WO2019166030A1 - 一种智能声场适配系统、智能声场校准系统及智能语音处理系统 - Google Patents

一种智能声场适配系统、智能声场校准系统及智能语音处理系统 Download PDF

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Publication number
WO2019166030A1
WO2019166030A1 PCT/CN2019/082366 CN2019082366W WO2019166030A1 WO 2019166030 A1 WO2019166030 A1 WO 2019166030A1 CN 2019082366 W CN2019082366 W CN 2019082366W WO 2019166030 A1 WO2019166030 A1 WO 2019166030A1
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Prior art keywords
speaker
unit
audio
sound field
information
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PCT/CN2019/082366
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English (en)
French (fr)
Inventor
古曦
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成都星环科技有限公司
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Priority claimed from CN201810165174.1A external-priority patent/CN108513243B/zh
Priority claimed from CN201810165962.0A external-priority patent/CN108495238A/zh
Priority claimed from CN201810165964.XA external-priority patent/CN108513192A/zh
Application filed by 成都星环科技有限公司 filed Critical 成都星环科技有限公司
Publication of WO2019166030A1 publication Critical patent/WO2019166030A1/zh

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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
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/02—Casings; Cabinets ; Supports therefor; Mountings therein
    • H04R1/04—Structural association of microphone with electric circuitry therefor
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R29/00—Monitoring arrangements; Testing arrangements
    • 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
    • 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
    • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response

Definitions

  • the present disclosure relates to the field of audio adjustment, and in particular, to an intelligent sound field adaptation system, an intelligent sound field calibration system, and an intelligent voice processing system.
  • the conventional sound playback device mainly includes a speaker and a speaker adapting device, and the speaker is connected with the speaker adapting device, and the source data or signal input to the speaker adapting device is digitally processed to obtain a digital audio signal, and finally the digital audio signal is converted into an analog signal.
  • the audio signal is output to the speaker to convert the electrical signal to the sound signal.
  • Important indicators for measuring the sound effects of sound playback devices include stereoscopic, positioning, spatial, distortion, subwoofer, and treble. Stereo refers to a sound with a stereoscopic effect. If the entire system can restore the spatial sense of the original sound to some extent from recording to playback, then the playback sound with a certain degree of spatial distribution characteristics such as azimuth level is called For stereo in audio technology.
  • a stereo sound playback device can improve the clarity and intelligibility of the information and enhance the listener's presence.
  • the power amplifier and speaker mating elements are: 1. Power matching, 2. Impedance matching, 3. Damping coefficient matching, 4. Sensitivity matching, 5. Sound matching. If we recognize the above five points when mating, the performance of the equipment used can be maximized and fully utilized.
  • the existing speaker playback system once installed, the entire sound field system is determined, and the sound field can not be adapted according to the number of listeners, the position of the listener in the sound field, and the type of audio information, thereby reducing the sound of the speaker system.
  • the performance tension of information has caused a waste of playing resources. Therefore, there is a need for a personalized and intelligent sound field adaptation system for real-time sound field adjustment.
  • the present disclosure provides an intelligent sound field adaptation system
  • the sound field adaptation system includes at least a self-test portion, an audio information collection portion, a data processing portion, and at least one speaker; the data processing portion collects the number and position of the speakers based on the self-test portion.
  • the information is completed to allocate the audio information frequency band collected by the audio information collecting unit, and based on the allocated frequency band information of each speaker and the positional relationship of the listener, the angle of at least one speaker is adjusted, and the number of the speaker and the position parameter and the speaker band allocation parameter are adjusted.
  • the audience position parameter and the speaker angle adjustment parameter are stored in the data processing unit as a sound field adaptation model.
  • the self-testing portion is configured to perform labeling and counting of at least one speaker on the name and position based on the collected at least one speaker identification information and environment information, and perform equivalent grouping on the at least one speaker. Get at least one equivalent group.
  • the self-test portion is specifically configured to divide at least one speaker in the same position in the sound field or in a symmetrical position on the left and right sides of the listener into an equivalent group.
