown in Figure 2a is that the vocal enhancement filter is moved from the DSP chip to the CODEC.
  • the TWS headset can implement the following functions:
  • the FF microphone picks up the external noise signal and generates the feedforward noise reduction signal through the feedforward filter; the FB microphone picks up the residual noise signal inside the ear canal, and generates the feedback noise reduction signal through the feedback filter.
  • the feedforward noise reduction signal, the feedback noise reduction signal and the downlink signal are superimposed to form the final speaker drive signal, and the analog speaker drive signal is generated after digital-to-analog conversion by the DAC.
  • the analog loudspeaker driving signal is played in a reverse manner in the moving coil loudspeaker, the analog audio signal can be obtained by canceling the audio signal in the space. In this way, low-frequency noise within a certain frequency band can be eliminated, thereby achieving the purpose of noise reduction.
  • the FF microphone picks up the external noise signal and generates the feedforward compensation signal through the feedforward filter; the FB microphone picks up the acoustic signal inside the ear canal and generates the feedback suppression signal through the feedback filter.
  • the feedforward compensation signal, the feedback suppression signal and the downlink signal are superimposed to form the final speaker drive signal, and the analog speaker drive signal is generated after digital-to-analog conversion by the DAC.
  • the analog speaker driving signal is played in the moving coil speaker, which can weaken or suppress low-frequency noise and compensate high-frequency signals, so as to obtain an analog audio signal that realizes acoustic compensation.
  • the "occlusion" effect of active voice listening (the wearer's own voice heard by himself) or the "stethoscope” effect of body part vibration (such as walking, chewing, scratching your head, etc. with headphones) is weakened or suppressed, passive
  • the high-frequency sound within a certain frequency band of listening sounds (such as human voice or music in the environment) is compensated, so as to achieve the purpose of transparent transmission.
  • the superimposed signal may include the feedforward compensation signal, the feedback suppression signal and the downlink signal, or may include the feedforward compensation signal and the feedback suppression signal. and either or both of the downlink signal.
  • the superimposed signal may include the aforementioned three; when the user just wants to perform noise reduction through headphones so as to be in a quiet environment, the superimposed signal may include the feed compensation signal and feedback suppression signals, excluding downlink signals.
  • Vocal enhancement and music/call concurrency When the vocal enhancement function is implemented on the DSP side, the FF microphone and main microphone signals are sent to the vocal enhancement filter for processing to obtain a signal in which ambient noise is suppressed and the human voice is preserved. Then the signal is down-streamed to the CODEC, superimposed with the signals output by the feedforward filter and the feedback filter respectively to obtain the speaker drive signal, and after digital-to-analog conversion by the DAC, the analog speaker drive signal is output for the moving coil speaker to play.
  • the estimated signal played by the moving coil speaker can be subtracted from the signal picked up by the FB microphone, In this way, this part of the signal will not be noise-reduced by the feedback filter, thus realizing the concurrency of vocal enhancement and music/calling, and the ANC/HT function can also be realized on this basis.
  • the realization of vocal enhancement in DSP is a guarantee of computational power overhead.
  • the noise reduction effect and the playback effect after processing are stable, but the delay is long, and it sounds like a reverberation.
  • the human voice enhancement is implemented in the CODEC.
  • the algorithm is relatively fixed and simple, the time delay is short, and the reverberation sense is low, but the noise reduction effect is limited.
  • the structure of the TWS earphone shown in Figure 1 to Figure 2b for high-frequency music, especially for music signals in the frequency band above 8.5kHz, the playback sound quality may be seriously damaged, affecting the playback effect of music; for high-frequency speech, especially for 8.5kHz Voice signals in the above frequency bands are not supported, resulting in limited bandwidth for voice calls.
  • the present application provides a speaker structure of a TWS earphone, which can improve the above technical problems.
  • FIG. 3 is a schematic structural diagram of an exemplary TWS headset of the present application.
  • the TWS headset 30 includes three types of microphones, namely the main microphone 31 , the FF microphone 32 and the FB microphone 33 , wherein the main microphone 31 is used to pick up the human voice during the call, and the FF microphone 32 is used to pick up the external noise signal , the FB microphone 33 is used to pick up the residual noise signal inside the ear canal.
  • the TWS earphone 30 also includes a moving coil speaker 34a and a moving iron speaker 34b.
  • the main working frequency band of the moving coil speaker 34a is less than 8.5kHz, and the main working frequency band of the moving iron speaker 34b is greater than 8.5kHz. It should be noted that this application does not specifically limit the number of speakers, as long as there are at least two speakers, and the at least two speakers may have different main working frequency bands of each speaker, or may also have the main operating frequency bands of some speakers.
