WO2018010375A1 - Method and device for realising karaoke function through earphone, and earphone - Google Patents
Method and device for realising karaoke function through earphone, and earphone Download PDFInfo
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- WO2018010375A1 WO2018010375A1 PCT/CN2016/110995 CN2016110995W WO2018010375A1 WO 2018010375 A1 WO2018010375 A1 WO 2018010375A1 CN 2016110995 W CN2016110995 W CN 2016110995W WO 2018010375 A1 WO2018010375 A1 WO 2018010375A1
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L13/00—Speech synthesis; Text to speech systems
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/02—Circuits for transducers, loudspeakers or microphones for preventing acoustic reaction, i.e. acoustic oscillatory feedback
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L13/00—Speech synthesis; Text to speech systems
- G10L13/02—Methods for producing synthetic speech; Speech synthesisers
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
- H04M1/724—User interfaces specially adapted for cordless or mobile telephones
- H04M1/72403—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality
- H04M1/7243—User 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
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
- H04M1/724—User interfaces specially adapted for cordless or mobile telephones
- H04M1/72403—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality
- H04M1/7243—User 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
- H04M1/72433—User 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
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
- G10L2021/02087—Noise filtering the noise being separate speech, e.g. cocktail party
Definitions
- the present invention relates to the field of earphone technology, and more particularly, to a method, device and earphone for implementing a karaoke function through an earphone.
- a method for implementing a karaoke function through a headset including:
- the method further includes:
- the digital voice signal includes a portion whose amplitude exceeds the first set value, and if so, filtering out the vocal signal in the digital music signal received by the earphone to obtain a background music signal; if not, the The digital music signal is subjected to digital-to-analog conversion processing to obtain an analog music signal, and the analog music signal is transmitted to the earpiece of the earphone.
- the method further includes:
- the voice signal collected by the microphone of the receiving headset includes:
- performing beamforming processing on the first digital voice signal and the second digital voice signal, and obtaining the digital voice signal includes:
- the digital speech signal is obtained by eliminating a portion of the signal obtained by the beamforming process that has an amplitude exceeding a second set value.
- an apparatus for implementing a karaoke function through a headset including:
- a receiving module configured to receive a voice signal collected by a microphone of the earphone
- An analog-to-digital conversion module configured to perform analog-to-digital conversion processing on the voice signal to obtain a digital voice signal
- a superimposing module configured to superimpose the background music signal and the digital speech signal to obtain a digital superimposed signal
- the first digital-to-analog conversion module is configured to perform digital-to-analog conversion processing on the digital superimposed signal to obtain an analog superimposed signal, and send the analog superimposed signal to an earpiece of the earphone.
- the device further includes a detection module and a second digital-to-analog conversion module, the detection module is configured to detect whether the digital voice signal includes detecting whether the digital voice signal includes an amplitude exceeding a first setting. a portion of the value, if yes, controlling the filtered vocal signal module to filter out a vocal signal in the digital music signal received by the earphone to obtain a background music signal; if not, controlling the second digital to analog conversion module And performing digital-to-analog conversion processing on the digital music signal to obtain an analog music signal, and transmitting the analog music signal to an earpiece of the earphone.
- the detection module is configured to detect whether the digital voice signal includes detecting whether the digital voice signal includes an amplitude exceeding a first setting. a portion of the value, if yes, controlling the filtered vocal signal module to filter out a vocal signal in the digital music signal received by the earphone to obtain a background music signal; if not, controlling the second digital to analog conversion module And performing digital-to-analog conversion processing on the digital music
- the device further includes:
- a first reverberation module configured to add a reverberation signal corresponding to the background music signal to the background music signal
- a second plus reverberation module configured to add a reverberation signal corresponding to the digital voice signal to the digital voice signal.
- the receiving module includes:
- a first receiving unit configured to receive a voice signal collected by a microphone mounted on a earphone as a first voice signal
- a second receiving unit configured to receive a voice signal collected by a microphone mounted on another earphone as a second voice signal
- the first digital to analog conversion module includes:
- a digital-to-analog conversion unit configured to perform analog-to-digital conversion processing on the first voice signal and the second voice signal, respectively, to obtain a corresponding first digital voice signal and a second digital voice signal;
- a beam forming unit configured to perform beamforming processing on the first digital voice signal and the second digital voice signal to obtain the digital voice signal.
- the beam forming unit includes:
- a beamforming subunit configured to perform beamforming processing on the first digital voice signal and the second digital voice signal
- a eliminating subunit configured to cancel a portion of the signal obtained by the beamforming process and having an amplitude exceeding a second set value, to obtain the digital voice signal
- an earphone comprising the aforementioned means for realizing a karaoke function through a headphone.
- an earphone comprising a microphone, a processor for collecting a voice signal, and a memory for storing the instruction for storing an instruction
- the processor operates to perform the aforementioned method of implementing a karaoke function through a headset.
- the inventors of the present invention found that in the prior art, there is a problem that the karaoke software cannot realize the karaoke effect in real time.
- the karaoke effect can be realized only by the earphone in real time, thereby improving the user experience. Therefore, the technical task to be achieved by the present invention or the technical problem to be solved is not thought of or expected by those skilled in the art, so the present invention is a new technical solution.
- FIG. 1 is a flow chart of an embodiment of a method of implementing a karaoke function through a headset in accordance with the present invention
- FIG. 2 is a flow chart of another embodiment of a method for implementing a karaoke function through a headset in accordance with the present invention
- FIG. 3 is a block schematic diagram of an implementation structure of an apparatus for implementing a karaoke function through an earphone according to the present invention
- Figure 4 is a schematic illustration of an embodiment of an earphone in accordance with the present invention.
- the invention solves the problem that the existing K-song software can not realize the karaoke effect in real time, and provides a new technical solution for realizing the karaoke function through the earphone.
- FIG. 1 is a flow chart of an embodiment of a method of implementing a karaoke function through a headset of the present invention.
- the method of the invention comprises the following steps:
- Step S101 receiving a voice signal collected by a microphone.
- the voice signal collected by the microphone is received for subsequent processing steps.
- Step S102 performing analog-to-digital conversion processing on the voice signal to obtain a digital voice signal.
- Converting the analog voice signal into a digital voice signal can realize the karaoke function through the DSP processor in the earphone, while the analog signal cannot realize the karaoke function through the processor.
- Step S103 filtering out the human voice signal in the digital music signal received by the earphone to obtain a background music signal.
- the characteristics of the original vocal are roughly divided into two types: the sound image position of the human voice is in the center of the entire sound field (the left and right channels are balancedly distributed); the sound frequency of the human voice is concentrated in the intermediate frequency and the high frequency portion. Therefore, the vocal signal in the digital music signal received by the earphone is filtered out, specifically, the background music signal can be obtained by filtering out the equal sound of the left and right channels in the digital music signal and the frequency is concentrated in the intermediate frequency and the high frequency part. .
- the music file played by the electronic device includes two tracks: the original vocal track and the vocal track, and the background music is obtained by switching the audio track by controlling the electronic device connected to the earphone to switch between the original vocal and the accompaniment. signal.
- Step S104 superimposing the background music signal and the digital voice signal to obtain a digital superposition signal.
- step S105 the digital superimposed signal is subjected to digital-to-analog conversion processing, and then sent to the earphone.
- the digital superimposed signal needs to be digital-to-analog converted to obtain an analog superimposed signal and sent to the earpiece.
- the headset realizes the karaoke effect in real time.
- the opening of the karaoke function can be controlled by a button on the earphone, or the earphone can be automatically turned on.
- the method of the present invention further includes: detecting whether the digital voice signal includes a portion whose amplitude exceeds the first set value, and if so, performing step S103; if not, digitizing the digital music signal The analog conversion process obtains an analog music signal and sends the analog music signal to the earpiece of the earphone.
- the amplitude of the singing signal is the largest, so by detecting whether the digital voice signal contains a portion whose amplitude exceeds the first set value, It is judged whether the wearer is singing, and if so, the karaoke function is turned on, and if not, the music signal received by the earphone is played. In this way, the function of automatically turning on karaoke is realized.
- the present invention can also implement a karaoke function through an earphone having two microphones
- FIG. 2 is another embodiment of a method for implementing a karaoke function through a headphone according to the present invention. flow chart.
- the method includes the following steps:
- Step S201 receiving a first voice signal and a second voice signal.
- the earphones can also realize the functions of active noise reduction and call noise reduction through the two microphones, wherein the active noise reduction is to process the inverse of the microphone signal collected by the microphone.
- the active noise reduction is to process the inverse of the microphone signal collected by the microphone.
- the call noise reduction may specifically eliminate the noise signal in the first voice signal, and send the first voice signal that cancels the noise signal to the earphone Connected electronic devices.