  • the audio classification unit in the data processing unit is configured to perform frequency band allocation based on the number of equivalent groups counted by the self-test unit.
  • the audio classification unit in the data processing unit is configured to: according to the number of equivalent groups divided by the self-test unit and the positional relationship between the speaker and the listener in each equivalent group, The audio information collected by the audio information collecting unit is allocated to the frequency band to obtain at least one frequency band; for each frequency band obtained, the target playing position of the audio information corresponding to the frequency band is separately determined, and the frequency band is allocated to the target The speaker in the playback position.
  • the speaker adjusting unit in the data processing unit is configured to calculate each of the equivalent groups based on the positional relationship between the speaker devices of the equivalent group and the listener, and the frequency band information allocated by the speaker devices of each equivalent group. The angle of the speaker device is adjusted.
  • the audio information collecting unit includes a microphone and an audio acquirer; and the speaker adjusting unit in the data processing unit is configured to:
  • the audio information collecting unit includes a timbre adjusting unit
  • the microphone is configured to collect raw vocal sound data
  • the audio acquirer is configured to collect background music data
  • the timbre adjusting unit is configured to perform real-time adjustment on the original vocal sound data based on the background music data and a preset tuning function, and mix the adjusted original vocal sound data and the background music data into a mixed data. And transmitting the mixed data to the data processing unit for processing.
  • the data processing unit is further provided with at least a DSP processor, and the DSP processor is configured to add the audio data collected by the audio information collecting unit based on a frequency band allocation result of the audio classification unit.
  • the band mark and the position information mark are sent, and the marked audio data is sent to the speaker device corresponding to the added position information mark to complete the playing of the audio information of the corresponding frequency band.
  • the data processing unit is further provided with at least a storage unit configured to match a sound field including a number of speakers, a position parameter, a speaker band allocation parameter, a listener position parameter, and a speaker angle adjustment parameter.
  • the data of the model is collected and recorded.
  • the data processing unit is further provided with at least a data triggering unit, and the data triggering unit is configured to trigger the said method according to the speaker number parameter, the speaker position parameter, and the audience position parameter collected by the self-checking unit.
  • At least one sound field adaptation model stored in the storage unit is rapidly adjusted according to the triggered sound field adaptation model.
  • the environmental information includes position information of each speaker, and audience position information, and length, width, and height information of a space in which the speaker is located.
  • the environment information is uploaded to the self-testing unit through the smart terminal.
  • the present disclosure also provides an intelligent sound field calibration system, the sound field calibration system at least comprising a self-testing portion, an audio information collecting portion, a data processing portion and at least one sound box, wherein the audio information collecting portion is provided with at least an audio acquirer and a microphone ;
  • the data processing unit completes the frequency band allocation of the audio information data collected by the audio acquirer based on the number of the speaker and the position information collected by the self-test unit, and completes the angle of the at least one speaker based on the allocated frequency band information of each speaker and the positional relationship of the listener. Preliminary adjustment
  • the data processing unit completes the secondary adjustment of the angle of the at least one sound box based on the same sound information data, the frequency band allocation data, and the positional relationship of the listener acquired by the microphone, which are the same as the audio information data acquired by the audio acquirer;
  • the data processing unit stores the speaker number and position parameters, the speaker band allocation parameter, the audience position parameter, and the speaker angle adjustment parameter as a sound field adaptation model in the data processing unit.
  • the present disclosure further provides an intelligent voice processing system, where the voice processing system includes at least a self-testing unit, an audio information collecting unit, a data processing unit, and at least one sound box; at least an audio acquirer and a microphone are disposed in the audio information collecting unit. And a tone color adjustment unit, the tone color adjustment unit performs real-time adjustment of the sound information collected by the microphone based on the preset tuning function and the original sound frequency in the audio information acquired by the audio acquirer, and the sound collected by the adjusted microphone
  • the audio data collected by the data and audio acquirer is sent to the data processing unit, and the data processing unit completes the frequency band allocation of the audio information collected by the audio information collecting unit based on the number of the speakers and the position information collected by the self-checking unit.