  • the working frequency band is the same, and the main working frequency band of another part of the speakers is different from the main working frequency band of the aforementioned part of the speakers.
  • the main working frequency band of speaker 1 is the same as that of speaker 2, and the main working frequency band of speaker 3 is different from that of speaker 1 and speaker 2.
  • the main working frequency bands of the speaker 1, the speaker 2 and the speaker 3 are all different.
  • the at least two speakers may also include a moving coil speaker, a moving iron speaker, a micro-electro-mechanical systems (MEMS) MEMS speaker and a planar vibrating diaphragm, wherein the main working frequency band of the moving coil speaker is Less than 8.5kHz, the main working frequency band of the moving iron speaker is greater than 8.5kHz, the main working frequency band of the MEMS speaker depends on the application form, the main working frequency band of the in-ear headphone is the full frequency band, and the main working frequency band of the headphone is below 7kHz. Weak, the main working frequency band is high frequency above 7kHz, and the main working frequency band of the plane vibrating diaphragm is 10kHz ⁇ 20kHz.
  • MEMS micro-electro-mechanical systems
  • FIG. 4 is a schematic structural diagram of an exemplary TWS headset of the present application.
  • the TWS earphone 40 includes: an audio signal processing path 41, a frequency divider 42 and at least two speakers 43; wherein, the output end of the audio signal processing path 41 is connected to the input end of the frequency divider 42; frequency division The output end of the speaker 42 is connected to at least two speakers 43 .
  • the audio signal processing path 41 is configured to output a speaker drive signal after performing noise reduction or transparent transmission processing on the audio source; the audio source is the original music or call voice; music or call voice.
  • the frequency divider 42 is configured to divide the speaker drive signal into sub-audio signals of at least two frequency bands, the at least two frequency bands correspond to the main working frequency bands of the at least two speakers 43; the adjacent frequency band parts in the at least two frequency bands Overlapping, or at least two adjacent bands in the two bands do not overlap. At least two speakers 43 are configured to play corresponding sub audio signals.
  • the frequency band of the processed speaker driving signal corresponds to the frequency band of the audio source, and may include the whole frequency range of low, medium and high. However, because the main working frequency band of a single speaker may only cover part of the low, middle and high frequency bands, a single speaker cannot operate in the entire frequency band. reflect high-quality sound.
  • the frequency divider 42 of the present application may be configured to divide the frequency of the speaker drive signal based on the main operating frequency bands of the at least two speakers, to obtain sub-audio signals of at least two frequency bands corresponding to the main operating frequency bands of the at least two speakers, respectively, Then let each speaker play the sub audio signal of the corresponding frequency band, so that the speaker maintains the best frequency response when playing the sub audio signal transmitted to it.
  • the frequency divider 42 can be controlled to divide the frequency of the speaker driving signal in a preset manner by setting the parameters of the frequency divider 42 .
  • the above-mentioned audio signal processing path 41 may adopt the structure shown in FIG. 2 a or FIG. 2 b .
  • the SP filter is set in the codec (coder-decoder, CODEC), and the fixed value is obtained through the CODEC. SP filter.
  • the CODEC can obtain the estimated SP filter according to the preset speaker drive signal and the ear canal signal picked up by the FB microphone.
  • the ear canal signal includes the residual noise signal inside the ear canal; when the signal obtained by the estimated SP filter and the ear canal signal When the difference signal of is within the set range, the estimated SP filter is determined as a fixed SP filter.
  • Fig. 5a is an exemplary acquisition flow chart of the fixed SP filter of the present application.
  • a speaker driving signal x[n] is preset, and after passing through the frequency divider, at least two speakers are pushed to emit sound.
  • the sound is transmitted to the FB microphone, picked up by the FB microphone, and converted into a digital signal y[n].
  • the transfer function of the SP filter can be assumed to be a high-order FIR filter, which can be iteratively modeled by the least mean square error (LMS) algorithm.
  • LMS least mean square error
  • the transfer function S(z) of the real SP filter is unknown, but its All the information is contained in the speaker drive signal x[n] and the digital signal y[n] picked up by the FB microphone, so it can be passed through a higher order FIR filter Model S(z) as an estimated SP filter.
  • input x[n] get for y[n] and Calculate the difference to get the error signal e[n].
  • Fig. 5b is an exemplary acquisition flow chart of the fixed SP filter of the present application. As shown in Fig. 5b, when the difference signal between the signal obtained by the estimated SP filter and the ear canal signal is within the set range, according to the estimation The parameters of the cascaded second-order filter are obtained from the target frequency response of the SP filter and the preset frequency division requirements; the SP cascaded second-order filter is obtained according to the parameters of the cascaded second-order filter, and the SP cascaded second-order filter is as a fixed SP filter.