- two microphones collect the voice of the person and the surrounding noise.
- the mouth is located in the middle of the ears in the vertical direction.
- the distance from the first microphone to the mouth is the same.
- the distance between the two microphones and the mouth is equal, so that the time when the first microphone and the second microphone receive the same sound signal from the human mouth is the same; in general, the distance between the first microphone and the noise source is the same.
- the distance between the two microphones to the noise source is not equal, thus causing a time difference between the time when the two microphones receive the same noise signal from the noise source.
- the distance from the first microphone to the mouth is not equal to the distance from the second microphone to the mouth.
- the mouth is offset from the vertical direction of the center point of the ears by a certain angle, and the mouth is in the angular range.
- the time difference between the two microphones receiving the sound signal from the mouth should not exceed the set value ⁇ T.
- the microphone signal received from the first microphone is extracted from the microphone signal received by the second microphone, but the time difference is not greater than the set value ⁇ T component, as the sound signal emitted by the mouth;
- the microphone signal received by the microphone extracts a component of the same signal as the microphone signal received by the second microphone but has a received time difference greater than the set value ⁇ T, and as the noise signal, inverts the noise signal to obtain the noise.
- the noise reduction signal with the same signal amplitude and opposite phase is superimposed and processed by the noise reduction signal and transmitted to the electronic device connected to the earphone. In this way, the headset can achieve call noise reduction through two microphones.
- Earphones with two microphones can also be implemented while enabling karaoke Active noise reduction and call noise reduction.
- Step S202 performing analog-to-digital conversion processing on the first speech signal and the second speech signal, respectively, to obtain a first digital speech signal and a second digital speech signal.
- the first speech signal is subjected to analog-to-digital conversion processing to obtain a first digital speech signal
- the second speech signal is subjected to analog-to-digital conversion processing to obtain a second digital speech signal.
- Step S203 performing beamforming processing on the first digital voice signal and the second digital voice signal.
- the beamforming process is performed on the first digital voice signal and the second digital voice signal, so that the voice sung by the wearer in the voice signal collected by the microphone can be made clearer, wherein the beamforming process can mainly suppress the first voice signal and the second voice signal.
- the sound signal in the left and right direction during the wearing of the earphone outputs a sound signal in the front and rear direction, that is, the sound of the wearer singing.
- beamforming refers to a method of forming spatial directivity by processing (eg, weighting, delay, summation, etc.) of each array element of a multi-array array arranged with a certain geometric shape (straight line, cylinder, arc, etc.).
- a wide-side microphone array refers to a series of microphones arranged in a direction perpendicular to the direction of the sound waves to be picked. d is the spacing of the two microphone elements in the array. The sound from the wide side of the array is usually the sound to pick up.
- the wide-edge array can be implemented by basic processing, and the microphones in the array are simply added. The disadvantage of this type of array is that it only attenuates the sound from the sides of the array.
- the rear response is always consistent with the front response because the array is axisymmetric and cannot distinguish between sound pressure waves arriving from the front and from the rear to the microphone.
- the minimum response occurs at 90° and 270°.
- the signal attenuation at these points is highly dependent on the frequency.
- the half wavelength of the incident frequency approaches the pitch of the microphone, the response approaches cancellation completely.
- the frequency is about 2.3 kHz (343 m/s ⁇ (0.075 m x 2) ⁇ 2.3 kHz)
- the response is completely cancelled.
- the frequencies will alias and the polar coordinate response will begin to show zero at other angles. At this point, the side attenuation begins to decrease again.
- the omnidirectional microphone since the omnidirectional microphone is mostly used in the earphone, it is capable of equally responding to sounds from all directions.
- the two microphones can be configured in an array to form a directional response or beam pattern. Beamformed microphone arrays are designed to be more sensitive to sound from one or more specific directions. Thus, by the beamforming process, the noise signals in the first speech signal and the second speech signal can be filtered out to obtain a clearer singing voice signal.
- Step S204 the portion of the signal obtained by the beamforming process whose amplitude exceeds the second set value is removed, and a digital voice signal is obtained.
- the portion of the signal obtained by removing the beamforming process that exceeds the second set value is a howling, and since the microphone is mounted on the earpiece of the earphone, a harsh whistling may occur, in order to prevent the human ear from hearing howling, resulting in a comparison.
- a poor user experience can eliminate the portion of the signal resulting from the beamforming process that exceeds the set value.
- the conditions for howling are required to satisfy three points: the microphone and the speaker are used at the same time; the sound reproduced by the sound system can be transmitted to the microphone through the space; the sound energy emitted by the speaker is large enough, and the pickup sensitivity of the microphone is sufficiently high.
- the microphone should be non-directional.
- the directional microphone especially the sharp directional microphone
- the adjustment distance has little effect on increasing the sound volume and preventing whistling.
- the sound reinforcement system is easy to whistle is not directly related to the sensitivity of the microphone.
- Only the high-sensitivity microphones are sharply directional, which is prone to whistling. Shortening the distance between the sounding device and the listener can actually increase the loudness of the sound. The total gain of the system can be appropriately reduced. If it is supplemented by a near-field speaker with wide directivity, the microphone can be screamed away from the far point.
- the farther the microphone is from the speaker the closer the speaker is to the listener.
- the microphone should be placed on the back of the speaker's radiation direction. If the microphone is likely to be held around, the speaker should be placed where the microphone is not close enough.
- the frequency equalizer can be used to compensate the sound reinforcement curve, and the frequency response of the system is adjusted to an approximate straight line, so that the gains of the respective frequency bands are basically the same, and the sound transmission gain of the system is improved.
- the audio feedback automatic suppression device is commonly used.
- This device can automatically track the feedback point frequency, automatically adjust the Q-value bandwidth, automatically eliminate the sound feedback and maximize the protection. Sound quality.
- the principle is to suppress howling by notching.
- Sabine's FBX series feedback suppressor which is a 9-segment narrow-band automatic compression device controlled by a microcomputer, can better distinguish between feedback self-excited signals and music signals, and can react quickly when the system is self-excited.
- set a very narrow digital filter at the feedback frequency the notch depth is also automatically set, the filtering bandwidth is only 1/3 octave, such a narrow notch band, almost no loudness and The tone has an effect.
- Inverting cancellation prevents self-excitation in high frequency amplification circuits.
- two microphones of the same specification can be used to respectively pick up the direct sound and the reflected sound, and the inverted sound circuit cancels the phase of the reflected sound signal before entering the power amplifier, which can effectively prevent the howling self-excitation.
- the self-excited howling of the sound reinforcement system, the feedback loop is positive feedback, if the microphone signal is phase-modulated, it will destroy the self-excited phase condition, thus preventing the self-excited howling of the system. It is shown that the stability is best when the phase deviation value is 140°; and the higher the frequency of modulation, the better the stability of the system. In order to prevent the processed sound quality from being too distorted, the maximum allowable value of the phase modulation frequency is 4 Hz.
- the method for removing the howling may also be to eliminate the portion of the signal obtained by the beamforming process whose amplitude exceeds the second set value, so that the generated howling is filtered out and played out without the earphone. This makes the human ear not hear the howling and enhance the user experience.
- Step S205 adding a reverberation signal corresponding to the digital voice signal to the digital voice signal.
- the reverberation signal corresponding to the digital voice signal may be added to the digital voice signal.
- the method for obtaining the reverberation signal of the digital speech signal can be:
- the digital speech signal is subjected to a first delay processing to generate a first signal; the digital speech signal is subjected to a second delay processing and a frequency domain compression filtering process to generate a second signal; and the digital speech signal is subjected to a third delay processing and frequency
- the domain compression filtering process generates a third signal; the first sound signal, the second signal, and the third signal are combined into a fourth signal as a reverberation signal.
- the number of delay processing and the number of reflections are the same, and may be more than three times, and may be more.
- Adding a reverberation signal corresponding to the digital voice signal to the digital voice signal enables the earphone wearer to experience the stereo sound effect and enhance the user experience.
- Step S206 detecting whether the digital speech signal includes a portion whose amplitude exceeds the first set value, and if yes, executing step S207; if not, executing step S211.
- Step S207 filtering out the human voice signal in the digital music signal received by the earphone to obtain a background music signal.
- Step S208 adding a reverberation signal corresponding to the background music signal to the background music signal.
- the reverberation corresponding to the background music signal may be added to the background music signal.
- the specific method is similar to the method of adding the reverberation corresponding to the digital voice signal in the digital voice signal, and details are not described herein again.
- step S209 the background music signal and the digital voice signal are superimposed to obtain a digital superimposed signal.
- step S210 the digital superimposed signal is subjected to digital-to-analog conversion processing, and then sent to the earphone receiver.