  • FIG. 1 is a schematic diagram of functional modules of an intelligent sound field adaptation system provided by the present disclosure
  • FIG. 2 is a schematic diagram of functional modules of an audio information collecting unit according to the present disclosure
  • FIG. 3 is a schematic diagram of functional modules of a data processing unit according to the present disclosure.
  • the intelligent sound field adaptation system includes at least a self-testing unit, an audio information collecting unit, a data processing unit, and at least one speaker.
  • the self-testing unit may be a data collector configured to collect and classify speaker information and collect environmental information.
  • the audio information collecting unit is configured to implement audio information acquisition, for example, completing audio information by the network next week or directly acquiring specific audio information from the hard disk.
  • FIG. 2 is a functional block diagram of an audio information collecting unit according to an embodiment of the present invention.
  • the audio information collecting section may include a microphone and an audio acquirer.
  • the microphone is configured to collect audio information, such as original vocal data
  • the audio acquirer is configured to collect audio information, such as background music data.
  • the audio information collecting unit may further include a tone color adjusting unit.
  • the timbre adjusting unit is configured to perform real-time adjustment on the original vocal sound data based on an original sound frequency in the background music data and a preset tuning function, and mix the adjusted original vocal sound data with the background music data. Forming a mix data, and transmitting the mix data to the data processing unit for subordinate data processing.
  • the audio information collected by the audio information collecting unit mentioned later may refer to the mixed data processed by the timbre adjusting unit, or may be processed by the timbre adjusting unit.
  • the audio information is not limited by this embodiment.
  • the sound adjustment unit realizes real-time adjustment of the sound information collected by the microphone, so that the frequency of the sound collected by the microphone tends to be the original sound data, so that the sound is more pleasant and more suitable for the user when singing through the microphone, thereby further improving the sound.
  • the user's use of the speaker playback system is friendly.
  • the self-testing portion is configured to perform labeling and counting of the name and location of the at least one sound box based on the collected at least one speaker identification information and environmental information, and perform equivalent grouping of the at least one sound box to obtain at least one equivalent group (Also called "sound effect group").
  • the speaker is equivalently grouped into at least one speaker that is in the same position in the sound field or symmetrically positioned on the left and right sides of the listener, and is equally divided into the same sound effect group.
  • the environmental information may include length, width, and height information of a space in which the speaker is located.
  • the environmental information may also include location information of each speaker and audience location information.
  • the environment information may be uploaded by the user to the self-testing department via the smart terminal, or may be automatically acquired by the corresponding environment information collecting device.
  • FIG. 3 is a functional block diagram of a data processing unit according to the present disclosure.
  • the data processing unit at least includes an audio classification unit, a speaker adjustment unit, a DSP processor, a storage unit, and a data triggering unit, wherein the data processing unit is configured to process the audio information and send the processed audio information.
  • the data processing unit is further configured to perform frequency band allocation on the audio information collected by the audio information collecting unit based on the number of speakers and the position information collected by the self-checking unit.
  • the audio classification unit is configured to perform frequency band information allocation and play position allocation on the audio information collected by the audio information collection unit.
  • the audio classification unit is configured to complete the frequency band allocation based on the number of equivalent groups counted by the self-test unit.
  • the audio classification unit may be configured to: collect, according to the number of equivalent groups divided by the self-test unit and the positional relationship between the speaker and the listener in each equivalent group, the audio information collection unit collects The audio information is allocated to the frequency band to divide the audio information collected by the audio collection unit into at least one frequency band; for each frequency band obtained, the target playback position of the audio information corresponding to the frequency band is determined, and the frequency band is allocated to the The speaker of the target playback position.
  • the speaker adjustment unit is configured to perform angle adjustment on each of the speakers.
  • the speaker adjusting unit may be configured to adjust an angle of the at least one speaker based on the frequency band information allocated by each speaker and the positional relationship between each speaker and the listener.
  • the speaker adjusting unit is configured to adjust an angle of each equivalent group of speaker devices based on a positional relationship between each equivalent group of speaker devices and a listener and frequency band information assigned to each equivalent group of speaker devices.
