WO2016176951A1 - 声音信号优化方法及装置 - Google Patents

声音信号优化方法及装置 Download PDF

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Publication number
WO2016176951A1
WO2016176951A1 PCT/CN2015/090275 CN2015090275W WO2016176951A1 WO 2016176951 A1 WO2016176951 A1 WO 2016176951A1 CN 2015090275 W CN2015090275 W CN 2015090275W WO 2016176951 A1 WO2016176951 A1 WO 2016176951A1
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Prior art keywords
sound
terminal
sound source
sound signal
specified
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Ceased
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PCT/CN2015/090275
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English (en)
French (fr)
Inventor
颜嘉甫
王楠楠
张鹏
张柳军
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Xiaomi Inc
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Xiaomi Inc
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Application filed by Xiaomi Inc filed Critical Xiaomi Inc
Priority to KR1020157031849A priority Critical patent/KR20160142217A/ko
Priority to RU2015155321A priority patent/RU2628473C2/ru
Priority to JP2017516025A priority patent/JP6314286B2/ja
Priority to MX2015017147A priority patent/MX361588B/es
Priority to BR112015031855A priority patent/BR112015031855A2/pt
Publication of WO2016176951A1 publication Critical patent/WO2016176951A1/zh
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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    • G—PHYSICS
    • G06—COMPUTING OR CALCULATING; COUNTING
    • G06F—ELECTRIC DIGITAL DATA PROCESSING
    • G06F3/00—Input arrangements for transferring data to be processed into a form capable of being handled by the computer; Output arrangements for transferring data from processing unit to output unit, e.g. interface arrangements
    • G06F3/16—Sound input; Sound output
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R5/00—Stereophonic arrangements
    • H04R5/04—Circuit arrangements, e.g. for selective connection of amplifier inputs/outputs to loudspeakers, for loudspeaker detection, or for adaptation of settings to personal preferences or hearing impairments
    • 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
    • G10L21/0272—Voice signal separating
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
    • H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
    • H04R1/40—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
    • H04R1/406—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R29/00—Monitoring arrangements; Testing arrangements
    • H04R29/007—Monitoring arrangements; Testing arrangements for public address systems
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers
    • H04R3/005—Circuits for transducers for combining the signals of two or more microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M3/00—Automatic or semi-automatic exchanges
    • H04M3/42—Systems providing special services or facilities to subscribers
    • H04M3/56—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
    • H04M3/568—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities audio processing specific to telephonic conferencing, e.g. spatial distribution, mixing of participants
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N7/00—Television systems
    • H04N7/14—Systems for two-way working
    • H04N7/15—Conference systems
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R2227/00—Details of public address [PA] systems covered by H04R27/00 but not provided for in any of its subgroups
    • H04R2227/007—Electronic adaptation of audio signals to reverberation of the listening space for PA
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/01—Aspects of volume control, not necessarily automatic, in sound systems
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/20—Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R2499/00—Aspects covered by H04R or H04S not otherwise provided for in their subgroups
    • H04R2499/10—General applications
    • H04R2499/11—Transducers incorporated or for use in hand-held devices, e.g. mobile phones, PDA's, camera's
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
    • H04S2400/15—Aspects of sound capture and related signal processing for recording or reproduction

Definitions

  • the present disclosure relates to the field of voice signal processing, and in particular, to a sound signal optimization method and apparatus.
  • terminals that support voice calls or recording functions such as smartphones, tablets, e-book readers, and electronic recorders have become an indispensable tool in people's lives, providing people with daily life. Great convenience.
  • multiple users can use the same recording support function terminal to record the conference, or multiple users can use the same terminal that supports voice calls to conduct a conference call with other users at the remote end;
  • the terminal stores the recorded sound signal or sends it to the remote user of the voice call.
  • the present disclosure provides a sound signal optimization method and apparatus.
  • the technical solution is as follows:
  • a sound signal optimization method for a terminal including at least two sound collection units, the method comprising:
  • Volume optimization is performed on the sound signal emitted by the specified sound source.
  • the volume optimization of the sound signal sent by the specified sound source includes:
  • the positioning, by the at least two sound collection units, the sound source around the terminal includes:
  • the filtering the specified sound source from the sound source determined by the positioning comprises:
  • the designated sound source is screened from the respective sound sources according to the intensity of the sound signals of the respective sound sources and/or the continuity of the sound signals.
  • the method before determining the sound signal sent by the specified sound source in the sound signal collected by the terminal, the method further includes:
  • Determining, by the determining, the sound signal emitted by the specified sound source in the sound signal collected by the terminal comprising:
  • the posture of the terminal includes an angle between a front surface of the terminal and a horizontal plane, and the gain compensation is performed on the sound signal collected by the terminal according to the posture of the terminal, including:
  • Determining a compensation value according to an angle between a front surface of the terminal and a horizontal plane, and an angle between the front surface and the horizontal plane of the terminal is proportional to the compensation value
  • a sound signal optimization apparatus for use in a terminal including at least two sound collection units, the apparatus comprising:
  • a positioning module configured to locate a sound source around the terminal by using the at least two sound collection units
  • a screening module configured to filter out a specified sound source from each sound source determined by the positioning
  • a sound signal determining module configured to determine, according to a spatial position of the specified sound source obtained by the positioning, a sound signal emitted by the specified sound source in the sound signal collected by the terminal;
  • an optimization module configured to perform volume optimization on the sound signal emitted by the specified sound source.
  • the optimization module includes:
  • a comparison submodule configured to compare an intensity of the sound signal emitted by the specified sound source with a preset first intensity threshold and a second intensity threshold, the first intensity threshold being greater than or equal to the second intensity threshold;
  • a reducing submodule configured to reduce an intensity of the sound signal emitted by the specified sound source to the first intensity threshold if the intensity of the sound signal emitted by the specified sound source is greater than the first intensity threshold
  • the positioning module includes:
  • Obtaining a sub-module configured to acquire a sound signal sent by a sound source to achieve a time difference and a phase difference between the at least two sound collecting units;
  • a positioning submodule configured to determine a spatial position of the sound source relative to the terminal according to a time difference and a phase difference of the at least two sound collecting units according to a sound signal emitted by the sound source.
  • the screening module is configured to filter the specified sound source from the respective sound sources according to the intensity of the sound signal of the respective sound source and/or the continuity of the sound signal.
  • the device further includes: a posture acquiring module and a compensation module;
  • the gesture acquiring module is configured to acquire the terminal of the terminal by using an attitude sensor in the terminal before the sound signal determining module determines the sound signal sent by the specified sound source in the sound signal collected by the terminal attitude;
  • the compensation module is configured to perform gain compensation on the sound signal collected by the terminal according to the posture of the terminal;
  • the sound signal determining module is configured to determine a sound signal emitted by the specified sound source in the sound signal after the gain compensation.
  • the compensation module includes:
  • a numerical determination submodule configured to determine a compensation value according to an angle between a front surface of the terminal and a horizontal plane, wherein an angle between the front surface and the horizontal plane of the terminal is proportional to the compensation value;
  • a compensation submodule configured to perform gain compensation on the sound signal collected by the terminal according to the compensation value
  • the posture of the terminal includes an angle between a front surface of the terminal and a horizontal plane.
  • a sound signal optimization method apparatus for a terminal including at least two sound collection units, the apparatus comprising:
  • a memory for storing processor executable instructions
  • processor is configured to:
  • Volume optimization is performed on the sound signal emitted by the specified sound source.
  • the sound source around the terminal is positioned by at least two sound collecting units, and the specified sound source is selected from each sound source determined by the positioning, and the sound signal collected by the terminal is determined according to the spatial position of the specified sound source obtained by the positioning.
  • the sound signal emitted by the designated sound source is used to optimize the sound signal of the specified sound source; when the terminal collects the sound signal, the peripheral sound source is screened, and the sound signal sent by the selected sound source is Optimize to improve the collection of sound signals from the terminal.
  • FIG. 1 is a flowchart of a sound signal optimization method according to an exemplary embodiment
  • FIG. 2 is a flowchart of a sound signal optimization method according to another exemplary embodiment
  • FIG. 3 is a flowchart of a positioning method according to another exemplary embodiment
  • FIG. 4 is a flowchart of a gain compensation method according to another exemplary embodiment
  • FIG. 5 is a block diagram of an audio signal optimization apparatus according to an exemplary embodiment
  • FIG. 6 is a block diagram showing an apparatus for optimizing a sound signal according to another exemplary embodiment
  • FIG. 7 is a block diagram of an apparatus, according to an exemplary embodiment.