- step S211 the digital music signal is subjected to digital-to-analog conversion processing, and then sent to the earphone.
- the present invention also provides a K song processing device for an earphone, and FIG. 3 is a block schematic diagram of an implementation structure of the device.
- the apparatus 300 includes a receiving module 301, an analog to digital conversion module 302, a filtered vocal signal module 303, a superimposing module 304, and a first digital to analog conversion module 305.
- the receiving module 301 is configured to receive a voice signal collected by a microphone of the earphone.
- the analog-to-digital conversion module 302 is configured to perform analog-to-digital conversion processing on the voice signal to obtain a digital voice signal.
- the ergonomic signal module 303 is configured to filter out a vocal signal in the digital music signal received by the earphone to obtain a background music signal.
- the superimposing module 304 is configured to superimpose the background music signal and the digital speech signal to obtain a digital superimposed signal.
- the first digital-to-analog conversion module 305 is configured to perform digital-to-analog conversion processing on the digital superimposed signal to obtain an analog superimposed signal, and send the analog superimposed signal to an earpiece of the earphone.
- the apparatus 300 further includes a detecting module and a second digital-to-analog conversion module, configured to detect whether the digital voice signal includes detecting whether the digital voice signal includes a portion whose amplitude exceeds the first set value, and if yes, Controlling the ergonomic signal module to filter out the vocal signal in the digital music signal received by the earphone to obtain a background music signal; if not, controlling the second digital-to-analog conversion module to perform digital-to-analog conversion processing on the digital music signal to obtain analog music Signal and send an analog music signal to the handset of the headset.
- a detecting module and a second digital-to-analog conversion module configured to detect whether the digital voice signal includes detecting whether the digital voice signal includes a portion whose amplitude exceeds the first set value, and if yes, Controlling the ergonomic signal module to filter out the vocal signal in the digital music signal received by the earphone to obtain a background music signal; if not, controlling the second digital-to-analog conversion module to perform digital-to-ana
- the receiving module includes a first receiving unit and a second receiving unit, where the first receiving unit is configured to receive a voice signal collected by a microphone mounted on a earphone as a first voice signal; the second receiving unit is configured to: Receiving a voice signal collected by a microphone mounted on another earphone as a second voice signal.
- the first digital-to-analog conversion module includes a digital-to-analog conversion unit and a beamforming unit, and the digital-to-analog conversion unit is configured to respectively perform analog-to-digital conversion processing on the first voice signal and the second voice signal to obtain a corresponding first digital voice signal and a first And a second digital voice signal; the beam forming unit is configured to perform beamforming processing on the first digital voice signal and the second digital voice signal to obtain a digital voice signal.
- the beamforming unit includes a beamforming subunit and a cancellation subunit for performing beamforming processing on the first digital speech signal and the second digital speech signal; the cancellation subunit is configured to eliminate beamforming processing A portion of the obtained signal whose amplitude exceeds the second set value yields a digital speech signal.
- the apparatus 300 further includes a first reverberation module and a second reverberation module, where the first reverberation module is configured to add a reverberation signal corresponding to the background music signal to the background music signal;
- the second reverberation module is configured to add a reverberation signal corresponding to the digital speech signal to the digital speech signal.
- the present invention also provides an earphone, which in one aspect comprises the aforementioned apparatus 300 for implementing a karaoke function through a headset.
- the earphone includes a memory and a processor, wherein the memory is for storing instructions that control the processor to perform the aforementioned method of implementing a karaoke function through the earphone.
- the earphone includes an earphone earpiece 1, a headphone cable 2, a control box 3, and a headphone plug 4, as shown in FIG. 4, wherein the earphone is a digital earphone, and the earphone plug 4 is a USB plug.
- the earphone further includes a microphone for collecting a voice signal, and if the earphone has a microphone, the microphone can be mounted on the control box 3 or on the earphone 1; if the earphone has two microphones, each microphone can be installed differently Headphones on the handset 1.
- the processor can be, for example, a DSP digital processor.
- the memory includes, for example, a ROM (Read Only Memory), a RAM (Random Access Memory), a nonvolatile memory such as a hard disk, and the like.
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Abstract
Description
本发明涉及耳机技术领域,更具体地,本发明涉及一种通过耳机实现卡拉OK功能的方法、装置及耳机。The present invention relates to the field of earphone technology, and more particularly, to a method, device and earphone for implementing a karaoke function through an earphone.
随着社会的发展进步,唱卡拉OK已经不再需要走进歌舞厅、KTV包房了,K歌软件实现了随时随地唱卡拉OK的效果。但是,使唱卡拉OK软件是将演唱者的声音进行录音,在演唱结束后才能够听到演唱者的演唱效果,不能实时实现卡拉OK效果,用户体验较差。With the development and progress of the society, singing karaoke no longer needs to enter the dance hall, KTV room, and K-software realizes the effect of singing karaoke anytime and anywhere. However, the singer karaoke software is to record the singer's voice, and the singer's singing effect can be heard after the singing is finished. The karaoke effect cannot be realized in real time, and the user experience is poor.
因此,提出一种通过耳机实现实时卡拉OK功能的处理方法是十分有价值的。Therefore, it is very valuable to propose a processing method for real-time karaoke function through headphones.
发明内容Summary of the invention
本发明的一个目的是提供一种通过耳机实现实时卡拉OK功能的新技术方案。It is an object of the present invention to provide a new technical solution for realizing a real-time karaoke function through headphones.
根据本发明的第一方面,提供了一种通过耳机实现卡拉OK功能的方法,包括:According to a first aspect of the present invention, a method for implementing a karaoke function through a headset is provided, including:
接收耳机的麦克风采集的语音信号;Receiving a voice signal collected by a microphone of the earphone;
对所述语音信号进行模数转换处理,得到数字语音信号;Performing analog-to-digital conversion processing on the voice signal to obtain a digital voice signal;
滤除所述耳机接收的数字音乐信号中的人声信号,得到背景音乐信号;Filtering a vocal signal in the digital music signal received by the earphone to obtain a background music signal;
将所述背景音乐信号和所述数字语音信号进行叠加,得到数字叠加信号;Superimposing the background music signal and the digital voice signal to obtain a digital superposition signal;
将所述数字叠加信号进行数模转换处理,得到模拟叠加信号,并将所述模拟叠加信号发送至所述耳机的听筒。And performing digital-to-analog conversion processing on the digital superimposed signal to obtain an analog superimposed signal, and transmitting the analog superimposed signal to an earpiece of the earphone.
可选的是,所述方法还包括: Optionally, the method further includes:
检测所述数字语音信号中是否包含振幅超过第一设定值的部分,如是,再滤除所述耳机接收的数字音乐信号中的人声信号,得到背景音乐信号;如否,则将所述数字音乐信号进行数模转换处理,得到模拟音乐信号,并将所述模拟音乐信号发送至所述耳机的听筒。Detecting whether the digital voice signal includes a portion whose amplitude exceeds the first set value, and if so, filtering out the vocal signal in the digital music signal received by the earphone to obtain a background music signal; if not, the The digital music signal is subjected to digital-to-analog conversion processing to obtain an analog music signal, and the analog music signal is transmitted to the earpiece of the earphone.
可选的是,所述方法还包括:Optionally, the method further includes:
在所述背景音乐信号中加入所述背景音乐信号对应的混响信号;Adding a reverberation signal corresponding to the background music signal to the background music signal;
在所述数字语音信号中加入所述数字语音信号对应的混响信号。Adding a reverberation signal corresponding to the digital voice signal to the digital voice signal.
可选的是,所述接收耳机的麦克风采集的语音信号包括:Optionally, the voice signal collected by the microphone of the receiving headset includes:
接收安装在一耳机听筒上的麦克风采集到的语音信号,作为第一语音信号;Receiving a voice signal collected by a microphone mounted on a earphone as a first voice signal;
接收安装在另一耳机听筒上的麦克风采集到的语音信号,作为第二语音信号;Receiving a voice signal collected by a microphone installed on another earphone as a second voice signal;
所述对所述语音信号进行模数转换处理,得到数字语音信号具体为:Performing analog-to-digital conversion processing on the voice signal to obtain a digital voice signal, specifically:
对所述第一语音信号和所述第二语音信号分别进行模数转换处理,得到对应的第一数字语音信号和第二数字语音信号;Performing analog-to-digital conversion processing on the first voice signal and the second voice signal, respectively, to obtain a corresponding first digital voice signal and a second digital voice signal;
对所述第一数字语音信号和所述第二数字语音信号进行波束形成处理,得到所述数字语音信号。And performing beamforming processing on the first digital voice signal and the second digital voice signal to obtain the digital voice signal.