  • the data processing unit may be specifically configured to perform angle adjustment by using the sound box adjusting unit.
  • the speaker adjustment unit may be configured to: perform frequency band allocation on the audio information collected by the audio acquirer based on the number of speakers and position information collected by the self-test unit, and based on the frequency band allocated by the at least one speaker Information and a positional relationship between the at least one speaker and the listener to initially adjust an angle of the at least one speaker; determining, from the audio information collected by the microphone, the same target information as the audio information collected by the audio acquirer, according to The target information, the result of the band allocation, and the positional relationship of the at least one speaker to the listener are adjusted twice for the angle of the at least one speaker.
  • the DSP processor is configured to distribute captured audio information into a particular speaker device.
  • the DSP processor is configured to add a frequency band mark and a position information mark to the audio data collected by the audio information collecting unit based on a frequency band allocation result of the audio classification unit.
  • the DSP processor labels the band allocation information and the location allocation information as the annotation data in the collected audio information.
  • the DSP processor sends the marked completed audio data to the box device corresponding to the sound of the position information (ie, the added position information indicated by the position information mark) to complete the playing of the audio information of the corresponding frequency band.
  • the at least one speaker device receives the audio data sent by the DSP processor, and plays the audio data of the corresponding frequency band according to the frequency band labeling information (ie, the foregoing frequency band mark) in the received audio data.
  • the frequency band labeling information ie, the foregoing frequency band mark
  • the storage unit in the data processing unit is configured to collect and record data of a sound field adaptation model including a number of speakers, a position parameter, a speaker band allocation parameter, a listener position parameter, and a speaker angle adjustment parameter.
  • a sound field adaptation model including a number of speakers, a position parameter, a speaker band allocation parameter, a listener position parameter, and a speaker angle adjustment parameter.
  • the data triggering unit in the data processing unit is configured to trigger at least one sound field adaptation model recorded in the storage unit based on a speaker number parameter, a speaker position parameter, and a listener position parameter collected by the self-test unit in real time, and according to the The triggered sound field adaptation model is used to quickly adjust the angle of the speakers in the sound field. In this way, when the speaker number parameter, the speaker position parameter, and the audience position parameter are changed, the entire speaker playing system can quickly complete the real-time and intelligent adjustment of the speaker playing system based on the stored sound field matching model, thereby bringing better to the user. Hearing experience.
  • the sound field calibration system includes at least a self-testing portion, an audio information collecting portion, a data processing portion, and at least one sound box.
  • the self-testing unit is a data collector configured to implement collection and classification of speaker information and collection of environmental information.
  • the audio information collecting unit is configured to implement audio information acquisition, such as downloading audio information through a network or directly acquiring specific audio information from a hard disk.
  • the self-checking unit performs labeling and counting of the name and position of the at least one speaker based on the collected at least one speaker identification information and the environment information, and performs equivalent grouping.
  • the speaker is equivalently grouped into at least one speaker that is in the same position in the sound field or symmetrically positioned on the left and right sides of the listener, and is equally divided into the same sound effect group.
  • the environmental information is length, width and height information of a space in which the speaker is located.
  • the environmental information also includes location information of each speaker and audience location information.
  • the environment information may be uploaded by the user to the self-testing department via the smart terminal.
  • the audio information collecting unit is provided with at least an audio acquirer and a microphone, which are respectively configured to complete collection of background music data and original vocal sound data.
  • the data processing unit includes at least an audio classification unit, a speaker adjustment unit, a DSP processor, a storage unit, and a trigger unit, where the data processing unit is configured to complete audio information processing and send the corresponding information.
  • Speaker equipment the data processing unit performs frequency band allocation on the audio information collected by the audio information collecting unit based on the number of the speaker and the position information collected by the self-checking unit.
  • the audio classification unit is configured to perform frequency band information distribution and play position allocation on the audio information collected by the audio information collection unit.
  • the audio classification unit completes the frequency band allocation based on the equivalent number of groups of the speaker devices counted by the self-test unit.