  • FIG. 1 is a flow chart showing a sound signal optimization method according to an exemplary embodiment.
  • the sound signal optimization method can be used in a terminal including at least two sound collection units, and the terminal can be a smart terminal, such as a smart phone, a tablet computer, an e-book reader, and a smart voice recorder, which supports a voice call or a recording function.
  • the sound signal optimization method may include the following steps.
  • step 102 the sound source around the terminal is positioned by at least two sound collection units.
  • step 104 the designated sound source is selected from the respective sound sources determined by the positioning.
  • step 106 based on the spatial position of the specified sound source obtained by the positioning, the sound signal emitted by the designated sound source in the sound signal collected by the terminal is determined.
  • step 108 the sound signal emitted by the designated sound source is volume optimized.
  • the collected sound signal of the specified sound source is volume optimized, including:
  • the intensity of the sound signal emitted by the specified sound source is greater than the first intensity threshold, reducing the intensity of the sound signal emitted by the specified sound source to the first intensity threshold;
  • the sound source of the terminal is located by the at least two sound collection units, including:
  • the selected sound source is selected from the sound source determined by the positioning, including:
  • the designated sound source is selected from the respective sound sources based on the intensity of the sound signals of the respective sound sources and/or the continuity of the sound signals.
  • the method before determining the sound signal sent by the specified sound source in the sound signal collected by the terminal, the method further includes:
  • the sound signal sent by the specified sound source in the sound signal collected by the terminal includes:
  • the posture of the terminal includes an angle between the front surface of the terminal and the horizontal plane, and the sound signal collected by the terminal is compensated according to the posture of the terminal, including:
  • Determining the compensation value according to the angle between the front surface of the terminal and the horizontal plane, and the angle between the front surface and the horizontal plane of the terminal is proportional to the compensation value
  • the sound signal collected by the terminal is subjected to gain compensation according to the compensation value.
  • the sound signal optimization method locates the sound source around the terminal through at least two sound collection units, and selects a specified sound source from each sound source determined by the positioning, and obtains the specified sound source according to the positioning.
  • the spatial position of the designated sound source determines a sound signal emitted by the designated sound source in the sound signal collected by the terminal, and performs volume optimization on the sound signal emitted by the specified sound source; when the terminal collects the sound signal, The surrounding sound source is screened, and the sound signal emitted by the selected sound source is optimized to improve the collection effect of the terminal on the sound signal.
  • FIG. 2 is a flowchart of a sound signal optimization method according to another exemplary embodiment.
  • the sound signal optimization method can be used in a terminal including at least two sound collection units, and the terminal can be a smart terminal, such as a smart phone, a tablet computer, an e-book reader, and a smart voice recorder, which supports a voice call or a recording function.
  • the sound signal optimization method may include the following steps.
  • step 202 the sound source around the terminal is positioned by at least two sound collection units.
  • step of positioning the sound source around the terminal by the at least two sound collecting units may be as follows:
  • step 202a a sound signal emitted by a sound source is obtained to reach a time difference and a phase difference of the at least two sound collecting units.
  • the at least two sound collection units included in the terminal form an array of sound collection units. Since the distances and directions of the sound sources to the sound collection units in the sound collection unit array are different, the sound emitted by the sound source is The time and phase passed to each sound collection unit are also different.
  • the terminal can acquire the time difference and phase difference of the sounds collected by the respective sound collection units in the sound collection unit array collected by the same sound source.
  • step 202b the time difference and phase difference of the at least two sound collection units are determined according to the sound signal emitted by the sound source to determine the spatial position of the sound source relative to the terminal.
  • the terminal can use beamforming technology to locate the position of the sound source, for example, the sound that the terminal can emit according to the sound source.
  • the signal reaches a time difference and a phase difference of the at least two sound collection units, and the position of the sound source relative to the terminal is identified by an algorithm such as delay summation or eigenvalue decomposition.
  • the terminal can not only accurately locate the spatial position of the fixed sound source, but also track the moving sound source to continuously optimize the signal sent by the moving sound source.
  • the designated sound source is selected from the respective sound sources according to the intensity of the sound signal of the respective sound source and/or the continuity of the sound signal.
  • the terminal When the terminal is in the state of a call or a recording, the sound signal sent by the user is usually a valid sound signal, and the other sound signals are mostly noise.
  • the terminal can be based on each sound source. The intensity of the sound signal and/or the continuity of the sound signal selects the sound source most likely to be the user from each sound source, and then only optimizes the sound source that is most likely to be filtered by the user.
  • the terminal may set different weights for different strengths of the sound signal, and set different weights for different continuous times of the sound signal. After receiving the sound signal from a sound source, the terminal determines the weight and duration of the sound intensity respectively. The weights corresponding to the time are weighted, and each sound source is sorted according to the corresponding weight, and the specified sound source with the highest weight is determined.
  • the intensity of the closest human voice can be set to a higher weight, and the difference between the strength of the spoken voice and the voice is greater, and the corresponding weight is set.
  • the lower For example, if a person's normal speaking sound intensity is about 40 decibels, the highest weighting can be set for 40 decibels, and the farther away from the sound intensity of 40 decibels, the lower the weight.
  • the duration of the sound signal closest to the person's speech can be set to a higher weight, and the difference between the duration of the sound signal when talking to the person is more Large, the lower the weight of the corresponding settings. For example, when a person speaks, the duration of a normal sentence may be 2 s, then the highest weighting can be set for the duration of 2 s, and the longer the difference between the two s, the lower the weight.
  • the quantity for example, filters out three specified sound sources at the same time, and optimizes only the sound signals from the three sound sources at the same time.
  • step 206 the sound signal emitted by the designated sound source in the sound signal collected by the terminal is determined according to the spatial position of the specified sound source obtained by the positioning.
  • the terminal When the terminal processes the received sound signal, it can process only the sound signal emitted by the specified sound source according to the spatial position of the specified sound source.
  • step 208 the intensity of the sound signal emitted by the specified sound source is compared with a preset first intensity threshold and a second intensity threshold, the first intensity threshold being greater than or equal to the second intensity threshold.
  • step 210 if the intensity of the sound signal emitted by the designated sound source is greater than the first intensity threshold, the intensity of the sound signal emitted by the designated sound source is reduced to the first intensity threshold.
  • step 212 if the intensity of the sound signal emitted by the specified sound source is less than the second intensity threshold, the finger is The intensity of the sound signal emitted by the fixed sound source is increased to the second intensity threshold.
  • the intensity of the sound signal emitted by the specified sound source is not processed.
  • the terminal may set the first intensity threshold and the second intensity threshold according to the sound signal strength when the person speaks normally. If the sound signal strength of the specified sound source is large, the intensity of the sound signal of the specified sound source is lowered, if the sound source is specified. If the sound signal strength is small, the intensity of the sound signal of the specified sound source is increased. If the sound signal strength of the designated sound source is moderate, the intensity of the sound signal of the specified sound source is not processed. Thereby, the intensity of the sound signal, which may be the user's designated sound source, is kept within a predetermined range, neither too high nor too low, so that the effect of recording or talking is optimized.
  • the smart phone For example, if user A and user B make a remote conference call through a smart phone, or use a mobile phone to record a conference as an example, when the smart phone enters the recording state or the call state, the smart phone is switched to the conference recording mode.
  • the mobile phone collects surrounding sounds through at least two microphones and determines the position of each sound source.
  • the smart phone filters the speaker into a designated sound source, and performs sound signals emitted by the speaker when speaking.
  • the sound signal optimization method locates the sound source around the terminal through at least two sound collection units, and selects a specified sound source from each sound source determined by the positioning, and obtains the specified sound source according to the positioning.
  • the spatial position of the designated sound source determines a sound signal emitted by the specified sound source in the sound signal collected by the terminal, and adjusts the intensity of the sound signal emitted by the specified sound source to two preset intensity thresholds.
  • the intensity of the sound signal emitted by the specified sound source collected by the terminal is adjusted to an appropriate range, and the sound signal is collected from the terminal during recording or talking.