可选的是,所述对所述第一数字语音信号和所述第二数字语音信号进行波束形成处理,得到所述数字语音信号包括:Optionally, performing beamforming processing on the first digital voice signal and the second digital voice signal, and obtaining the digital voice signal includes:
对所述第一数字语音信号和所述第二数字语音信号进行波束形成处理;Performing beamforming processing on the first digital voice signal and the second digital voice signal;
消除波束形成处理得到的信号中振幅超过第二设定值的部分,得到所述数字语音信号。The digital speech signal is obtained by eliminating a portion of the signal obtained by the beamforming process that has an amplitude exceeding a second set value.
根据本发明的第二方面,提供了一种通过耳机实现卡拉OK功能的装置,包括:According to a second aspect of the present invention, an apparatus for implementing a karaoke function through a headset is provided, including:
接收模块,用于接收耳机的麦克风采集的语音信号;a receiving module, configured to receive a voice signal collected by a microphone of the earphone;
模数转换模块,用于对所述语音信号进行模数转换处理,得到数字语音信号;An analog-to-digital conversion module, configured to perform analog-to-digital conversion processing on the voice signal to obtain a digital voice signal;
滤除人声信号模块,用于滤除所述耳机接收的数字音乐信号中的人声 信号,得到背景音乐信号;Filtering a vocal signal module for filtering vocals in a digital music signal received by the earphone Signal, get the background music signal;
叠加模块,用于将所述背景音乐信号和所述数字语音信号进行叠加,得到数字叠加信号;以及,a superimposing module, configured to superimpose the background music signal and the digital speech signal to obtain a digital superimposed signal;
第一数模转换模块,用于将所述数字叠加信号进行数模转换处理,得到模拟叠加信号,并将所述模拟叠加信号发送至所述耳机的听筒。The first digital-to-analog conversion module is configured to perform digital-to-analog conversion processing on the digital superimposed signal to obtain an analog superimposed signal, and send the analog superimposed signal to an earpiece of the earphone.
可选的是,所述装置还包括检测模块和第二数模转换模块,所述检测模块用于检测所述数字语音信号中是否包含检测所述数字语音信号中是否包含振幅超过第一设定值的部分,如是,则控制所述滤除人声信号模块滤除所述耳机接收的数字音乐信号中的人声信号,得到背景音乐信号;如否,则控制所述第二数模转换模块将所述数字音乐信号进行数模转换处理,得到模拟音乐信号,并将所述模拟音乐信号发送至所述耳机的听筒。Optionally, the device further includes a detection module and a second digital-to-analog conversion module, the detection module is configured to detect whether the digital voice signal includes detecting whether the digital voice signal includes an amplitude exceeding a first setting. a portion of the value, if yes, controlling the filtered vocal signal module to filter out a vocal signal in the digital music signal received by the earphone to obtain a background music signal; if not, controlling the second digital to analog conversion module And performing digital-to-analog conversion processing on the digital music signal to obtain an analog music signal, and transmitting the analog music signal to an earpiece of the earphone.
可选的是,所述装置还包括:Optionally, the device further includes:
第一加混响模块,用于在所述背景音乐信号中加入所述背景音乐信号对应的混响信号;a first reverberation module, configured to add a reverberation signal corresponding to the background music signal to the background music signal;
第二加混响模块,用于在所述数字语音信号中加入所述数字语音信号对应的混响信号。And a second plus reverberation module, configured to add a reverberation signal corresponding to the digital voice signal to the digital voice signal.
可选的是,所述接收模块包括:Optionally, the receiving module includes:
第一接收单元,用于接收安装在一耳机听筒上的麦克风采集到的语音信号,作为第一语音信号;a first receiving unit, configured to receive a voice signal collected by a microphone mounted on a earphone as a first voice signal;
第二接收单元,用于接收安装在另一耳机听筒上的麦克风采集到的语音信号,作为第二语音信号;a second receiving unit, configured to receive a voice signal collected by a microphone mounted on another earphone as a second voice signal;
第一数模转换模块包括:The first digital to analog conversion module includes:
数模转换单元,用于对所述第一语音信号和所述第二语音信号分别进行模数转换处理,得到对应的第一数字语音信号和第二数字语音信号;a digital-to-analog conversion unit, configured to perform analog-to-digital conversion processing on the first voice signal and the second voice signal, respectively, to obtain a corresponding first digital voice signal and a second digital voice signal;
波束形成单元,用于对所述第一数字语音信号和所述第二数字语音信号进行波束形成处理,得到所述数字语音信号。And a beam forming unit configured to perform beamforming processing on the first digital voice signal and the second digital voice signal to obtain the digital voice signal.
可选的是,所述波束形成单元包括:Optionally, the beam forming unit includes:
波束形成子单元,用于对所述第一数字语音信号和所述第二数字语音信号进行波束形成处理; a beamforming subunit, configured to perform beamforming processing on the first digital voice signal and the second digital voice signal;
消除子单元,用于消除波束形成处理得到的信号中振幅超过第二设定值的部分,得到所述数字语音信号a eliminating subunit, configured to cancel a portion of the signal obtained by the beamforming process and having an amplitude exceeding a second set value, to obtain the digital voice signal
根据本发明的第三方面,提供了一种耳机,包括前述的通过耳机实现卡拉OK功能的装置。According to a third aspect of the present invention, there is provided an earphone comprising the aforementioned means for realizing a karaoke function through a headphone.
根据本发明的第四方面,提供了一种耳机,包括麦克风、处理器和存储器,所述麦克风用于采集语音信号,所述存储器用于所述存储器用于存储指令,所述指令用于控制所述处理器进行操作以执行前述的通过耳机实现卡拉OK功能的方法。According to a fourth aspect of the present invention, there is provided an earphone comprising a microphone, a processor for collecting a voice signal, and a memory for storing the instruction for storing an instruction The processor operates to perform the aforementioned method of implementing a karaoke function through a headset.
本发明的发明人发现,在现有技术中,存在K歌软件不能实时实现卡拉OK效果的问题。而在本发明中,通过耳机将麦克风采集的演唱者的唱歌语音信号和消除音乐信号中人声信号的背景音乐信号进行叠加,即可仅通过耳机实时实现卡拉OK效果,提升用户体验。因此,本发明所要实现的技术任务或者所要解决的技术问题是本领域技术人员从未想到的或者没有预期到的,故本发明是一种新的技术方案。The inventors of the present invention found that in the prior art, there is a problem that the karaoke software cannot realize the karaoke effect in real time. In the present invention, by superimposing the singing voice signal of the singer collected by the microphone and the background music signal of the vocal signal in the music signal through the earphone, the karaoke effect can be realized only by the earphone in real time, thereby improving the user experience. Therefore, the technical task to be achieved by the present invention or the technical problem to be solved is not thought of or expected by those skilled in the art, so the present invention is a new technical solution.
通过以下参照附图对本发明的示例性实施例的详细描述,本发明的其它特征及其优点将会变得清楚。Other features and advantages of the present invention will become apparent from the Detailed Description of the <RTIgt;
被结合在说明书中并构成说明书的一部分的附图示出了本发明的实施例,并且连同其说明一起用于解释本发明的原理。The accompanying drawings, which are incorporated in FIG
图1是根据本发明一种通过耳机实现卡拉OK功能的方法的一种实施方式的流程图;1 is a flow chart of an embodiment of a method of implementing a karaoke function through a headset in accordance with the present invention;
图2是根据本发明一种通过耳机实现卡拉OK功能的方法的另一种实施方式的流程图;2 is a flow chart of another embodiment of a method for implementing a karaoke function through a headset in accordance with the present invention;
图3是根据本发明一种通过耳机实现卡拉OK功能的装置的一种实施结构的方框原理图;3 is a block schematic diagram of an implementation structure of an apparatus for implementing a karaoke function through an earphone according to the present invention;
图4是根据本发明一种耳机的一种实施结构的示意图。Figure 4 is a schematic illustration of an embodiment of an earphone in accordance with the present invention.
附图标记说明:Description of the reference signs:
1-耳机听筒; 2-耳机线; 1-headphone handset; 2-headphone cord;
3-控制盒; 4-耳机插头。3-control box; 4-headphone plug.
现在将参照附图来详细描述本发明的各种示例性实施例。应注意到:除非另外具体说明,否则在这些实施例中阐述的部件和步骤的相对布置、数字表达式和数值不限制本发明的范围。Various exemplary embodiments of the present invention will now be described in detail with reference to the drawings. It should be noted that the relative arrangement of the components and steps, numerical expressions and numerical values set forth in the embodiments are not intended to limit the scope of the invention unless otherwise specified.