  • the speaker adjustment unit is configured to perform angle adjustment on each of the speakers.
  • the speaker adjusting unit completes the angle adjustment of each equivalent group of speaker devices based on the positional relationship between each equalizing group of speaker devices and the listener and the frequency band information allocated by each equivalent group of speaker devices.
  • the speaker adjusting unit completes the frequency band allocation of the audio information data collected by the audio acquirer based on the number of the speaker and the position information collected by the self-checking unit, and completes at least based on the allocated frequency band information of each speaker and the positional relationship of the listener.
  • the initial adjustment of the angle of a speaker is not limited to the frequency band allocation of the audio information data collected by the audio acquirer based on the number of the speaker and the position information collected by the self-checking unit, and completes at least based on the allocated frequency band information of each speaker and the positional relationship of the listener.
  • the speaker adjustment unit performs secondary adjustment of the angle of the at least one speaker based on the same sound information data, band allocation data, and positional relationship of the listener acquired by the microphone.
  • the DSP processor is configured to enable the collection of audio information to be distributed to a particular speaker device.
  • the DSP processor is configured to perform frequency band marking and location information marking on the audio data collected by the audio information collecting unit based on the frequency band allocation result of the audio classification unit.
  • the DSP processor labels the frequency band allocation information and the location allocation information as the annotation data in the collected audio information.
  • the DSP processor sends the marked audio data to the location information corresponding to the speaker device to play the audio information of the corresponding frequency band.
  • At least one speaker device completes receiving audio data sent by the DSP processor, and plays audio data of the corresponding frequency band according to the frequency band labeling information in the received audio data.
  • the storage unit in the data processing unit is configured to complete collecting and recording data of a sound field adaptation model including a number of speakers and a position parameter, a speaker band allocation parameter, a listener position parameter, and a speaker angle adjustment parameter, so as to be able to pass the storage
  • the unit stores at least one sound field adaptation model so that rapid triggering and invocation of the sound field adaptation model can be completed during subsequent use.
  • the data triggering unit in the data processing unit is configured to trigger at least one sound field adaptation model stored in the storage unit based on the speaker number parameter, the speaker position parameter, and the audience position parameter collected by the self-test unit in real time, and complete the speaker angle in the sound field. Adjustment. In this way, when the speaker number parameter, the speaker position parameter, and the audience position parameter are changed, the entire speaker playing system can complete the real-time and intelligent adjustment of the speaker playing system based on the stored sound field matching model, thereby providing a better listening experience for the user. .
  • the voice processing system includes at least a self-testing unit, an audio information collecting unit, a data processing unit, and at least one speaker.
  • the self-testing unit is a data collector configured to implement collection and classification of speaker information and collection of environmental information.
  • the audio information collecting unit is configured to implement audio information acquisition, such as downloading audio information through a network or directly acquiring specific audio information from a hard disk.
  • the self-checking unit performs labeling and counting of the name and position of the at least one speaker based on the collected at least one speaker identification information and the environment information, and performs equivalent grouping.
  • the speaker is equally grouped into at least one speaker that is in the same position in the sound field or in a symmetrical position on the left and right sides of the listener, and is equally divided into the same sound effect group (i.e., the aforementioned equivalent group).
  • the environmental information includes length, width, and height information of a space in which the speaker is located.
  • the environmental information may also include location information of each speaker and audience location information.
  • the environment information may be uploaded by the user to the self-testing department via the smart terminal, or may be acquired by using a preset environment information collecting device.
  • the audio information collecting unit is provided with at least an audio acquirer, a microphone, and a tone color adjusting unit configured to collect all sound information data of the entire voice processing system.
  • the timbre adjustment unit is configured to adjust the sound information collected by the microphone in real time based on a preset tuning function and an original sound frequency in the audio information acquired by the audio acquirer.
  • the timbre adjusting unit may form the sound data collected by the adjusted microphone (for example, the adjusted original vocal sound data) and the audio data collected by the audio acquirer (for example, the background sound data) to form a mixed data transmission.
  • the data processing unit performs subordinate data processing.