  • steps 205a to 205c in the flowchart of the gain compensation method shown in FIG. 4 may also be performed.
  • step 205a the posture of the terminal is acquired by an attitude sensor in the terminal, and the posture of the terminal includes an angle between the front surface of the terminal and a horizontal plane.
  • the terminal Under normal circumstances, when recording or conference call, the terminal is placed horizontally on the desktop horizontally, and in some cases, the posture of the terminal may not be horizontal, for example, when the terminal is placed on a book or other object.
  • the front side is not in a horizontal state.
  • there is a certain angle between the front surface and the horizontal plane of the terminal which causes a certain deviation between the sound signal collected by the terminal and the ideal state, and is collected when the front side of the terminal is in a horizontal state.
  • the sound signal is relatively small.
  • the terminal may first compensate the collected sound signal according to the current posture.
  • step 205b the compensation value is determined according to the angle between the front surface of the terminal and the horizontal plane, and the angle between the front surface and the horizontal plane of the terminal is proportional to the compensation value.
  • the terminal can be based on the front side and the horizontal plane of the terminal.
  • the angle of the calculation is calculated by a preset algorithm, or the value to be compensated is queried by a pre-stored correspondence.
  • step 205c the sound signal collected by the terminal is subjected to gain compensation according to the compensation value.
  • the terminal After the terminal determines the compensation value, it performs gain compensation on all the collected sound signals.
  • the terminal when determining, by the terminal, the sound signal emitted by the designated sound source in the sound signal collected by the terminal, the terminal may determine the sound signal emitted by the designated sound source in the sound signal after the gain compensation.
  • the method provided in the embodiment of the present disclosure acquires the posture of the terminal and determines the posture of the terminal according to the posture of the terminal, before determining and optimizing the sound signal of the specified sound source from the collected sound signal.
  • the sound signal is compensated by the gain, which further improves the collection effect of the sound signal when the terminal is recording or talking.
  • FIG. 5 is a block diagram of an apparatus for optimizing a sound signal, which may be used in a terminal including at least two sound collection units, which may be a smartphone, a tablet, or the like, according to an exemplary embodiment.
  • An intelligent terminal supporting an audio call or a recording function such as an e-book reader and a smart voice recorder, performs the method shown in any of FIGS. 1 to 4.
  • the sound signal optimization device includes, but is not limited to, a positioning module 501, a screening module 502, a sound signal determining module 503, and an optimization module 504;
  • the positioning module 501 is configured to locate a sound source around the terminal by the at least two sound collection units;
  • the screening module 502 is configured to filter out a specified sound source from each of the sound sources determined by the positioning;
  • the sound signal determining module 503 is configured to determine a sound signal emitted by the specified sound source in the sound signal collected by the terminal according to the spatial position of the specified sound source obtained by the positioning;
  • the optimization module 504 is configured to perform volume optimization on a sound signal emitted by the specified sound source.
  • the sound signal optimization device locates the sound source around the terminal through at least two sound collection units, and selects the specified sound source from each sound source determined by the positioning, and obtains the specified sound source according to the positioning.
  • the spatial position of the designated sound source determines a sound signal emitted by the designated sound source in the sound signal collected by the terminal, and performs volume optimization on the sound signal emitted by the specified sound source; when the terminal collects the sound signal, The surrounding sound source is screened, and the sound signal emitted by the selected sound source is optimized to improve the collection effect of the terminal on the sound signal.
  • FIG. 6 is a block diagram of an audio signal optimization apparatus according to an exemplary embodiment, the sound signal optimization apparatus
  • the device can be used in a terminal including at least two sound collection units, and the terminal can be a smart terminal, such as a smart phone, a tablet computer, an e-book reader, and a smart voice recorder, which supports voice call or recording function, and is executed as shown in FIG. 1 to 4 Any of the methods shown.
  • the sound signal optimization device includes, but is not limited to, a positioning module 501, a screening module 502, a sound signal determining module 503, and an optimization module 504;
  • the positioning module 501 is configured to locate a sound source around the terminal by the at least two sound collection units;
  • the screening module 502 is configured to filter out a specified sound source from each of the sound sources determined by the positioning;
  • the sound signal determining module 503 is configured to determine a sound signal emitted by the specified sound source in the sound signal collected by the terminal according to the spatial position of the specified sound source obtained by the positioning;
  • the optimization module 504 is configured to perform volume optimization on a sound signal emitted by the specified sound source.
  • the optimization module 504 includes: a comparison submodule 504a, a reduction submodule 504b, and an addition submodule 504c;
  • the comparison sub-module 504a is configured to compare an intensity of a sound signal emitted by the specified sound source with a preset first intensity threshold and a second intensity threshold, the first intensity threshold being greater than or equal to Second intensity threshold;
  • the reducing submodule 504b is configured to reduce the intensity of the sound signal emitted by the specified sound source to the first if the intensity of the sound signal emitted by the specified sound source is greater than the first intensity threshold Intensity threshold
  • the adding sub-module 504c is configured to increase the intensity of the sound signal emitted by the specified sound source to the second if the intensity of the sound signal emitted by the specified sound source is less than the second intensity threshold Intensity threshold.
  • the positioning module 501 includes: an obtaining submodule 501a and a positioning submodule 501b;
  • the obtaining sub-module 501a is configured to acquire a sound signal emitted by a sound source to achieve a time difference and a phase difference of the at least two sound collecting units;
  • the positioning sub-module 501b is configured to determine a spatial position of the sound source relative to the terminal according to a time difference and a phase difference of the at least two sound collecting units according to a sound signal emitted by the sound source.
  • the screening module 502 is configured to filter the specified sound source from the respective sound sources according to the intensity of the sound signal of the respective sound source and/or the continuity of the sound signal.
  • the device further includes: a posture acquiring module 505 and a compensation module 506;
  • the gesture acquiring module 505 is configured to obtain, by the attitude sensor in the terminal, before the sound signal determining module 503 determines a sound signal emitted by the specified sound source in the sound signal collected by the terminal The attitude of the terminal;
  • the compensation module 506 is configured to perform gain compensation on the sound signal collected by the terminal according to the posture of the terminal;
  • the sound signal determining module 503 is configured to determine a sound signal emitted by the specified sound source among the gain-compensated sound signals.
  • the compensation module 506 includes: a value determining submodule 506a and a compensating submodule 506b;
  • the value determining sub-module 506a is configured to determine a compensation value according to an angle between a front surface of the terminal and a horizontal plane, and an angle between the front surface and the horizontal plane of the terminal is proportional to the compensation value;
  • the compensation sub-module 506b is configured to perform gain compensation on the sound signal collected by the terminal according to the compensation value
  • the posture of the terminal includes an angle between a front surface of the terminal and a horizontal plane.
  • the sound signal optimization device locates the sound source around the terminal through at least two sound collection units, and selects the specified sound source from each sound source determined by the positioning, and obtains the specified sound source according to the positioning.
  • the spatial position of the designated sound source determines a sound signal emitted by the specified sound source in the sound signal collected by the terminal, and adjusts the intensity of the sound signal emitted by the specified sound source to two preset intensity thresholds.
  • the intensity of the sound signal emitted by the specified sound source collected by the terminal is adjusted to an appropriate range, and the sound signal is collected from the terminal during recording or talking.
  • the apparatus provided in the embodiment of the present disclosure acquires the posture of the terminal and determines the sound signal collected by the terminal according to the posture of the terminal, before determining and optimizing the sound signal of the specified sound source from the collected sound signals.
  • the gain compensation further improves the collection effect of the sound signal when the terminal is recording or talking.
  • FIG. 7 is a block diagram of an apparatus 700, according to an exemplary embodiment.
  • the device comprises at least two sound collecting units.
  • device 700 can be a mobile phone, a computer, a digital broadcast terminal, a messaging device, a routing device, a gaming console, a tablet device, a medical device, a fitness device, a personal digital assistant, and the like.
  • apparatus 700 can include one or more of the following components: processing component 702, memory 704, power component 706, multimedia component 708, audio component 710, input/output (I/O) interface 712, sensor component 714, And a communication component 716.
  • Processing component 702 typically controls the overall operation of device 700, such as operations associated with display, telephone calls, data communications, camera operations, and recording operations.
  • Processing component 702 can include one or more processors 718 to execute instructions to perform all or part of the steps of the methods described above.