以下对至少一个示例性实施例的描述实际上仅仅是说明性的,决不作为对本发明及其应用或使用的任何限制。The following description of the at least one exemplary embodiment is merely illustrative and is in no way
对于相关领域普通技术人员已知的技术、方法和设备可能不作详细讨论,但在适当情况下,所述技术、方法和设备应当被视为说明书的一部分。Techniques, methods and apparatus known to those of ordinary skill in the relevant art may not be discussed in detail, but the techniques, methods and apparatus should be considered as part of the specification, where appropriate.
在这里示出和讨论的所有例子中,任何具体值应被解释为仅仅是示例性的,而不是作为限制。因此,示例性实施例的其它例子可以具有不同的值。In all of the examples shown and discussed herein, any specific values are to be construed as illustrative only and not as a limitation. Thus, other examples of the exemplary embodiments may have different values.
应注意到:相似的标号和字母在下面的附图中表示类似项,因此,一旦某一项在一个附图中被定义,则在随后的附图中不需要对其进行进一步讨论。It should be noted that similar reference numerals and letters indicate similar items in the following figures, and therefore, once an item is defined in one figure, it is not required to be further discussed in the subsequent figures.
本发明解决了现有K歌软件不能实时实现卡拉OK效果的问题,提供了一种通过耳机实现卡拉OK功能的新的技术方案。The invention solves the problem that the existing K-song software can not realize the karaoke effect in real time, and provides a new technical solution for realizing the karaoke function through the earphone.
图1是本发明的一种通过耳机实现卡拉OK功能的方法的一种实施方式的流程图。1 is a flow chart of an embodiment of a method of implementing a karaoke function through a headset of the present invention.
根据图1所示,本发明方法包括如下步骤:According to Figure 1, the method of the invention comprises the following steps:
步骤S101,接收麦克风采集的语音信号。Step S101, receiving a voice signal collected by a microphone.
具体的,如果耳机只具有一个安装在控制盒或者一耳机听筒上的麦克风,则接收该麦克风采集的语音信号,以进行后续处理步骤。Specifically, if the earphone has only one microphone installed on the control box or a earphone, the voice signal collected by the microphone is received for subsequent processing steps.
步骤S102,对语音信号进行模数转换处理,得到数字语音信号。Step S102, performing analog-to-digital conversion processing on the voice signal to obtain a digital voice signal.
将模拟的语音信号转换为数字语音信号,可以通过耳机内的DSP处理器实现卡拉OK功能,而模拟信号则无法通过处理器实现卡拉OK功能。Converting the analog voice signal into a digital voice signal can realize the karaoke function through the DSP processor in the earphone, while the analog signal cannot realize the karaoke function through the processor.
步骤S103,滤除耳机接收的数字音乐信号中的人声信号,得到背景音乐信号。 Step S103, filtering out the human voice signal in the digital music signal received by the earphone to obtain a background music signal.
在数字音乐信号中,原唱的特征大致分为两种:人声的声像位置在整个声场的中央(左右声道平衡分布);人声的声音频率集中在中频和高频部分。因此,滤除耳机接收的数字音乐信号中的人声信号,具体可以为滤除数字音乐信号中左右声道的对等声音且频率集中在中频和高频部分的声音,就能够得到背景音乐信号。In the digital music signal, the characteristics of the original vocal are roughly divided into two types: the sound image position of the human voice is in the center of the entire sound field (the left and right channels are balancedly distributed); the sound frequency of the human voice is concentrated in the intermediate frequency and the high frequency portion. Therefore, the vocal signal in the digital music signal received by the earphone is filtered out, specifically, the background music signal can be obtained by filtering out the equal sound of the left and right channels in the digital music signal and the frequency is concentrated in the intermediate frequency and the high frequency part. .
还可以是电子设备播放的音乐文件中包含双音轨:原唱音轨和伴唱音轨,通过控制与耳机连接的电子设备切换音轨来实现原唱和伴唱之间的切换,来得到背景音乐信号。It is also possible that the music file played by the electronic device includes two tracks: the original vocal track and the vocal track, and the background music is obtained by switching the audio track by controlling the electronic device connected to the earphone to switch between the original vocal and the accompaniment. signal.
步骤S104,将背景音乐信号和数字语音信号进行叠加,得到数字叠加信号。Step S104, superimposing the background music signal and the digital voice signal to obtain a digital superposition signal.
步骤S105,将数字叠加信号进行数模转换处理后,发送至耳机听筒。In step S105, the digital superimposed signal is subjected to digital-to-analog conversion processing, and then sent to the earphone.
由于耳机听筒只能通过模拟信号才能驱动,因此,需将数字叠加信号进行数模转换处理,得到模拟叠加信号后,发送至耳机听筒。Since the earphone can only be driven by an analog signal, the digital superimposed signal needs to be digital-to-analog converted to obtain an analog superimposed signal and sent to the earpiece.
这样,通过将背景音乐信号和数字语音信号进行叠加,再进行数模转换处理,发送至耳机听筒,使得佩戴者既能够听到背景音乐,又能够同时听到佩戴者唱歌的声音,就能够使得耳机实时实现卡拉OK效果。In this way, by superimposing the background music signal and the digital voice signal, and then performing digital-to-analog conversion processing, it is sent to the earphone, so that the wearer can both hear the background music and simultaneously hear the sound of the wearer singing. The headset realizes the karaoke effect in real time.
其中,卡拉OK功能的开启可以是通过耳机上的按键等控制,也可以是耳机自动开启。在本发明的一个具体实施例中,本发明方法还包括:检测数字语音信号中是否包含振幅超过第一设定值的部分,如是,再执行步骤S103;如否,则将数字音乐信号进行数模转换处理,得到模拟音乐信号,并将模拟音乐信号发送至耳机的听筒。Among them, the opening of the karaoke function can be controlled by a button on the earphone, or the earphone can be automatically turned on. In a specific embodiment of the present invention, the method of the present invention further includes: detecting whether the digital voice signal includes a portion whose amplitude exceeds the first set value, and if so, performing step S103; if not, digitizing the digital music signal The analog conversion process obtains an analog music signal and sends the analog music signal to the earpiece of the earphone.
由于在耳机佩戴过程中,佩戴者的嘴巴距离麦克风的距离最近,如果佩戴者唱歌,则该唱歌信号的振幅最大,因此通过检测数字语音信号中是否包含振幅超过第一设定值的部分,可以判断佩戴者是否在唱歌,如果是,则开启卡拉OK功能,如否,则播放耳机接收的音乐信号。这样,就实现了自动开启卡拉OK的功能。Since the wearer's mouth is closest to the microphone during the wearing of the earphone, if the wearer sings, the amplitude of the singing signal is the largest, so by detecting whether the digital voice signal contains a portion whose amplitude exceeds the first set value, It is judged whether the wearer is singing, and if so, the karaoke function is turned on, and if not, the music signal received by the earphone is played. In this way, the function of automatically turning on karaoke is realized.
在本发明的另一个具体实施例中,本发明还可以通过具有两个麦克风的耳机实现卡拉OK功能,图2是本发明的一种通过耳机实现卡拉OK功能的方法的另一种实施方式的流程图。 In another embodiment of the present invention, the present invention can also implement a karaoke function through an earphone having two microphones, and FIG. 2 is another embodiment of a method for implementing a karaoke function through a headphone according to the present invention. flow chart.
如图2所示,该方法包括以下步骤:As shown in FIG. 2, the method includes the following steps:
步骤S201,接收第一语音信号和第二语音信号。Step S201, receiving a first voice signal and a second voice signal.
具体为,接收安装在一耳机听筒上的麦克风采集到的语音信号,作为第一语音信号,接收安装在另一耳机听筒上的麦克风采集到的语音信号,作为第二语音信号,以便分别进行处理。Specifically, receiving a voice signal collected by a microphone mounted on a earphone, as a first voice signal, receiving a voice signal collected by a microphone mounted on another earphone as a second voice signal for separately processing .
同时,如果耳机的两个听筒上均安装有麦克风,该耳机还可以通过这两个麦克风实现主动降噪和通话降噪的功能,其中,主动降噪是处理得到麦克风采集的麦克风信号对应的反相麦克风信号,并通过耳机听筒播放该反相麦克风信号,从而实现降噪的效果;通话降噪具体可以为消除第一语音信号中噪声信号,并将消除噪声信号的第一语音信号发送至于耳机连接的电子设备。At the same time, if the microphones are installed on the two earpieces of the earphone, the earphones can also realize the functions of active noise reduction and call noise reduction through the two microphones, wherein the active noise reduction is to process the inverse of the microphone signal collected by the microphone. Phase microphone signal, and playing the inverted microphone signal through the earphone, thereby achieving the effect of noise reduction; the call noise reduction may specifically eliminate the noise signal in the first voice signal, and send the first voice signal that cancels the noise signal to the earphone Connected electronic devices.