  • FIR finite-length unit impulse response
  • IIR infinite impulse response
  • CODEC is based on the FIR filter that has been obtained Calculate the target frequency response.
  • N the filter order
  • k the frequency point number.
  • the IIR filter parameters are obtained from the conversion process of the FIR filter to the IIR filter, and the process requires the target responses of the two to be as consistent as possible.
  • B[k] and A[k] are the complex frequency responses of the IIR filter coefficients b and a respectively, and the calculation method is the same as that of formula (1).
  • a set of IIR filter parameters under the minimum mean square can be found, and the IIR filter obtained from this set of IIR filter parameters can be used as the final fixed SP filter and put into the CODEC for hardening implementation.
  • the above-mentioned audio signal processing path 41 may adopt the structure shown in FIG. 6 .
  • the SP filter is set in a digital signal processing (digital signal process, DSP) chip, and the adaptive filter is obtained through the DSP chip. SP filter.
  • DSP digital signal processing
  • the DSP chip can obtain the real-time noise signal, and obtain the estimated SP according to the audio source (the audio source is the original music or voice of the call; or, the audio source includes the voice signal that has been enhanced by the human voice and the original music or voice of the call) and the real-time noise signal.
  • the filter determines the estimated SP filter as an adaptive SP filter when the difference signal between the signal obtained by the estimated SP filter and the real-time noise signal is within the set range.
  • the above-mentioned acquisition of the real-time noise signal may be to acquire the external signal picked up by the FF microphone (the external signal includes the external noise signal and music or voice of a call) and the ear canal signal picked up by the FB microphone (the ear canal signal includes the residual noise signal inside the ear canal and music. Or call voice), and then obtain the voice signal picked up by the main microphone, and subtract the external signal and the ear canal signal from the voice signal to obtain a real-time noise signal.
  • x[n] represents an audio source
  • y[n] fb[n]-ff[n]*A(z)-fb[n-1]*C(z)
  • fb[n] represents the voice signal picked up by the main microphone
  • ff[n]*A(z) It represents the external signal picked up by the FF microphone
  • fb[n-1]*C(z) represents the ear canal signal picked up by the FB microphone.
  • the noise reduction or transparent transmission function of the TWS headset will be concurrent with the music or the voice of the call.
  • the signal picked up by the FB microphone cannot be directly estimated by the SP filter, and it is necessary to remove the influence of other signals.
  • Modeling analysis is to obtain real-time noise signal, so the SP filter obtained at this time can be adapted to the real-time situation of the audio source and noise signal instead of being fixed.
  • the SP filter (including fixed SP filter or adaptive SP filter) is determined by the above method, which can not only realize the function of noise reduction or transparent transmission, but also prevent the transparent transmission or noise reduction technology in the process of playing music or passing through. Also remove music or call sounds.
  • the TWS earphone includes two speakers: a moving coil speaker and a moving iron speaker.
  • the dotted line represents the frequency response curve of the moving coil speaker
  • the single-dotted line represents the frequency response curve of the moving iron speaker
  • the solid line represents the crossover line.
  • FIG. 7a is an exemplary schematic diagram of signal frequency division of the present application.
  • the frequency divider 42 is configured to attenuate the dynamic speaker in the non-main operating frequency band, and keep the dynamic speaker in the main operating frequency band.
  • the power of the moving iron speaker is attenuated in the non-main working frequency band of the moving iron speaker, the power of the moving iron speaker in the main working frequency band is retained, and the frequency is divided at the intersection frequency of the attenuation frequency band of the moving coil speaker and the moving iron speaker. Get the sub audio signal of the two frequency bands.
  • FIG. 7b is an exemplary schematic diagram of signal frequency division of the application.
  • the frequency divider 42 is configured to align and attenuate in the non-main working frequency band of the moving iron speaker, and retain the moving iron speaker in the main working frequency band. power, and divide the frequency at a certain frequency point in the attenuation frequency band of the moving iron speaker to obtain sub-audio signals of two frequency bands.
  • Fig. 7c is an exemplary schematic diagram of the frequency division of the signal of the present application.
  • the frequency divider 42 is configured to perform two-stage attenuation in the non-main operating frequency band of the moving coil speaker, leaving the moving coil speaker at The power of the main working frequency band is attenuated in two stages in the non-main working frequency band of the moving iron speaker, retaining the power of the moving iron speaker in the main working frequency band, and attenuating the first and second attenuation bands of the moving iron speaker
  • a frequency division is performed at the transition frequency point of the frequency band, and a frequency division is performed at the transition frequency point of the first attenuation frequency band and the second attenuation frequency band of the moving coil speaker to obtain sub audio signals of three frequency bands.