  • processing component 702 can include one or more modules to facilitate interaction between component 702 and other components.
  • processing component 702 can include a multimedia module to facilitate interaction between multimedia component 708 and processing component 702.
  • Memory 704 is configured to store various types of data to support operation at device 700. Examples of such data include instructions for any application or method operating on device 700, contact data, phone book data, messages, pictures, videos, and the like. Memory 704 can be implemented by any type of volatile or non-volatile storage device, or a combination thereof, such as static random access memory (SRAM), electrically erasable programmable read only memory (EEPROM), erasable Programmable Read Only Memory (EPROM), Programmable Read Only Memory (PROM), Read Only Memory (ROM), Magnetic Memory, Flash Memory, Disk or Optical Disk. Also stored in memory 704 is one or more modules configured to be executed by the one or more processors 720 to perform all or part of the steps of the method illustrated in any of Figures 1 through 4 above. .
  • SRAM static random access memory
  • EEPROM electrically erasable programmable read only memory
  • EPROM erasable Programmable Read Only Memory
  • PROM Programmable Read Only Memory
  • ROM Read Only Memory
  • Magnetic Memory Flash Memory
  • Power component 706 provides power to various components of device 700.
  • the power component 706 can include a power management system.
  • the multimedia component 708 includes a screen between the device 700 and the user that provides an output interface.
  • the screen can include a liquid crystal display (LCD) and a touch panel (TP). If the screen includes a touch panel, the screen can be implemented as a touch screen to receive input signals from the user.
  • the touch panel includes one or more touch sensors to sense touches, slides, and gestures on the touch panel. The touch sensor may sense not only the boundary of the touch or sliding action, but also the duration and pressure associated with the touch or slide operation.
  • the multimedia component 708 includes a front camera and/or a rear camera. When the device 700 is in an operation mode, such as a shooting mode or a video mode, the front camera and/or the rear camera can receive external multimedia data. Each front and rear camera can be a fixed optical lens system or have focal length and optical zoom capabilities.
  • the audio component 710 is configured to output and/or input an audio signal.
  • audio component 710 includes a microphone (MIC) that is configured to receive an external audio signal when device 700 is in an operational mode, such as a call mode, a recording mode, and a voice recognition mode.
  • the received audio signal may be further stored in memory 704 or transmitted via communication component 716.
  • audio component 710 also includes a speaker for outputting an audio signal.
  • the I/O interface 712 provides an interface between the processing component 702 and the peripheral interface module, which may be a keyboard, a click wheel, a button, or the like. These buttons may include, but are not limited to, a home button, a volume button, a start button, and a lock button.
  • Sensor assembly 714 includes one or more sensors for providing device 700 with various aspects of status assessment.
  • sensor assembly 714 can detect an open/closed state of device 700, relative positioning of components, such as the display and keypad of device 700, and sensor component 714 can also detect a change in position of one component of device 700 or device 700. The presence or absence of user contact with device 700, device 700 orientation or acceleration/deceleration, and temperature variation of device 700.
  • Sensor assembly 714 can include a proximity sensor configured to detect the presence of nearby objects without any physical contact.
  • Sensor component 714 can also include a light sensor, such as a CMOS or CCD image sensor, for use in imaging applications.
  • the sensor component 714 can also include an acceleration sensor, a gyro sensor, a magnetic sensor, a pressure sensor, or a temperature sensor.