具体的,通话时,两个麦克风采集人说话声和周围的噪声,人耳正常佩戴时,嘴巴在竖直方向上位于两耳中间,在理想状态下,第一个麦克风到嘴巴的距离与第二个麦克风到嘴巴的距离相等,这样,第一个麦克风和第二个麦克风接收到人嘴巴发出的同一声音信号的时间是相同的;一般情况下,第一个麦克风到噪声源的距离与第二个麦克风到噪声源的距离不相等,这样,就导致两个麦克风接收到噪声源发出的同一噪声信号的时间之间存在时间差。在实际情况下,第一个麦克风到嘴巴的距离与第二个麦克风到嘴巴的距离不相等,因此,可以认为嘴巴在距离两耳中心点的竖直方向上偏离一定角度,嘴巴在该角度范围内时,两个麦克风接收到嘴巴发出的声音信号的时间差不应该超过设定值ΔT。这样,从第一个麦克风接收到的麦克风信号中,提取与第二个麦克风接收到的麦克风信号中特征相同、但时间差不大于设定值ΔT成分,作为嘴巴发出的声音信号;从第一个麦克风接收到的麦克风信号中,提取与第二个麦克风接收到的麦克风信号中特征相同、但接收时间差大于设定值ΔT的信号的成分,作为噪声信号,反相该噪声信号,就得到与噪声信号振幅相同、相位相反的降噪信号,将该降噪信号与第一语音信号叠加处理后发送至于耳机连接的电子设备。这样,该耳机通过两个麦克风就能够实现通话降噪。Specifically, when talking, two microphones collect the voice of the person and the surrounding noise. When the ear is normally worn, the mouth is located in the middle of the ears in the vertical direction. In the ideal state, the distance from the first microphone to the mouth is the same. The distance between the two microphones and the mouth is equal, so that the time when the first microphone and the second microphone receive the same sound signal from the human mouth is the same; in general, the distance between the first microphone and the noise source is the same The distance between the two microphones to the noise source is not equal, thus causing a time difference between the time when the two microphones receive the same noise signal from the noise source. In the actual case, the distance from the first microphone to the mouth is not equal to the distance from the second microphone to the mouth. Therefore, it can be considered that the mouth is offset from the vertical direction of the center point of the ears by a certain angle, and the mouth is in the angular range. The time difference between the two microphones receiving the sound signal from the mouth should not exceed the set value ΔT. In this way, the microphone signal received from the first microphone is extracted from the microphone signal received by the second microphone, but the time difference is not greater than the set value ΔT component, as the sound signal emitted by the mouth; The microphone signal received by the microphone extracts a component of the same signal as the microphone signal received by the second microphone but has a received time difference greater than the set value ΔT, and as the noise signal, inverts the noise signal to obtain the noise. The noise reduction signal with the same signal amplitude and opposite phase is superimposed and processed by the noise reduction signal and transmitted to the electronic device connected to the earphone. In this way, the headset can achieve call noise reduction through two microphones.
具有两个麦克风的耳机在能够实现卡拉OK功能的同时,还能够实现 主动降噪和通话降噪的功能。Earphones with two microphones can also be implemented while enabling karaoke Active noise reduction and call noise reduction.
步骤S202,对第一语音信号和第二语音信号分别进行模数转换处理,得到第一数字语音信号和第二数字语音信号。Step S202, performing analog-to-digital conversion processing on the first speech signal and the second speech signal, respectively, to obtain a first digital speech signal and a second digital speech signal.
具体为,对第一语音信号进行模数转换处理,得到第一数字语音信号;对第二语音信号进行模数转换处理,得到第二数字语音信号。Specifically, the first speech signal is subjected to analog-to-digital conversion processing to obtain a first digital speech signal, and the second speech signal is subjected to analog-to-digital conversion processing to obtain a second digital speech signal.
步骤S203,将第一数字语音信号和第二数字语音信号进行波束形成处理。Step S203, performing beamforming processing on the first digital voice signal and the second digital voice signal.
对第一数字语音信号和第二数字语音信号进行波束形成处理,可以使得麦克风采集的语音信号中佩戴者唱歌的声音更加清晰,其中,波束形成处理主要能够抑制第一语音信号和第二语音信号中在耳机佩戴过程中左右方向的声音信号,输出一路前后方向的声音信号,即佩戴者唱歌的声音。The beamforming process is performed on the first digital voice signal and the second digital voice signal, so that the voice sung by the wearer in the voice signal collected by the microphone can be made clearer, wherein the beamforming process can mainly suppress the first voice signal and the second voice signal. The sound signal in the left and right direction during the wearing of the earphone outputs a sound signal in the front and rear direction, that is, the sound of the wearer singing.
其中,波束形成是指将一定几何形状(直线、圆柱、弧形等)排列的多元基阵各阵元输出经过处理(例如加权、时延、求和等)形成空间指向性的方法。宽边麦克风阵列是指一系列麦克风的排列方向与要拾取的声波方向垂直。d是阵列中两个麦克风元件的间距。来自阵列宽边的声音通常就是要拾取的声音。宽边阵列可以通过基本处理实现,阵列中的麦克风简单地相加。此类阵列的缺点是它只能衰减来自阵列侧边的声音。后方响应始终与前方响应一致,因为阵列具有轴对称性,无法区分从前方与从后方到达麦克风的声压波。在双麦克风宽边阵列中,响应的最小值出现在90°和270°。这些点的信号衰减在很大程度上取决于频率。当入射频率的半波长接近麦克风的间距时,响应接近完全抵消。对于两个间距75mm的麦克风组成的阵列,理论上,当频率约为2.3kHz(343m/s÷(0.075m×2)≈2.3kHz)时,响应完全抵消。高于理想衰减的频率时,频率将混叠,极坐标响应开始在其它角度显示零点。此时,侧边衰减再次开始降低。Wherein, beamforming refers to a method of forming spatial directivity by processing (eg, weighting, delay, summation, etc.) of each array element of a multi-array array arranged with a certain geometric shape (straight line, cylinder, arc, etc.). A wide-side microphone array refers to a series of microphones arranged in a direction perpendicular to the direction of the sound waves to be picked. d is the spacing of the two microphone elements in the array. The sound from the wide side of the array is usually the sound to pick up. The wide-edge array can be implemented by basic processing, and the microphones in the array are simply added. The disadvantage of this type of array is that it only attenuates the sound from the sides of the array. The rear response is always consistent with the front response because the array is axisymmetric and cannot distinguish between sound pressure waves arriving from the front and from the rear to the microphone. In a two-microphone wide-edge array, the minimum response occurs at 90° and 270°. The signal attenuation at these points is highly dependent on the frequency. When the half wavelength of the incident frequency approaches the pitch of the microphone, the response approaches cancellation completely. For an array of two microphones with a pitch of 75 mm, theoretically, when the frequency is about 2.3 kHz (343 m/s ÷ (0.075 m x 2) ≈ 2.3 kHz), the response is completely cancelled. Above the ideal attenuated frequency, the frequencies will alias and the polar coordinate response will begin to show zero at other angles. At this point, the side attenuation begins to decrease again.
具体的,由于耳机中多采用全指向麦克风,也就是能够均等地响应来自四面八方的声音。两个麦克风可以配置成阵列,形成定向响应或波束场型。经过设计,波束成形麦克风阵列可以对来自一个或多个特定方向的声音更敏感。这样,通过波束形成处理,可以将第一语音信号和第二语音信号中的噪声信号过滤掉,得到更加清晰的唱歌的声音信号。 Specifically, since the omnidirectional microphone is mostly used in the earphone, it is capable of equally responding to sounds from all directions. The two microphones can be configured in an array to form a directional response or beam pattern. Beamformed microphone arrays are designed to be more sensitive to sound from one or more specific directions. Thus, by the beamforming process, the noise signals in the first speech signal and the second speech signal can be filtered out to obtain a clearer singing voice signal.
步骤S204,去除波束形成处理得到的信号中振幅超过第二设定值的部分,得到数字语音信号。Step S204, the portion of the signal obtained by the beamforming process whose amplitude exceeds the second set value is removed, and a digital voice signal is obtained.
其中,去除波束形成处理得到信号中振幅超过第二设定值的部分即为啸叫,由于麦克风安装在耳机听筒上,可能会产生刺耳的啸叫,为了避免人耳听到啸叫、产生较差的用户体验,可以消除波束形成处理得到的信号中振幅超过设定值的部分。Wherein, the portion of the signal obtained by removing the beamforming process that exceeds the second set value is a howling, and since the microphone is mounted on the earpiece of the earphone, a harsh whistling may occur, in order to prevent the human ear from hearing howling, resulting in a comparison. A poor user experience can eliminate the portion of the signal resulting from the beamforming process that exceeds the set value.