  • the TWS earphone includes three speakers: a moving coil speaker, a moving iron speaker, and a MEMS speaker.
  • the dotted line represents the frequency response curve of the moving coil speaker
  • the single-dotted line represents the frequency response curve of the moving iron speaker
  • the double-dotted line represents the frequency response curve of the MEMS speaker
  • the solid line represents the crossover line.
  • Fig. 7d is an exemplary schematic diagram of the frequency division of the signal of the present application.
  • the frequency divider 42 is configured to attenuate the MEMS speaker in the non-main operating frequency band , retain the power of the MEMS speaker in the main working frequency band, and perform a frequency division at the intersection frequency point in the respective attenuation frequency bands of the moving iron speaker and the MEMS speaker, and obtain a total of four frequency bands of sub-audio signals.
  • FIGS. 7 a to 7 d are several examples in which the frequency divider 42 divides the speaker driving signal, and the present application does not limit the specific frequency division method of the frequency divider 42 .
  • At least two speakers are set in the TWS earphone of the present application, and the main working frequency bands of the at least two speakers are not identical.
  • the frequency divider can divide the speaker drive signal into sub-audio signals of at least two frequency bands, and the adjacent frequency bands in the at least two frequency bands partially overlap or do not overlap, so that each sub-audio signal is respectively transmitted to the speakers with matching frequency bands,
  • the aforementioned frequency band matching can mean that the main working frequency band of the speaker covers the frequency band of the sub-audio signal transmitted to it, so that the speaker maintains the best frequency response when playing the sub-audio signal transmitted to it. It reflects high sound quality and supports ultra-bandwidth voice calls.
  • FIG. 8a is a schematic structural diagram of an exemplary TWS headset of the present application.
  • the TWS earphone 40 further includes: a first DAC 44 .
  • the input end of the first DAC 44 is connected with the output end of the audio signal processing path 41, and the output end of the first DAC 44 is connected with the input end of the frequency divider 42.
  • the first DAC 44 is configured to convert the speaker drive signal from digital to analog.
  • the frequency divider 42 is an analog frequency divider.
  • the speaker driving signal is obtained, if the signal does not pass through the DAC, it is in digital form, but the signal played by the speaker needs to be in analog form, so it can pass the first DAC first.
  • the speaker driving signal in digital form is converted into a speaker driving signal in analog form, and then the analog frequency divider is used to divide the frequency of the speaker driving signal in analog form into sub audio signals of at least two frequency bands.
  • This embodiment adopts a structure of converting first and then dividing the frequency.
  • FIG. 8b is an exemplary structural schematic diagram of the TWS headset of the present application. As shown in Fig. 8b, the structure of this embodiment is a more detailed implementation of the structure shown in Fig. 8a.
  • the main microphone 601 and the FF microphone 602 in the TWS earphone 60 are respectively connected to the input end of the vocal enhancement filter 603 , and the output end of the vocal enhancement filter 603 and the audio source 604 (including music and call speech)
  • the downlink signal is obtained after superposition; the downlink signal is transmitted to an input of the stacker 2; the downlink signal is also transmitted to the input of the SP filter 605; connected, the output end of the feedforward filter 606 is connected to the other input end of the stacker 2; the FB microphone 607 is connected to one input end of the stacker 3, and the output end of the SP filter 605 is connected to the other input end of the stacker 3 connected, the output end of the stacker 3 is connected to the input end of the feedback filter 608, and the output end of the feedback filter 608 is connected to the third input end of the stacker 2; the output end of the stacker 2 is connected to the digital-to-analog converter (DAC) ) 609 is connected to the input end, the output end of the DAC
  • the TWS earphone 60 in this embodiment is provided with two speakers, namely a moving coil speaker 611a and a moving iron speaker 611b.
  • the main working frequency band of the moving coil speaker 611a is below 8.5 kHz, and the main working frequency band of the moving iron speaker 611b is above 8.5 kHz.
  • the frequency divider 42 can be set to divide the analog speaker drive signal into a sub audio signal below 8.5kHz and a sub audio signal above 8.5kHz, and the moving coil speaker 611a can maintain the optimal playback of the sub audio signal below 8.5kHz Frequency response, the moving iron speaker 611b can maintain the best frequency response by playing sub-audio signals above 8.5kHz, so that the TWS earphone 60 can not only reflect high sound quality in various frequency bands of the audio source, but also support ultra-bandwidth voice calls.
  • FIG. 8c is an exemplary schematic structural diagram of the TWS headset of the present application. As shown in FIG. 8c, the structure of this embodiment is another more detailed implementation of the structure shown in FIG. 8a.