  • Communication component 716 is configured to facilitate wired or wireless communication between device 700 and other devices.
  • the device 700 can access a wireless network based on a communication standard, such as WiFi, 2G or 3G, or a combination thereof.
  • communication component 716 receives broadcast signals or broadcast associated information from an external broadcast management system via a broadcast channel.
  • the communication component 716 also includes a near field communication (NFC) module to facilitate short range communication.
  • NFC near field communication
  • the NFC module can be implemented based on radio frequency identification (RFID) technology, infrared data association (IrDA) technology, ultra-wideband (UWB) technology, Bluetooth (BT) technology, and other technologies.
  • RFID radio frequency identification
  • IrDA infrared data association
  • UWB ultra-wideband
  • Bluetooth Bluetooth
  • apparatus 700 may be implemented by one or more application specific integrated circuits (ASICs), digital signal processors (DSPs), digital signal processing devices (DSPDs), programmable logic devices (PLDs), field programmable A gate array (FPGA), controller, microcontroller, microprocessor, or other electronic component implementation for performing the above methods.
  • ASICs application specific integrated circuits
  • DSPs digital signal processors
  • DSPDs digital signal processing devices
  • PLDs programmable logic devices
  • FPGA field programmable A gate array
  • controller microcontroller, microprocessor, or other electronic component implementation for performing the above methods.
  • non-transitory computer readable storage medium comprising instructions, such as a memory 704 comprising instructions executable by processor 718 of apparatus 700 to perform the above method.
  • the non-transitory computer readable storage medium may be a ROM, a random access memory (RAM), a CD-ROM, a magnetic tape, a floppy disk, and an optical data storage device.

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Abstract

一种声音信号优化方法及装置，属于语音信号处理领域。所述方法用于包含有至少两个声音采集单元的终端中，包括：通过所述至少两个声音采集单元对所述终端周边的声源进行定位（102）；从定位确定的各个声源中筛选出指定声源（104）；根据定位获得的所述指定声源的空间位置，确定所述终端采集到的声音信号中由所述指定声源发出的声音信号（106）；对所述指定声源发出的声音信号进行音量优化（108）。所述方法及装置在采集声音信号时，通过对周边声源进行筛选，并对筛选出的声源发出的声音信号进行优化，从提高终端对声音信号的采集效果。

Description

声音信号优化方法及装置
本申请基于申请号为201510226844.2、申请日为2015年05月06日的中国专利申请提出，并要求该中国专利申请的优先权，该中国专利申请的全部内容在此引入本申请作为参考。
技术领域
本公开涉及语音信号处理领域，特别涉及一种声音信号优化方法及装置。
背景技术
随着通讯和电子技术的不断发展，智能手机、平板电脑、电子书阅读器以及电子录音笔等支持语音通话或者录音功能的终端逐渐成为人们生活中不可缺少的工具，为人们的日常生活提供了极大的便利。
在此类终端的一种应用方式中，多个用户可以使用同一个支持录音功能终端进行会议录音，或者，多个用户可以使用同一个支持语音通话的终端与远端的其它用户进行电话会议；终端将录制的声音信号进行存储，或者，发送给语音通话的远端用户。
发明内容
本公开提供了一种声音信号优化方法及装置。所述技术方案如下：
第一方面，提供一种声音信号优化方法，用于包含有至少两个声音采集单元的终端中，所述方法包括：
通过所述至少两个声音采集单元对所述终端周边的声源进行定位；
从定位确定的各个声源中筛选出指定声源；
根据定位获得的所述指定声源的空间位置，确定所述终端采集到的声音信号中由所述指定声源发出的声音信号；
对所述指定声源发出的声音信号进行音量优化。
可选的，所述对所述指定声源发出的声音信号进行音量优化，包括：
将所述指定声源发出的声音信号的强度与预设的第一强度阈值和第二强度阈值进行比较，所述第一强度阈值大于或者等于所述第二强度阈值；
若所述指定声源发出的声音信号的强度大于所述第一强度阈值，则将所述指定声源发出的声音信号的强度降低至所述第一强度阈值；
若所述指定声源发出的声音信号的强度小于所述第二强度阈值，则将所述指定声源发出的声音信号的强度增加至所述第二强度阈值。
可选的，所述通过所述至少两个声音采集单元对所述终端周边的声源进行定位，包括：
获取一声源发出的声音信号达到所述至少两个声音采集单元的时间差和相位差；
根据所述声源发出的声音信号达到所述至少两个声音采集单元的时间差和相位差确定所述声源相对于所述终端的空间位置。
可选的，所述从定位确定的声源中筛选出指定声源，包括：
根据所述各个声源的声音信号的强度和/或声音信号的连续性，从所述各个声源中筛选出所述指定声源。
可选的，所述确定所述终端采集到的声音信号中由所述指定声源发出的声音信号之前，还包括：
通过所述终端中的姿态传感器获取所述终端的姿态；
根据所述终端的姿态对所述终端采集到的声音信号进行增益补偿；
所述确定所述终端采集到的声音信号中由所述指定声源发出的声音信号，包括：
确定经过增益补偿后的声音信号中，由所述指定声源发出的声音信号。
可选的，所述终端的姿态包括所述终端正面与水平面之间的夹角，所述根据所述终端的姿态对所述终端采集到的声音信号进行增益补偿，包括：
根据所述终端正面与水平面之间的夹角确定补偿数值，所述终端正面与水平面之间的夹角与所述补偿数值之间成正比例关系；
根据所述补偿数值对所述终端采集到的声音信号进行增益补偿。
在第二方面，提供一种声音信号优化装置，用于包含有至少两个声音采集单元的终端中，所述装置包括：
定位模块，用于通过所述至少两个声音采集单元对所述终端周边的声源进行定位；
筛选模块，用于从定位确定的各个声源中筛选出指定声源；
声音信号确定模块，用于根据定位获得的所述指定声源的空间位置，确定所述终端采集到的声音信号中由所述指定声源发出的声音信号；
优化模块，用于对所述指定声源发出的声音信号进行音量优化。
可选的，所述优化模块，包括：
比较子模块，用于将所述指定声源发出的声音信号的强度与预设的第一强度阈值和第二强度阈值进行比较，所述第一强度阈值大于或者等于所述第二强度阈值；
降低子模块，用于若所述指定声源发出的声音信号的强度大于所述第一强度阈值，则将所述指定声源发出的声音信号的强度降低至所述第一强度阈值；
增加子模块，用于若所述指定声源发出的声音信号的强度小于所述第二强度阈值，则将所述指定声源发出的声音信号的强度增加至所述第二强度阈值。
可选的，所述定位模块，包括：
获取子模块，用于获取一声源发出的声音信号达到所述至少两个声音采集单元的时间差和相位差；
定位子模块，用于根据所述声源发出的声音信号达到所述至少两个声音采集单元的时间差和相位差确定所述声源相对于所述终端的空间位置。