啸叫产生的条件需满足三点:话筒与音箱同时使用;音响系统重放的声音能够通过空间传到话筒;音箱发出的声音能量足够大,话筒的拾音灵敏度足够高。The conditions for howling are required to satisfy three points: the microphone and the speaker are used at the same time; the sound reproduced by the sound system can be transmitted to the microphone through the space; the sound energy emitted by the speaker is large enough, and the pickup sensitivity of the microphone is sufficiently high.
消除反馈啸叫要从产生反馈啸叫的必要条件入手,只要能破坏其中一个条件,就可达到目的。消除反馈啸叫的方法包括:Eliminating feedback whistling starts with the necessary conditions for generating feedback whistling, as long as one of the conditions can be destroyed, the goal can be achieved. Ways to eliminate feedback whistling include:
a)调整距离法a) Adjusting distance method
既避免啸叫又能提升扩音音量最有效的方法之一就是将话筒尽量靠近声源拾音,同时话筒应使用无指向性的。具体的是,指向性话筒(尤其是锐指向性话筒)远距离声源的拾音衰减很小,调整距离对提升扩音音量和防止啸叫的作用不大。扩声系统是否容易啸叫,与话筒的灵敏度没有直接关系。只不过高灵敏度的话筒都是锐指向性的,容易产生啸叫罢了。缩短发声设备与听众的距离,实际上可以提升扩音的响度。可适当的减小系统的总增益。若同时辅以指向性宽的近场音箱,话筒稍微离远点就能避免啸叫。One of the most effective ways to avoid whistling and increase the volume of the amplified sound is to pick up the microphone as close as possible to the sound source, and the microphone should be non-directional. Specifically, the directional microphone (especially the sharp directional microphone) has a small pickup attenuation of the remote sound source, and the adjustment distance has little effect on increasing the sound volume and preventing whistling. Whether the sound reinforcement system is easy to whistle is not directly related to the sensitivity of the microphone. Only the high-sensitivity microphones are sharply directional, which is prone to whistling. Shortening the distance between the sounding device and the listener can actually increase the loudness of the sound. The total gain of the system can be appropriately reduced. If it is supplemented by a near-field speaker with wide directivity, the microphone can be screamed away from the far point.
对于扬声器的直接反馈声场来说,就是话筒距扬声器越远越好,扬声器距听众越近越好。话筒应放在扬声器辐射方向的背面,如果话筒有可能被拿着四处走动,扬声器应放在话筒无法靠得很近的地方。For the direct feedback sound field of the speaker, the farther the microphone is from the speaker, the closer the speaker is to the listener. The microphone should be placed on the back of the speaker's radiation direction. If the microphone is likely to be held around, the speaker should be placed where the microphone is not close enough.
b)频率均衡法(宽带陷波法)b) Frequency equalization method (wideband notch method)
由于话筒拾音和发声设备的频率曲线不是理想平坦的直线(特别是一些质量比较差的放音设备),以及厅堂声场的声学谐振作用,使频率响应起伏很大。可以用频率均衡器补偿扩声曲线,把系统的频率响应调成近似的直线,使各频段的增益基本一致,提高系统的传声增益。Since the frequency curve of the microphone pickup and sounding equipment is not an ideal flat straight line (especially some poor quality playback equipment), and the acoustic resonance of the auditorium sound field, the frequency response fluctuates greatly. The frequency equalizer can be used to compensate the sound reinforcement curve, and the frequency response of the system is adjusted to an approximate straight line, so that the gains of the respective frequency bands are basically the same, and the sound transmission gain of the system is improved.
应该使用21段以上的均衡器,在要求比较高的地方应该配置参量均 衡器,要求更高时,可采用反馈抑制器。实际上扩声系统在出现反馈自激时,其频率只是固定在某一点上的纯音,所以,只要用一个频带很窄的陷波器将此频率切除,即可抑制系统啸叫。Equiliators with 21 or more segments should be used, and parameters should be configured where the requirements are relatively high. When the instrument is more demanding, a feedback suppressor can be used. In fact, when the feedback system is self-excited, its frequency is only a pure tone fixed at a certain point. Therefore, if the frequency is cut off by a narrow-band trap, the system howling can be suppressed.
c)反馈抑制器法(窄带陷波法)c) feedback suppressor method (narrowband notch method)
在要求很高的场合,如一些现场演唱的地方,普遍使用声频反馈自动抑制装置,这种装置可以自动跟踪反馈点频率,自动调整Q值带宽,自动将声反馈消除而又最大限度地保护了音质。其原理就是通过陷波抑制啸叫的。例如Sabine的FBX系列反馈抑制器,它是一种由微电脑控制的9段窄带自动压限装置,可以较好地区别反馈自激信号与音乐信号,可在系统出现自激时,迅速作出反应,并在反馈频点上设定一个很窄的数字滤波器,其陷波深度也会自动设定,滤波带宽只有1/3倍频程,如此之窄的陷波频段,几乎不会对响度以及音色有影响。In high-demand occasions, such as some live singing places, the audio feedback automatic suppression device is commonly used. This device can automatically track the feedback point frequency, automatically adjust the Q-value bandwidth, automatically eliminate the sound feedback and maximize the protection. Sound quality. The principle is to suppress howling by notching. For example, Sabine's FBX series feedback suppressor, which is a 9-segment narrow-band automatic compression device controlled by a microcomputer, can better distinguish between feedback self-excited signals and music signals, and can react quickly when the system is self-excited. And set a very narrow digital filter at the feedback frequency, the notch depth is also automatically set, the filtering bandwidth is only 1/3 octave, such a narrow notch band, almost no loudness and The tone has an effect.
d)反相抵消法d) inverse cancellation method
反相抵消防止自激在高频放大电路比较常见。可以在音频放大电路中采用两个同规格的话筒分别拾取直达声和反射声,通过反相电路使反射声信号在进入功放前相位相互抵消,能有效的防止啸叫自激。Inverting cancellation prevents self-excitation in high frequency amplification circuits. In the audio amplifying circuit, two microphones of the same specification can be used to respectively pick up the direct sound and the reflected sound, and the inverted sound circuit cancels the phase of the reflected sound signal before entering the power amplifier, which can effectively prevent the howling self-excitation.
e)调相法e) Phase modulation
扩音系统的自激啸叫,其反馈回路是正反馈,如果把话筒信号调相处理,就会破坏自激的相位条件,从而防止系统的自激啸叫。有资料表明,当相位偏差值在140°时,稳定度最好;并且,调制的频率越高,系统的稳定性越好。为了使处理后的音质不发生太大的畸变,其调相频率的最大允许值是4Hz。The self-excited howling of the sound reinforcement system, the feedback loop is positive feedback, if the microphone signal is phase-modulated, it will destroy the self-excited phase condition, thus preventing the self-excited howling of the system. It is shown that the stability is best when the phase deviation value is 140°; and the higher the frequency of modulation, the better the stability of the system. In order to prevent the processed sound quality from being too distorted, the maximum allowable value of the phase modulation frequency is 4 Hz.
在本实施例中,去除啸叫的方法还可以为消除波束形成处理得到的信号中振幅超过第二设定值的部分,这样,就将产生的啸叫过滤掉,不经耳机听筒播放出来,使得人耳不会听到啸叫,提升用户体验。In this embodiment, the method for removing the howling may also be to eliminate the portion of the signal obtained by the beamforming process whose amplitude exceeds the second set value, so that the generated howling is filtered out and played out without the earphone. This makes the human ear not hear the howling and enhance the user experience.
步骤S205,在数字语音信号中加入该数字语音信号对应的混响信号。Step S205, adding a reverberation signal corresponding to the digital voice signal to the digital voice signal.
为了使得佩戴者唱歌的声音实现卡拉OK的音效,可以在数字语音信号中加入该数字语音信号对应的混响信号。In order to enable the sound of the singer of the wearer to realize the sound effect of the karaoke, the reverberation signal corresponding to the digital voice signal may be added to the digital voice signal.
其中,声波在室内传播时,要被墙壁、天花板、地板等障碍物反射, 每反射一次都要被障碍物吸收一些。这样,当声源停止发声后,声波在室内要经过多次反射和吸收,最后才消失,我们就感觉到声源停止发声后还有若干个声波混合持续一段时间,这种现象叫做混响。Among them, when sound waves are transmitted indoors, they are reflected by obstacles such as walls, ceilings, and floors. Every reflection must be absorbed by the obstacle. In this way, when the sound source stops sounding, the sound wave is reflected and absorbed multiple times in the room, and finally disappears. We feel that there are still several sound waves mixed for a period of time after the sound source stops sounding. This phenomenon is called reverberation.