  • the difference from the structure shown in FIG. 8b is that the output end of the analog frequency divider 610 is connected to the moving coil speaker 611a, the moving iron speaker 611b, the MEMS speaker 611c and the planar vibrating diaphragm 611d.
  • the TWS earphone 60 in this embodiment is provided with four speakers, namely a moving coil speaker 611a, a moving iron speaker 611b, a MEMS speaker 611c, and a plane vibration diaphragm 611d.
  • the main working frequency of the moving coil speaker 611a is below 8.5 kHz, and the moving iron
  • the main working frequency band of the speaker 611b is above 8.5kHz, and the main working frequency band of the MEMS speaker 611c depends on the application form.
  • the main working frequency band of the planar vibrating diaphragm 611d is 10 kHz to 20 kHz.
  • the frequency divider 42 can be set to divide the analog speaker driving signal into four sub-frequency bands, and the moving coil speaker 611a plays 8.5 kHz.
  • the following sub-audio signals can maintain the best frequency response
  • the moving iron speaker 611b can maintain the best frequency response by playing the sub-audio signal above 8.5kHz
  • the MEMS speaker 611c can maintain the best frequency response by playing the sub-audio signal above 7kHz
  • the plane vibrates The diaphragm 611d can maintain the best frequency response by playing the sub-audio signal above 10 kHz, so that the TWS earphone 60 can not only reflect the high sound quality in each frequency band of the audio source, but also support ultra-bandwidth voice calls.
  • the vocal enhancement filter 603, the audio source 604 and the AHA joint controller 612 are arranged in a digital signal processing (digital signal process, DSP) chip, the feedforward filter 606, the feedback filter 608, and the SP filter 605.
  • DSP digital signal processing
  • the DAC 609 is set in the codec (coder-decoder, CODEC).
  • FIG. 8d is an exemplary schematic structural diagram of the TWS headset of the present application. As shown in Fig. 8d, the structure of this embodiment is a more detailed implementation of the structure shown in Fig. 8a.
  • the difference from the structure shown in Fig. 8b is that the vocal enhancement filter 603 is moved from the DSP chip to the CODEC.
  • FIG. 8e is a schematic structural diagram of an exemplary TWS headset of the present application. As shown in FIG. 8e, the structure of this embodiment is a more detailed implementation of the structure shown in FIG. 8a.
  • the difference from the structure shown in Fig. 8c is that the vocal enhancement filter 603 is moved from the DSP chip to the CODEC.
  • FIG. 9a is a schematic structural diagram of an exemplary TWS headset of the present application.
  • the TWS earphone 40 further includes: at least two second DACs 45 .
  • the input ends of the at least two second DACs 45 are connected to the output ends of the frequency divider 42, and the output ends of the at least two second DACs 45 are respectively connected to one of the at least two speakers 43.
  • the second DAC 45 is configured to convert one of the sub-audio signals of at least two frequency bands from a digital form to an analog form.
  • the frequency divider 42 is a digital frequency divider.
  • the speaker driving signal After the speaker driving signal is obtained, if the signal does not pass through the DAC, it is in digital form, but the signal played by the speaker needs to be in analog form, so it can be divided by digital frequency first.
  • the device divides the speaker driving signal in digital form into sub audio signals in at least two frequency bands, and then converts the sub audio signals in digital form transmitted to it into sub audio in analog form through at least two second DACs respectively. Signal.
  • This embodiment adopts the structure of frequency division first and then conversion.
  • FIG. 9b is an exemplary schematic structural diagram of the TWS headset of the present application. As shown in FIG. 9b, the structure of this embodiment is a more detailed implementation of the structure shown in FIG. 9a.
  • the main microphone 701 and the FF microphone 702 in the TWS earphone 70 are respectively connected to the input end of the vocal enhancement filter 703 , and the output end of the vocal enhancement filter 703 and the audio source 704 (including music and talking voice) are connected to the
  • the downlink signal is obtained after superposition; the downlink signal is transmitted to one input of the stacker 2; the downlink signal is also transmitted to the input of the SP filter 705; connected, the output end of the feedforward filter 706 is connected to the other input end of the stacker 2; the FB microphone 707 is connected to one input end of the stacker 3, and the output end of the SP filter 705 is connected to the other input end of the stacker 3 connected, the output end of the superimposed device 3 is connected to the input end of the feedback filter 708, and the output end of the feedback filter 708 is connected to the third input end of the superimposed device 2; The input end is connected, and the output end of the digital frequency divider 709 is connected to the input ends of the two DACs 710a and 7
  • FIG. 9c is an exemplary schematic structural diagram of the TWS headset of the present application. As shown in Fig. 9c, the structure of this embodiment is a more detailed implementation of the structure shown in Fig. 9a.