可选的，所述筛选模块，用于根据所述各个声源的声音信号的强度和/或声音信号的连续性，从所述各个声源中筛选出所述指定声源。
可选的，所述装置还包括：姿态获取模块和补偿模块；
所述姿态获取模块，用于在所述声音信号确定模块确定所述终端采集到的声音信号中由所述指定声源发出的声音信号之前，通过所述终端中的姿态传感器获取所述终端的姿态；
所述补偿模块，用于根据所述终端的姿态对所述终端采集到的声音信号进行增益补偿；
所述声音信号确定模块，用于确定经过增益补偿后的声音信号中，由所述指定声源发出的声音信号。
可选的，所述补偿模块，包括：
数值确定子模块，用于根据所述终端正面与水平面之间的夹角确定补偿数值，所述终端正面与水平面之间的夹角与所述补偿数值之间成正比例关系；
补偿子模块，用于根据所述补偿数值对所述终端采集到的声音信号进行增益补偿；
其中，所述终端的姿态包括所述终端正面与水平面之间的夹角。
第三方面，提供一种声音信号优化方法装置，用于包含有至少两个声音采集单元的终端中，所述装置包括：
处理器；
用于存储处理器可执行指令的存储器；
其中，所述处理器被配置为：
通过所述至少两个声音采集单元对所述终端周边的声源进行定位；
从定位确定的各个声源中筛选出指定声源；
根据定位获得的所述指定声源的空间位置，确定所述终端采集到的声音信号中由所述指定声源发出的声音信号；
对由所述指定声源发出的声音信号进行音量优化。
本公开的实施例提供的技术方案可以包括以下有益效果：
通过至少两个声音采集单元对终端周边的声源进行定位，从定位确定的各个声源中筛选出指定声源，根据定位获得的该指定声源的空间位置，确定该终端采集到的声音信号中由该指定声源发出的声音信号，对该指定声源发出的声音信号进行音量优化；当终端在采集声音信号时，对周边声源进行筛选，并对筛选出的声源发出的声音信号进行优化，从提高终端对声音信号的采集效果。
应当理解的是，以上的一般描述和后文的细节描述仅是示例性的，并不能限制本公开。
附图说明
此处的附图被并入说明书中并构成本说明书的一部分，示出了符合本发明的实施例， 并于说明书一起用于解释本发明的原理。
图1是根据一示例性实施例示出的一种声音信号优化方法的流程图；
图2是根据另一示例性实施例示出的一种声音信号优化方法的流程图；
图3是根据另一示例性实施例示出的一种定位方法的流程图；
图4是根据另一示例性实施例示出的一种增益补偿方法的流程图；
图5是根据一示例性实施例示出的一种声音信号优化装置的框图；
图6是根据另一示例性实施例示出的一种声音信号优化装置的框图；
图7是根据一示例性实施例示出的一种装置的框图。
具体实施方式
这里将详细地对示例性实施例执行说明，其示例表示在附图中。下面的描述涉及附图时，除非另有表示，不同附图中的相同数字表示相同或相似的要素。以下示例性实施例中所描述的实施方式并不代表与本发明相一致的所有实施方式。相反，它们仅是与如所附权利要求书中所详述的、本发明的一些方面相一致的装置和方法的例子。
图1是根据一示例性实施例示出的一种声音信号优化方法的流程图。该声音信号优化方法可以用于包含有至少两个声音采集单元的终端中，该终端可以是智能手机、平板电脑、电子书阅读器以及智能录音笔等支持语音通话或者录音功能的智能终端。如图1所示，该声音信号优化方法可以包括以下步骤。
在步骤102中，通过至少两个声音采集单元对终端周边的声源进行定位。
在步骤104中，从定位确定的各个声源中筛选出指定声源。
在步骤106中，根据定位获得的该指定声源的空间位置，确定该终端采集到的声音信号中由该指定声源发出的声音信号。
在步骤108中，对该指定声源发出的声音信号进行音量优化。
可选的，该对采集到的该指定声源发出的声音信号进行音量优化，包括：
将该指定声源发出的声音信号的强度与预设的第一强度阈值和第二强度阈值进行比较，该第一强度阈值大于或者等于该第二强度阈值；
若该指定声源发出的声音信号的强度大于该第一强度阈值，则将该指定声源发出的声音信号的强度降低至该第一强度阈值；
若该指定声源发出的声音信号的强度小于该第二强度阈值，则将该指定声源发出的声音信号的强度增加至该第二强度阈值。
可选的，该通过该至少两个声音采集单元对该终端周边的声源进行定位，包括：
获取一声源发出的声音信号达到该至少两个声音采集单元的时间差和相位差；
根据该声源发出的声音信号达到该至少两个声音采集单元的时间差和相位差确定该声源相对于该终端的空间位置。
可选的，该从定位确定的声源中筛选出指定声源，包括：
根据该各个声源的声音信号的强度和/或声音信号的连续性，从该各个声源中筛选出指定声源。
可选的，该确定终端采集到的声音信号中由指定声源发出的声音信号之前，还包括：
通过该终端中的姿态传感器获取该终端的姿态；
根据该终端的姿态对该终端采集到的声音信号进行增益补偿；
该确定终端采集到的声音信号中由指定声源发出的声音信号，包括：
确定经过增益补偿后的声音信号中，由该指定声源发出的声音信号。
可选的，该终端的姿态包括该终端正面与水平面之间的夹角，该根据该终端的姿态对该终端采集到的声音信号进行增益补偿，包括：
根据该终端正面与水平面之间的夹角确定补偿数值，该终端正面与水平面之间的夹角与该补偿数值之间成正比例关系；
根据该补偿数值对该终端采集到的声音信号进行增益补偿。
综上所述，本公开实施例中提供的声音信号优化方法，通过至少两个声音采集单元对终端周边的声源进行定位，从定位确定的各个声源中筛选出指定声源，根据定位获得的该指定声源的空间位置，确定该终端采集到的声音信号中由该指定声源发出的声音信号，对该指定声源发出的声音信号进行音量优化；当终端在采集声音信号时，对周边声源进行筛选，并对筛选出的声源发出的声音信号进行优化，从提高终端对声音信号的采集效果。
图2是根据另一示例性实施例示出的一种声音信号优化方法的流程图。该声音信号优化方法可以用于包含有至少两个声音采集单元的终端中，该终端可以是智能手机、平板电脑、电子书阅读器以及智能录音笔等支持语音通话或者录音功能的智能终端。如图2所示，该声音信号优化方法可以包括以下步骤。
在步骤202中，通过至少两个声音采集单元对终端周边的声源进行定位。
可选的，请参考图3所示的定位方法的流程图，该通过该至少两个声音采集单元对该终端周边的声源进行定位的步骤可以如下：
在步骤202a中，获取一声源发出的声音信号达到该至少两个声音采集单元的时间差和相位差。
其中，终端中包含的至少两个声音采集单元组成一个声音采集单元阵列，由于声源到该声音采集单元阵列中的各个声音采集单元的距离和方向都不相同，因此，该声源发出的声音传递到各个声音采集单元的时间和相位也不相同。终端可以获取该声音采集单元阵列中的各个声音采集单元采集到同一声源发出的声音的时间差和相位差。
在步骤202b中，根据该声源发出的声音信号达到该至少两个声音采集单元的时间差和相位差确定该声源相对于该终端的空间位置。
终端可以利用波束成形技术定位声源的位置，比如，终端可以根据该声源发出的声音 信号达到该至少两个声音采集单元的时间差和相位差，通过延迟求和或者特征值分解等算法识别出声源相对于终端的位置。
利用波束成形技术，终端不仅可以准确的定位固定声源的空间位置，还可以对移动中的声源进行追踪，以便对移动中的声源发出的信号持续进行优化处理。
在步骤204中，根据该各个声源的声音信号的强度和/或声音信号的连续性，从该各个声源中筛选出指定声源。
终端在通话或者录音状态下时，通常只有用户说话时发出的声音信号才是有效的声音信号，其它的声音信号大多为噪声，在本公开实施例所示的方案中，终端可以根据各个声源的声音信号的强度和/或声音信号的连续性，从各个声源中筛选出最有可能是用户的声源，后续只对筛选出的最有可能是用户的声源进行优化。
比如，终端可以为声音信号的不同强度设置不同的权重，并为声音信号的不同连续时间也设置不同权重，终端接收到一个声源发出的声音信号后，分别确定其声音强度对应的权重和持续时间对应的权重进行加权，并对各个声源按照对应的加权进行排序，确定出加权最高的指定声源。
可选的，在为声音信号的不同强度设置不同的权重时，可以将最接近人说话声音的强度设置较高的权重，与人说话声音的强度之间的差值越大，对应设置的权重越低。比如，人平时说话正常的声音强度大约为40分贝，则可以对40分贝设置最高的加权，距离40分贝越远的声音强度，其权重越低。
同样的，在为声音信号的持续时间设置不同权重时，可以将最接近人说话时的声音信号的持续时间设置较高的权重，与人说话时的声音信号的持续时间之间的差值越大，对应设置的权重也越低。比如，人说话时，正常一句话的持续时间可能是2s，则可以对2s的持续时间设置最高的加权，与2s之间差距越大的持续时间，其权重越低。
由于录音或者通话时，同一时间通常只有一个或者两个用户在说话，为了避免对过多的声源进行不必要的优化处理，降低优化过程的复杂度，可以设置同时筛选出的指定声源的数量，比如，最多同一时间筛选出3个指定声源，后续同一时间最多只对3个声源发出的声音信号进行优化。
在步骤206中，根据定位获得的该指定声源的空间位置，确定该终端采集到的声音信号中由该指定声源发出的声音信号。
终端对接收到的声音信号进行处理时，可以根据指定声源的空间位置，只对该指定声源发出的声音信号进行处理。
在步骤208中，将指定声源发出的声音信号的强度与预设的第一强度阈值和第二强度阈值进行比较，该第一强度阈值大于或者等于该第二强度阈值。
在步骤210中，若该指定声源发出的声音信号的强度大于该第一强度阈值，则将该指定声源发出的声音信号的强度降低至该第一强度阈值。
在步骤212中，若该指定声源发出的声音信号的强度小于该第二强度阈值，则将该指 定声源发出的声音信号的强度增加至该第二强度阈值。
可选的，若该指定声源发出的声音信号的强度不小于该第一强度阈值，且不大于该第二强度阈值，则不对指定声源发出的声音信号的强度进行处理。
终端可以按照人正常说话时的声音信号强度设置第一强度阈值和第二强度阈值，若指定声源的声音信号强度较大，则将该指定声源的声音信号的强度降低，若指定声源的声音信号强度较小，则将该指定声源的声音信号的强度提高，若指定声源的声音信号强度适中，则不对指定声源的声音信号的强度进行处理。从而使可能是用户的指定声源发出的声音信号的强度保持在一个预定的范围内，既不会太高，也不会太低，从而使录音或者通话的效果达到最佳。