以反射三次为例,得到数字语音信号的混响信号的方法可以为:Taking the reflection three times as an example, the method for obtaining the reverberation signal of the digital speech signal can be:
将数字语音信号进行第一延时处理,生成第一信号;将数字语音信号进行第二延时处理和频域压缩滤波处理,生成第二信号;将数字语音信号进行第三延时处理和频域压缩滤波处理,生成第三信号;将第一声音信号、第二信号和第三信号合成第四信号作为混响信号。The digital speech signal is subjected to a first delay processing to generate a first signal; the digital speech signal is subjected to a second delay processing and a frequency domain compression filtering process to generate a second signal; and the digital speech signal is subjected to a third delay processing and frequency The domain compression filtering process generates a third signal; the first sound signal, the second signal, and the third signal are combined into a fourth signal as a reverberation signal.
其中,延时处理的次数和反射次数相同,可以不只是三次,可以为更多。Among them, the number of delay processing and the number of reflections are the same, and may be more than three times, and may be more.
在数字语音信号中加入该数字语音信号对应的混响信号,可以使耳机佩戴者体验到立体声的音效,提升用户体验。Adding a reverberation signal corresponding to the digital voice signal to the digital voice signal enables the earphone wearer to experience the stereo sound effect and enhance the user experience.
步骤S206,检测数字语音信号中是否包含振幅超过第一设定值的部分,如是则执行步骤S207;如否,则执行步骤S211。Step S206, detecting whether the digital speech signal includes a portion whose amplitude exceeds the first set value, and if yes, executing step S207; if not, executing step S211.
这样能够自动控制卡拉OK功能的开启。This automatically controls the opening of the karaoke function.
步骤S207,滤除耳机接收的数字音乐信号中的人声信号,得到背景音乐信号。Step S207, filtering out the human voice signal in the digital music signal received by the earphone to obtain a background music signal.
步骤S208,在背景音乐信号中加入该背景音乐信号对应的混响信号。Step S208, adding a reverberation signal corresponding to the background music signal to the background music signal.
为了实现卡拉OK的音效,可以在背景音乐信号中加入该背景音乐信号对应的混响。具体方法与在数字语音信号中加入该数字语音信号对应的混响的方法类似,在此不再赘述。In order to realize the sound effect of the karaoke, the reverberation corresponding to the background music signal may be added to the background music signal. The specific method is similar to the method of adding the reverberation corresponding to the digital voice signal in the digital voice signal, and details are not described herein again.
步骤S209,将背景音乐信号和数字语音信号进行叠加,得到数字叠加信号。In step S209, the background music signal and the digital voice signal are superimposed to obtain a digital superimposed signal.
步骤S210,将数字叠加信号进行数模转换处理后,发送至耳机听筒。In step S210, the digital superimposed signal is subjected to digital-to-analog conversion processing, and then sent to the earphone receiver.
步骤S211,将数字音乐信号进行数模转换处理后,发送至耳机听筒。In step S211, the digital music signal is subjected to digital-to-analog conversion processing, and then sent to the earphone.
本发明还提供了一种用于耳机的K歌处理装置,图3为该装置的一种实施结构的方框原理图。The present invention also provides a K song processing device for an earphone, and FIG. 3 is a block schematic diagram of an implementation structure of the device.
根据图3所示,该装置300包括接收模块301、模数转换模块302、滤除人声信号模块303、叠加模块304和第一数模转换模块305。
According to FIG. 3, the
上述接收模块301,用于接收耳机的麦克风采集的语音信号。The receiving module 301 is configured to receive a voice signal collected by a microphone of the earphone.
上述模数转换模块302,用于对所述语音信号进行模数转换处理,得到数字语音信号。The analog-to-digital conversion module 302 is configured to perform analog-to-digital conversion processing on the voice signal to obtain a digital voice signal.
上述滤除人声信号模块303,用于滤除所述耳机接收的数字音乐信号中的人声信号,得到背景音乐信号。The
上述叠加模块304,用于将所述背景音乐信号和所述数字语音信号进行叠加,得到数字叠加信号。The superimposing module 304 is configured to superimpose the background music signal and the digital speech signal to obtain a digital superimposed signal.
上述第一数模转换模块305,用于将所述数字叠加信号进行数模转换处理,得到模拟叠加信号,并将模拟叠加信号发送至耳机的听筒。The first digital-to-
进一步地,该装置300还包括检测模块和第二数模转换模块,该检测模块用于检测数字语音信号中是否包含检测数字语音信号中是否包含振幅超过第一设定值的部分,如是,则控制滤除人声信号模块滤除耳机接收的数字音乐信号中的人声信号,得到背景音乐信号;如否,则控制第二数模转换模块将数字音乐信号进行数模转换处理,得到模拟音乐信号,并将模拟音乐信号发送至耳机的听筒。Further, the
其中,接收模块包括第一接收单元和第二接收单元,该第一接收单元用于接收安装在一耳机听筒上的麦克风采集到的语音信号,作为第一语音信号;该第二接收单元用于接收安装在另一耳机听筒上的麦克风采集到的语音信号,作为第二语音信号。第一数模转换模块包括数模转换单元和波束形成单元,该数模转换单元用于对第一语音信号和第二语音信号分别进行模数转换处理,得到对应的第一数字语音信号和第二数字语音信号;该波束形成单元用于对第一数字语音信号和第二数字语音信号进行波束形成处理,得到数字语音信号。The receiving module includes a first receiving unit and a second receiving unit, where the first receiving unit is configured to receive a voice signal collected by a microphone mounted on a earphone as a first voice signal; the second receiving unit is configured to: Receiving a voice signal collected by a microphone mounted on another earphone as a second voice signal. The first digital-to-analog conversion module includes a digital-to-analog conversion unit and a beamforming unit, and the digital-to-analog conversion unit is configured to respectively perform analog-to-digital conversion processing on the first voice signal and the second voice signal to obtain a corresponding first digital voice signal and a first And a second digital voice signal; the beam forming unit is configured to perform beamforming processing on the first digital voice signal and the second digital voice signal to obtain a digital voice signal.
进一步地,波束形成单元包括波束形成子单元和消除子单元,该波束形成子单元用于对第一数字语音信号和第二数字语音信号进行波束形成处理;该消除子单元用于消除波束形成处理得到的信号中振幅超过第二设定值的部分,得到数字语音信号。Further, the beamforming unit includes a beamforming subunit and a cancellation subunit for performing beamforming processing on the first digital speech signal and the second digital speech signal; the cancellation subunit is configured to eliminate beamforming processing A portion of the obtained signal whose amplitude exceeds the second set value yields a digital speech signal.
具体的,该装置300还包括第一加混响模块和第二加混响模块,该第一加混响模块用于在背景音乐信号中加入背景音乐信号对应的混响信号;
该第二加混响模块用于在数字语音信号中加入数字语音信号对应的混响信号。Specifically, the
本发明还提供了一种耳机,在一方面,该耳机包括前述的通过耳机实现卡拉OK功能的装置300。The present invention also provides an earphone, which in one aspect comprises the
在另一方面,该耳机包括存储器和处理器,其中,存储器用于存储指令,该指令控制处理器进行操作以执行前述的通过耳机实现卡拉OK功能的方法。In another aspect, the earphone includes a memory and a processor, wherein the memory is for storing instructions that control the processor to perform the aforementioned method of implementing a karaoke function through the earphone.
该耳机包括耳机听筒1、耳机线2、控制盒3和耳机插头4,如图4所示,其中,该耳机为数字耳机,耳机插头4为USB插头。该耳机还包括用于采集语音信号的麦克风,如果该耳机具有一个麦克风,该麦克风可以安装在控制盒3上或者一个耳机听筒1上;如果该耳机具有两个麦克风,每一麦克风可以安装在不同的耳机听筒1上。The earphone includes an earphone earpiece 1, a
该处理器例如可以是DSP数字处理器。该存储器例如包括ROM(只读存储器)、RAM(随机存取存储器)、诸如硬盘的非易失性存储器等。The processor can be, for example, a DSP digital processor. The memory includes, for example, a ROM (Read Only Memory), a RAM (Random Access Memory), a nonvolatile memory such as a hard disk, and the like.
虽然已经通过例子对本发明的一些特定实施例进行了详细说明,但是本领域的技术人员应该理解,以上例子仅是为了进行说明,而不是为了限制本发明的范围。本领域的技术人员应该理解,可在不脱离本发明的范围和精神的情况下,对以上实施例进行修改。本发明的范围由所附权利要求来限定。 While the invention has been described in detail with reference to the preferred embodiments of the present invention, it is understood that It will be appreciated by those skilled in the art that the above embodiments may be modified without departing from the scope and spirit of the invention. The scope of the invention is defined by the appended claims.
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