  • the TWS earphone 70 in this embodiment is provided with four speakers, namely a moving coil speaker 711a, a moving iron speaker 711b, a MEMS speaker 711c, and a plane vibration diaphragm 711d.
  • the main working frequency of the moving coil speaker 711a is below 8.5 kHz, and the moving iron
  • the main working frequency band of the speaker 711b is above 8.5kHz, and the main working frequency band of the MEMS speaker 711c depends on the application form.
  • the main working frequency band of the planar vibrating diaphragm 711d is 10 kHz to 20 kHz.
  • the frequency divider 42 can be set to divide the analog speaker driving signal into four sub-frequency bands, and the moving coil speaker 711a plays 8.5 kHz.
  • the following sub-audio signals can maintain the best frequency response
  • the moving iron speaker 711b can maintain the best frequency response by playing the sub-audio signal above 8.5kHz
  • the MEMS speaker 711c can maintain the best frequency response by playing the sub-audio signal above 7kHz
  • the plane vibrates The diaphragm 711d can maintain the best frequency response by playing sub-audio signals above 10 kHz, so that the TWS earphone 70 can not only reflect high sound quality in various frequency bands of the audio source, but also support ultra-bandwidth voice calls.
  • the vocal enhancement filter 703, the audio source 704 and the AHA joint controller 712 are arranged in the DSP chip, and the feedforward filter 706, the feedback filter 708, the SP filter 705 and the DAC 709 are arranged in the CODEC.
  • FIG. 9d is an exemplary schematic structural diagram of the TWS headset of the present application. As shown in FIG. 9d, the structure of this embodiment is a more detailed implementation of the structure shown in FIG. 9a.
  • the difference from the structure shown in Fig. 9b is that the vocal enhancement filter 703 is moved from the DSP chip to the CODEC.
  • FIG. 9e is an exemplary schematic structural diagram of the TWS headset of the present application. As shown in FIG. 9e, the structure of this embodiment is a more detailed implementation of the structure shown in FIG. 9a.
  • the difference from the structure shown in Fig. 9c is that the vocal enhancement filter 703 is moved from the DSP chip to the CODEC.
  • FIG. 10 is an exemplary flowchart of a method for playing a TWS headset of the present application. As shown in FIG. 10 , the method of this embodiment may be applied to the TWS headset in the above-mentioned embodiments. The method can include:
  • Step 1001 Acquire an audio source.
  • the audio source is original music or call voice, that is, the audio source may be music, video sound, etc. that the user is listening to with the headset, or the call voice when the user is making a call with the headset.
  • the audio source may come from the player of the electronic device.
  • the audio source includes the voice signal enhanced by the human voice and the original music or voice of the call, that is, in addition to the music or voice of the call in the above two cases, the audio source can also superimpose the voice of the outside world enhanced by the human voice. Signal.
  • the external speech signal processed by the human voice enhancement can be obtained by the human voice enhancement filter in the structure shown in FIG. 2 a or FIG. 2 b , which will not be repeated here.
  • Step 1002 performing noise reduction or transparent transmission processing on the audio source to obtain a speaker driving signal.
  • the fixed secondary path SP filter can be obtained through the CODEC, then the audio source is processed according to the fixed SP filter to obtain a filtered signal, and then the filtered signal is subjected to noise reduction or transparent transmission processing to obtain the speaker driver Signal.
  • the CODEC can obtain the estimated SP filter according to the preset speaker drive signal and the ear canal signal picked up by the feedback FB microphone.
  • the ear canal signal includes the residual noise signal inside the ear canal.
  • the cascaded second-order filter is obtained according to the target frequency response of the estimated SP filter and the preset frequency division requirement. According to the parameters of the cascaded second-order filter, the SP cascaded second-order filter is obtained, and the SP cascaded second-order filter is used as a fixed SP filter.
  • the adaptive SP filter can be obtained through the DSP chip, and then the audio source is processed according to the adaptive SP filter to obtain a filtered signal, and then the filtered signal is subjected to noise reduction or transparent transmission processing to obtain the speaker driver Signal.
  • the DSP chip can obtain the real-time noise signal, and obtain the estimated SP filter according to the audio source and the real-time noise signal.
  • the estimated SP The filter is determined to be an adaptive SP filter.
  • the DSP chip can first obtain the external signal picked up by the FF microphone and the ear canal signal picked up by the FB microphone.