比如，以用户A和用户B通过智能手机进行远程电话会议，或者，通过手机进行会议录音为例，控制智能手机进入录音状态或者通话状态时，将智能手机切换至会议录音模式，此时，智能手机通过至少两个麦克风采集周围的声音并确定各个声源的位置，当用户A或者用户B发言时，智能手机将发言者筛选为指定声源，并对该发言者说话时发出的声音信号进行优化，当该发言者说话时发出的声音信号强度小于30分贝时，将该发言者说话时发出的声音信号的强度提升为30分贝，当该发言者说话时发出的声音信号大于50分贝时，将该发言者说话时发出的声音信号降低至50分贝，对于30分贝至50分贝之间的声音信号，则不做任何处理。从而保证与智能手机之间的距离不同的各个用户，其录制或者被远端收听到的说话音量都处于一个适当的范围内，从而保证了录音效果或者电话会议的通话效果。
综上所述，本公开实施例中提供的声音信号优化方法，通过至少两个声音采集单元对终端周边的声源进行定位，从定位确定的各个声源中筛选出指定声源，根据定位获得的该指定声源的空间位置，确定该终端采集到的声音信号中由该指定声源发出的声音信号，并将该指定声源发出的声音信号的强度调整至预先设置的两个强度阈值之间，将终端采集到的指定声源发出的声音信号的强度调整到一个合适的范围之内，从提高终端在录音或者通话时，对声音信号的采集效果。
作为另一种可选的实施例，在执行上述图2所示实施例中的步骤206之前，还可以执行如图4所示的增益补偿方法的流程图中的步骤205a～步骤205c。
在步骤205a中，通过该终端中的姿态传感器获取该终端的姿态，该终端的姿态包括该终端正面与水平面之间的夹角。
正常情况下，在进行录音或者电话会议时，终端正面向上水平放置在桌面上，而在某一情况下，终端的姿态可能并不是水平的，比如，当终端搭置在书本或者其它物体上时，其正面并不是水平状态，此时，终端正面与水平面之间会呈一定的夹角，导致终端采集到的声音信号与理想状态之间有一定的偏差，与终端正面处于水平状态时采集到的声音信号相比偏小。
对此，终端从采集到的声音信号中确定出指定声源发出的声音信号之前并做优化处理之前，首先可以根据当前姿态对采集到的声音信号做一些补偿。
在步骤205b中，根据该终端正面与水平面之间的夹角确定补偿数值，该终端正面与水平面之间的夹角与该补偿数值之间成正比例关系。
当终端正面与水平面之间的夹角越大时，终端采集到的声音信号与理想状态(即终端正面处于水平状态)之间的偏差也越大，因此，终端可以根据终端正面与水平面之间的夹角，通过预先设置算法计算补偿数值，或者通过预先存储的对应关系查询需要补偿的数值。
在步骤205c中，根据该补偿数值对该终端采集到的声音信号进行增益补偿。
终端确定出补偿数值后，对采集到的所有声音信号进行增益补偿。
其中，终端在确定该终端采集到的声音信号中由该指定声源发出的声音信号时，可以确定经过增益补偿后的声音信号中，由该指定声源发出的声音信号。
综上所述，本公开实施例中提供的方法，通过在从采集到的声音信号中确定出指定声源的声音信号并做优化之前，获取终端的姿态，根据终端的姿态对终端采集到的声音信号进行增益补偿，进一步提高了终端在录音或者通话时，对声音信号的采集效果。
下述为本公开装置实施例，可以用于执行本公开方法实施例。对于本公开装置实施例中未披露的细节，请参照本公开方法实施例。
图5是根据一示例性实施例示出的一种声音信号优化装置的框图，该声音信号优化装置可以用于包含有至少两个声音采集单元的终端中，该终端可以是智能手机、平板电脑、电子书阅读器以及智能录音笔等支持语音通话或者录音功能的智能终端，执行如图1至4任一所示的方法。如图5所示，该声音信号优化装置包括但不限于：定位模块501、筛选模块502、声音信号确定模块503以及优化模块504；
所述定位模块501被设置为用于通过所述至少两个声音采集单元对所述终端周边的声源进行定位；
所述筛选模块502被设置为用于从定位确定的各个声源中筛选出指定声源；
所述声音信号确定模块503被设置为用于根据定位获得的所述指定声源的空间位置，确定所述终端采集到的声音信号中由所述指定声源发出的声音信号；
所述优化模块504被设置为用于对所述指定声源发出的声音信号进行音量优化。
综上所述，本公开实施例中提供的声音信号优化装置，通过至少两个声音采集单元对终端周边的声源进行定位，从定位确定的各个声源中筛选出指定声源，根据定位获得的该指定声源的空间位置，确定该终端采集到的声音信号中由该指定声源发出的声音信号，对该指定声源发出的声音信号进行音量优化；当终端在采集声音信号时，对周边声源进行筛选，并对筛选出的声源发出的声音信号进行优化，从提高终端对声音信号的采集效果。
图6是根据一示例性实施例示出的一种声音信号优化装置的框图，该声音信号优化装 置可以用于包含有至少两个声音采集单元的终端中，该终端可以是智能手机、平板电脑、电子书阅读器以及智能录音笔等支持语音通话或者录音功能的智能终端，执行如图1至4任一所示的方法。如图6所示，该声音信号优化装置包括但不限于：定位模块501、筛选模块502、声音信号确定模块503以及优化模块504；
所述定位模块501被设置为用于通过所述至少两个声音采集单元对所述终端周边的声源进行定位；
所述筛选模块502被设置为用于从定位确定的各个声源中筛选出指定声源；
所述声音信号确定模块503被设置为用于根据定位获得的所述指定声源的空间位置，确定所述终端采集到的声音信号中由所述指定声源发出的声音信号；
所述优化模块504被设置为用于对所述指定声源发出的声音信号进行音量优化。
可选的，所述优化模块504，包括：比较子模块504a、降低子模块504b以及增加子模块504c；
所述比较子模块504a被设置为用于将所述指定声源发出的声音信号的强度与预设的第一强度阈值和第二强度阈值进行比较，所述第一强度阈值大于或者等于所述第二强度阈值；
所述降低子模块504b被设置为用于若所述指定声源发出的声音信号的强度大于所述第一强度阈值，则将所述指定声源发出的声音信号的强度降低至所述第一强度阈值；
所述增加子模块504c被设置为用于若所述指定声源发出的声音信号的强度小于所述第二强度阈值，则将所述指定声源发出的声音信号的强度增加至所述第二强度阈值。
可选的，所述定位模块501，包括：获取子模块501a以及定位子模块501b；
所述获取子模块501a被设置为用于获取一声源发出的声音信号达到所述至少两个声音采集单元的时间差和相位差；
所述定位子模块501b被设置为用于根据所述声源发出的声音信号达到所述至少两个声音采集单元的时间差和相位差确定所述声源相对于所述终端的空间位置。
可选的，所述筛选模块502被设置为用于根据所述各个声源的声音信号的强度和/或声音信号的连续性，从所述各个声源中筛选出所述指定声源。
可选的，所述装置还包括：姿态获取模块505和补偿模块506；
所述姿态获取模块505被设置为用于在所述声音信号确定模块503确定所述终端采集到的声音信号中由所述指定声源发出的声音信号之前，通过所述终端中的姿态传感器获取所述终端的姿态；
所述补偿模块506被设置为用于根据所述终端的姿态对所述终端采集到的声音信号进行增益补偿；
所述声音信号确定模块503被设置为用于确定经过增益补偿后的声音信号中，由所述指定声源发出的声音信号。
可选的，所述补偿模块506，包括：数值确定子模块506a以及补偿子模块506b；
所述数值确定子模块506a被设置为用于根据所述终端正面与水平面之间的夹角确定补偿数值，所述终端正面与水平面之间的夹角与所述补偿数值之间成正比例关系；
所述补偿子模块506b被设置为用于根据所述补偿数值对所述终端采集到的声音信号进行增益补偿；
其中，所述终端的姿态包括所述终端正面与水平面之间的夹角。
综上所述，本公开实施例中提供的声音信号优化装置，通过至少两个声音采集单元对终端周边的声源进行定位，从定位确定的各个声源中筛选出指定声源，根据定位获得的该指定声源的空间位置，确定该终端采集到的声音信号中由该指定声源发出的声音信号，并将该指定声源发出的声音信号的强度调整至预先设置的两个强度阈值之间，将终端采集到的指定声源发出的声音信号的强度调整到一个合适的范围之内，从提高终端在录音或者通话时，对声音信号的采集效果。
此外，本公开实施例中提供的装置，通过在从采集到的声音信号中确定出指定声源的声音信号并做优化之前，获取终端的姿态，根据终端的姿态对终端采集到的声音信号进行增益补偿，进一步提高了终端在录音或者通话时，对声音信号的采集效果。
图7是根据一示例性实施例示出的一种装置700的框图。其中，该装置中包含至少两个声音采集单元。例如，装置700可以是移动电话，计算机，数字广播终端，消息收发设备，路由设备，游戏控制台，平板设备，医疗设备，健身设备，个人数字助理等。
参照图7，装置700可以包括以下一个或多个组件：处理组件702，存储器704，电源组件706，多媒体组件708，音频组件710，输入/输出(I/O)的接口712，传感器组件714，以及通信组件716。
处理组件702通常控制装置700的整体操作，诸如与显示，电话呼叫，数据通信，相机操作和记录操作相关联的操作。处理组件702可以包括一个或多个处理器718来执行指令，以完成上述的方法的全部或部分步骤。此外，处理组件702可以包括一个或多个模块，便于处理组件702和其他组件之间的交互。例如，处理组件702可以包括多媒体模块，以方便多媒体组件708和处理组件702之间的交互。
存储器704被配置为存储各种类型的数据以支持在装置700的操作。这些数据的示例包括用于在装置700上操作的任何应用程序或方法的指令，联系人数据，电话簿数据，消息，图片，视频等。存储器704可以由任何类型的易失性或非易失性存储设备或者它们的组合实现，如静态随机存取存储器(SRAM)，电可擦除可编程只读存储器(EEPROM)，可擦除可编程只读存储器(EPROM)，可编程只读存储器(PROM)，只读存储器(ROM)，磁存储器，快闪存储器，磁盘或光盘。存储器704中还存储有一个或多个模块，该一个或多个模块被配置成由该一个或多个处理器720执行，以完成上述图1至4任一所示的方法的全部或部分步骤。