  • the external signal includes the external noise signal and music or call voice
  • the ear canal signal includes the residual noise signal inside the ear canal and music. Or call voice, and then obtain the voice signal picked up by the main microphone, and finally subtract the external signal and the ear canal signal from the voice signal to obtain a real-time noise signal.
  • Step 1003 Divide the speaker driving signal into sub-audio signals of at least two frequency bands.
  • the frequency band of the processed speaker driving signal corresponds to the frequency band of the audio source, and may include the whole frequency range of low, medium and high. However, because the main working frequency band of a single speaker may only cover part of the low, middle and high frequency bands, a single speaker cannot operate in the entire frequency band. reflect high-quality sound.
  • the frequency divider of the present application may be configured to divide the frequency of the speaker drive signal based on the main working frequency bands of the at least two speakers, so as to obtain sub-audio signals of at least two frequency bands corresponding to the main working frequency bands of the at least two speakers, and then Let each speaker play the sub audio signal of the corresponding frequency band, so that the speaker maintains the best frequency response when playing the sub audio signal transmitted to it.
  • Adjacent frequency bands in the aforementioned at least two frequency bands partially overlap, or adjacent frequency bands in the at least two frequency bands do not overlap. For example, a crossover divides the speaker drive signal into high and low frequency bands, the high and low frequency bands are completely separated, and there is no overlap; or, the high and low frequency bands partially overlap.
  • the frequency divider divides the speaker driving signal into three frequency bands: high, middle and low frequency.
  • the high frequency band and the middle frequency band are completely separated without overlapping.
  • the present application can control the frequency divider 42 to divide the frequency of the speaker driving signal in a preset manner by setting the parameters of the frequency divider 42 .
  • frequency division reference may be made to FIG. 6a to FIG. 6d , which will not be repeated here.
  • Step 1004 Play one of the sub audio signals of at least two frequency bands through at least two speakers respectively.
  • At least two speakers are set in the TWS earphone of the present application, and the main working frequency bands of the at least two speakers are not identical.
  • the frequency divider can divide the speaker drive signal into sub-audio signals of at least two frequency bands, and the adjacent frequency bands in the at least two frequency bands partially overlap or do not overlap, so that each sub-audio signal is respectively transmitted to the speakers with matching frequency bands,
  • the aforementioned frequency band matching can mean that the main working frequency band of the speaker covers the frequency band of the sub-audio signal transmitted to it, so that the speaker maintains the best frequency response when playing the sub-audio signal transmitted to it. It reflects high sound quality and supports ultra-bandwidth voice calls.
  • FIG. 11 is an exemplary structural diagram of a playback device for a TWS headset of the present application.
  • the device 1100 of this embodiment can be applied to the TWS headset in the above-mentioned embodiment, and the device 1100 includes: an acquisition module 1101, Processing module 1102 , frequency dividing module 1103 and playing module 1104 . in,
  • the acquisition module 1101 is used to acquire an audio source, the audio source is the original music or call voice, or, the audio source includes the voice signal enhanced by the human voice and the original music or call voice; processing module 1102 , used to perform noise reduction or transparent transmission processing on the audio source to obtain a speaker drive signal; the frequency division module 1103 is used to divide the speaker drive signal into sub-audio signals of at least two frequency bands, the at least two Adjacent frequency bands in the frequency bands partially overlap, or, adjacent frequency bands in the at least two frequency bands do not overlap; the playing module 1104 is configured to play the sub audio signals of the at least two frequency bands through at least two speakers respectively. one.
  • the processing module 1102 is specifically configured to obtain a fixed secondary path SP filter through a codec CODEC; process the audio source according to the fixed SP filter to obtain a filtered signal;
  • the speaker driving signal is obtained by performing noise reduction or transparent transmission processing on the filtered signal.
  • the processing module 1102 is specifically configured to obtain an estimated SP filter according to a preset speaker drive signal and an ear canal signal picked up by the feedback FB microphone, where the ear canal signal includes the ear canal Internal residual noise signal and the music or talking speech; when the difference signal between the signal obtained by the estimated SP filter and the ear canal signal is within a set range, the estimated SP filter is determined as the fixed SP filter.
  • the processing module 1102 is further configured to, when the difference signal between the signal obtained by the estimated SP filter and the ear canal signal is within a set range, according to the estimation
  • the parameters of the cascaded second-order filter are obtained from the target frequency response of the SP filter and the preset frequency division requirements; the SP cascaded second-order filter is obtained according to the parameters of the cascaded second-order filter, and the SP is cascaded A second order filter acts as the fixed SP filter.
  • the processing module 1102 is specifically configured to obtain an adaptive SP filter through a digital signal processing DSP chip; process the audio source according to the adaptive SP filter to obtain a filtered signal;
  • the speaker driving signal is obtained by performing noise reduction or transparent transmission processing on the filtered signal.
  • the processing module 1102 is specifically configured to obtain a real-time noise signal; obtain an estimated SP filter according to the audio source and the real-time noise signal; When the difference signal between the signal and the real-time noise signal is within a set range, the estimated SP filter is determined as the adaptive SP filter.
  • the processing module 1102 is specifically configured to acquire the external signal picked up by the feedforward FF microphone and the ear canal signal picked up by the feedback FB microphone, where the external signal includes external noise signals and the music Or call voice, the ear canal signal includes the residual noise signal inside the ear canal and the music or the call voice; obtain the voice signal picked up by the main microphone; subtract the external signal and the ear canal from the voice signal The signal obtains a signal difference; the estimated SP filter is obtained from the audio source and the signal difference.
  • the main working frequency bands of the at least two speakers are not identical.
  • the at least two speakers include a moving coil speaker and a moving iron speaker.
  • the at least two speakers include a moving coil speaker, a moving iron speaker, a microelectromechanical system MEMS speaker, and a planar vibrating diaphragm.
  • the apparatus of this embodiment can be used to execute the technical solution of the method embodiment shown in FIG. 10 , and its implementation principle and technical effect are similar, and details are not repeated here.
  • each step of the above method embodiments may be completed by a hardware integrated logic circuit in a processor or an instruction in the form of software.
  • the processor can be a general-purpose processor, a digital signal processor (DSP), an application-specific integrated circuit (ASIC), a field programmable gate array (FPGA), or other Programming logic devices, discrete gate or transistor logic devices, discrete hardware components.
  • DSP digital signal processor
  • ASIC application-specific integrated circuit
  • FPGA field programmable gate array
  • a general purpose processor may be a microprocessor or the processor may be any conventional processor or the like.
  • the steps of the methods disclosed in the embodiments of the present application may be directly embodied as executed by a hardware coding processor, or executed by a combination of hardware and software modules in the coding processor.
  • the software modules may be located in random access memory, flash memory, read-only memory, programmable read-only memory or electrically erasable programmable memory, registers and other storage media mature in the art.
  • the storage medium is located in the memory, and the processor reads the information in the memory, and completes the steps of the above method in combination with its hardware.
  • the memory mentioned in the above embodiments may be volatile memory or non-volatile memory, or may include both volatile and non-volatile memory.
  • the non-volatile memory may be read-only memory (ROM), programmable read-only memory (PROM), erasable programmable read-only memory (EPROM), electrically programmable Erase programmable read-only memory (electrically EPROM, EEPROM) or flash memory.
  • Volatile memory may be random access memory (RAM), which acts as an external cache.
  • RAM random access memory
  • DRAM dynamic random access memory
  • SDRAM synchronous DRAM
  • SDRAM double data rate synchronous dynamic random access memory
  • ESDRAM enhanced synchronous dynamic random access memory
  • SLDRAM synchronous link dynamic random access memory
  • direct rambus RAM direct rambus RAM
  • the disclosed system, apparatus and method may be implemented in other manners.
  • the apparatus embodiments described above are only illustrative.
  • the division of the units is only a logical function division. In actual implementation, there may be other division methods.
  • multiple units or components may be combined or Can be integrated into another system, or some features can be ignored, or not implemented.
  • the shown or discussed mutual coupling or direct coupling or communication connection may be through some interfaces, indirect coupling or communication connection of devices or units, and may be in electrical, mechanical or other forms.
  • the units described as separate components may or may not be physically separated, and components displayed as units may or may not be physical units, that is, may be located in one place, or may be distributed to multiple network units. Some or all of the units may be selected according to actual needs to achieve the purpose of the solution in this embodiment.
  • each functional unit in each embodiment of the present application may be integrated into one processing unit, or each unit may exist physically alone, or two or more units may be integrated into one unit.
  • the functions, if implemented in the form of software functional units and sold or used as independent products, may be stored in a computer-readable storage medium.
  • the technical solution of the present application can be embodied in the form of a software product in essence, or the part that contributes to the prior art or the part of the technical solution.
  • the computer software product is stored in a storage medium, including Several instructions are used to cause a computer device (personal computer, server, or network device, etc.) to execute all or part of the steps of the methods described in the various embodiments of the present application.
  • the aforementioned storage medium includes: U disk, mobile hard disk, read-only memory (ROM), random access memory (RAM), magnetic disk or optical disk and other media that can store program codes .

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Multimedia (AREA)
  • Otolaryngology (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
PCT/CN2022/083464 2021-04-28 2022-03-28 Tws耳机和tws耳机的播放方法及装置 WO2022227982A1 (zh)

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