电源组件706为装置700的各种组件提供电力。电源组件706可以包括电源管理系统， 一个或多个电源，及其他与为装置700生成、管理和分配电力相关联的组件。
多媒体组件708包括在所述装置700和用户之间的提供一个输出接口的屏幕。在一些实施例中，屏幕可以包括液晶显示器(LCD)和触摸面板(TP)。如果屏幕包括触摸面板，屏幕可以被实现为触摸屏，以接收来自用户的输入信号。触摸面板包括一个或多个触摸传感器以感测触摸、滑动和触摸面板上的手势。所述触摸传感器可以不仅感测触摸或滑动动作的边界，而且还检测与所述触摸或滑动操作相关的持续时间和压力。在一些实施例中，多媒体组件708包括一个前置摄像头和/或后置摄像头。当装置700处于操作模式，如拍摄模式或视频模式时，前置摄像头和/或后置摄像头可以接收外部的多媒体数据。每个前置摄像头和后置摄像头可以是一个固定的光学透镜系统或具有焦距和光学变焦能力。
音频组件710被配置为输出和/或输入音频信号。例如，音频组件710包括一个麦克风(MIC)，当装置700处于操作模式，如呼叫模式、记录模式和语音识别模式时，麦克风被配置为接收外部音频信号。所接收的音频信号可以被进一步存储在存储器704或经由通信组件716发送。在一些实施例中，音频组件710还包括一个扬声器，用于输出音频信号。
I/O接口712为处理组件702和外围接口模块之间提供接口，上述外围接口模块可以是键盘，点击轮，按钮等。这些按钮可包括但不限于：主页按钮、音量按钮、启动按钮和锁定按钮。
传感器组件714包括一个或多个传感器，用于为装置700提供各个方面的状态评估。例如，传感器组件714可以检测到装置700的打开/关闭状态，组件的相对定位，例如所述组件为装置700的显示器和小键盘，传感器组件714还可以检测装置700或装置700一个组件的位置改变，用户与装置700接触的存在或不存在，装置700方位或加速/减速和装置700的温度变化。传感器组件714可以包括接近传感器，被配置用来在没有任何的物理接触时检测附近物体的存在。传感器组件714还可以包括光传感器，如CMOS或CCD图像传感器，用于在成像应用中使用。在一些实施例中，该传感器组件714还可以包括加速度传感器，陀螺仪传感器，磁传感器，压力传感器或温度传感器。
通信组件716被配置为便于装置700和其他设备之间有线或无线方式的通信。装置700可以接入基于通信标准的无线网络，如WiFi，2G或3G，或它们的组合。在一个示例性实施例中，通信组件716经由广播信道接收来自外部广播管理系统的广播信号或广播相关信息。在一个示例性实施例中，所述通信组件716还包括近场通信(NFC)模块，以促进短程通信。例如，在NFC模块可基于射频识别(RFID)技术，红外数据协会(IrDA)技术，超宽带(UWB)技术，蓝牙(BT)技术和其他技术来实现。
在示例性实施例中，装置700可以被一个或多个应用专用集成电路(ASIC)、数字信号处理器(DSP)、数字信号处理设备(DSPD)、可编程逻辑器件(PLD)、现场可编程门阵列(FPGA)、控制器、微控制器、微处理器或其他电子元件实现，用于执行上述方法。
在示例性实施例中，还提供了一种包括指令的非临时性计算机可读存储介质，例如包括指令的存储器704，上述指令可由装置700的处理器718执行以完成上述方法。例如，所述非临时性计算机可读存储介质可以是ROM、随机存取存储器(RAM)、CD-ROM、磁带、软盘和光数据存储设备等。
关于上述实施例中的装置，其中各个模块执行操作的具体方式已经在有关该方法的实施例中执行了详细描述，此处将不做详细阐述说明。
应当理解的是，本发明并不局限于上面已经描述并在附图中示出的精确结构，并且可以在不脱离其范围执行各种修改和改变。本发明的范围仅由所附的权利要求来限制。

Claims (13)

  1. 一种声音信号优化方法，其特征在于，用于包含有至少两个声音采集单元的终端中，所述方法包括：
    通过所述至少两个声音采集单元对所述终端周边的声源进行定位；
    从定位确定的各个声源中筛选出指定声源；
    根据定位获得的所述指定声源的空间位置，确定所述终端采集到的声音信号中由所述指定声源发出的声音信号；
    对所述指定声源发出的声音信号进行音量优化。
  2. 根据权利要求1所述的方法，其特征在于，所述对所述指定声源发出的声音信号进行音量优化，包括：
    将所述指定声源发出的声音信号的强度与预设的第一强度阈值和第二强度阈值进行比较，所述第一强度阈值大于或者等于所述第二强度阈值；
    若所述指定声源发出的声音信号的强度大于所述第一强度阈值，则将所述指定声源发出的声音信号的强度降低至所述第一强度阈值；
    若所述指定声源发出的声音信号的强度小于所述第二强度阈值，则将所述指定声源发出的声音信号的强度增加至所述第二强度阈值。
  3. 根据权利要求1所述的方法，其特征在于，所述通过所述至少两个声音采集单元对所述终端周边的声源进行定位，包括：
    获取一声源发出的声音信号达到所述至少两个声音采集单元的时间差和相位差；
    根据所述声源发出的声音信号达到所述至少两个声音采集单元的时间差和相位差确定所述声源相对于所述终端的空间位置。
  4. 根据权利要求1所述的方法，其特征在于，所述从定位确定的声源中筛选出指定声源，包括：
    根据所述各个声源的声音信号的强度和/或声音信号的连续性，从所述各个声源中筛选出所述指定声源。
  5. 根据权利要求1所述的方法，其特征在于，所述确定所述终端采集到的声音信号中由所述指定声源发出的声音信号之前，还包括：
    通过所述终端中的姿态传感器获取所述终端的姿态；
    根据所述终端的姿态对所述终端采集到的声音信号进行增益补偿；
    所述确定所述终端采集到的声音信号中由所述指定声源发出的声音信号，包括：
    确定经过增益补偿后的声音信号中，由所述指定声源发出的声音信号。
  6. 根据权利要求5所述的方法，其特征在于，所述终端的姿态包括所述终端正面与水平面之间的夹角，所述根据所述终端的姿态对所述终端采集到的声音信号进行增益补偿，包括：
    根据所述终端正面与水平面之间的夹角确定补偿数值，所述终端正面与水平面之间的夹角与所述补偿数值之间成正比例关系；
    根据所述补偿数值对所述终端采集到的声音信号进行增益补偿。
  7. 一种声音信号优化装置，其特征在于，用于包含有至少两个声音采集单元的终端中，所述装置包括：
    定位模块，用于通过所述至少两个声音采集单元对所述终端周边的声源进行定位；
    筛选模块，用于从定位确定的各个声源中筛选出指定声源；
    声音信号确定模块，用于根据定位获得的所述指定声源的空间位置，确定所述终端采集到的声音信号中由所述指定声源发出的声音信号；
    优化模块，用于对所述指定声源发出的声音信号进行音量优化。
  8. 根据权利要求7所述的装置，其特征在于，所述优化模块，包括：
    比较子模块，用于将所述指定声源发出的声音信号的强度与预设的第一强度阈值和第二强度阈值进行比较，所述第一强度阈值大于或者等于所述第二强度阈值；
    降低子模块，用于若所述指定声源发出的声音信号的强度大于所述第一强度阈值，则将所述指定声源发出的声音信号的强度降低至所述第一强度阈值；
    增加子模块，用于若所述指定声源发出的声音信号的强度小于所述第二强度阈值，则将所述指定声源发出的声音信号的强度增加至所述第二强度阈值。
  9. 根据权利要求7所述的装置，其特征在于，所述定位模块，包括：
    获取子模块，用于获取一声源发出的声音信号达到所述至少两个声音采集单元的时间差和相位差；
    定位子模块，用于根据所述声源发出的声音信号达到所述至少两个声音采集单元的时间差和相位差确定所述声源相对于所述终端的空间位置。
  10. 根据权利要求7所述的装置，其特征在于，所述筛选模块，用于根据所述各个声源的声音信号的强度和/或声音信号的连续性，从所述各个声源中筛选出所述指定声源。
  11. 根据权利要求7所述的装置，其特征在于，所述装置还包括：姿态获取模块和补偿模块；
    所述姿态获取模块，用于在所述声音信号确定模块确定所述终端采集到的声音信号中由所述指定声源发出的声音信号之前，通过所述终端中的姿态传感器获取所述终端的姿态；
    所述补偿模块，用于根据所述终端的姿态对所述终端采集到的声音信号进行增益补偿；
    所述声音信号确定模块，用于确定经过增益补偿后的声音信号中，由所述指定声源发出的声音信号。
  12. 根据权利要求11所述的装置，其特征在于，所述补偿模块，包括：
    数值确定子模块，用于根据所述终端正面与水平面之间的夹角确定补偿数值，所述终端正面与水平面之间的夹角与所述补偿数值之间成正比例关系；
    补偿子模块，用于根据所述补偿数值对所述终端采集到的声音信号进行增益补偿；
    其中，所述终端的姿态包括所述终端正面与水平面之间的夹角。
  13. 一种声音信号优化方法装置，其特征在于，用于包含有至少两个声音采集单元的终端中，所述装置包括：
    处理器；
    用于存储处理器可执行指令的存储器；
    其中，所述处理器被配置为：
    通过所述至少两个声音采集单元对所述终端周边的声源进行定位；
    从定位确定的各个声源中筛选出指定声源；
    根据定位获得的所述指定声源的空间位置，确定所述终端采集到的声音信号中由所述指定声源发出的声音信号；
    对由所述指定声源发出的声音信号进行音量优化。
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