  • the sound color adjustment unit realizes real-time adjustment of the sound information collected by the microphone, so that the frequency data of the sound collected by the microphone tends to be the original sound data, so that when the user sings through the microphone, the sound is more pleasant and more suitable.
  • the tone further enhances the friendliness of the user using the speaker playback system.
  • the data processing unit includes at least an audio classification unit, a speaker adjustment unit, a DSP processor, a storage unit, and a trigger unit, where the data processing unit is configured to complete audio information processing and send the corresponding information.
  • Speaker equipment Preferably, the data processing unit completes the allocation of the audio information frequency band collected by the audio information collecting unit based on the number of the speaker and the position information collected by the self-checking unit.
  • the audio classification unit is configured to perform frequency band information distribution and play position allocation on the audio information collected by the audio information collection unit.
  • the audio classification unit performs frequency band allocation based on the number of equivalent groups of the speaker devices counted by the self-test department.
  • the speaker adjustment unit is configured to perform angle adjustment on each of the speakers.
  • the speaker adjustment unit adjusts an angle of the at least one speaker based on the allocated frequency band information of each speaker and the positional relationship of the listeners.
  • the speaker adjusting unit adjusts the angles of the speaker devices of the equivalent groups based on the positional relationship between the speaker devices of the respective equivalent groups and the position of the listener and the frequency band information allocated by the speaker devices of each equivalent group.
  • the DSP processor is configured to distribute captured audio information into a particular speaker device.
  • the DSP processor is configured to perform frequency band marking and location information marking on the audio data collected by the audio information collecting unit based on a frequency band allocation result of the audio classification unit.
  • the DSP processor labels the band allocation information and the location allocation information as the annotation data in the collected audio information.
  • the DSP processor sends the marked audio data to the position device corresponding to the speaker device to complete the playing of the corresponding frequency band audio information.
  • At least one speaker device completes receiving audio data sent by the DSP processor, and completes playing of the corresponding frequency band audio data according to the frequency band labeling information in the received audio data.
  • the storage unit in the data processing unit is configured to perform data acquisition and recording of a sound field adaptation model including a number of speakers and a position parameter, a speaker band allocation parameter, a listener position parameter, and a speaker angle adjustment parameter. Therefore, the at least one sound field adaptation model can be stored in the data processing unit through the storage unit, so that the fast triggering and invoking of the sound field adaptation model can be completed in the subsequent use.
  • the data triggering unit in the data processing unit is configured to trigger at least one sound field adaptation model stored in the storage unit based on the speaker number parameter, the speaker position parameter, and the audience position parameter collected by the self-test unit in real time, and complete the speaker angle in the sound field. Adjustment.
  • the entire speaker playing system can complete the real-time and intelligent adjustment of the speaker playing system based on the stored sound field matching model, thereby providing a better listening experience for the user.
  • the intelligent sound field adaptation system, the intelligent sound field calibration system and the intelligent voice processing system provided by the present disclosure can perform real-time sound field adaptation and sound field adjustment for a specific sound field environment, thereby satisfying the user's personalized demand for the sound field.

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Abstract

公开了一种智能声场适配系统、智能声场校准系统及智能语音处理系统，其中，该智能声场适配系统至少包括自检部、音频信息采集部、数据处理部和至少一个音箱；通过本公开的数据处理部完成音频信息处理并发送至对应的音箱设备，并通过数据处理部中数据触发单元实现与储存单元中储存的至少一个声场适配模型进行触发，完成声场内音箱角度的快速调整，从而实现了在音箱数量参数、音箱位置参数、听众位置参数发生改变时，整个音箱播放系统能够基于储存的声场适配模型完成音箱播放系统的实时和智能调节，为用户带来更好听觉体验。

Description

一种智能声场适配系统、智能声场校准系统及智能语音处理系统
相关申请的交叉引用
本申请要求于2018年02月28日提交中国专利局的申请号为201810165964.X、名称为“一种智能声场适配系统”的中国专利申请的优先权，于2018年02月28日提交中国专利局的申请号为201810165174.1、名称为“一种智能声场校准系统”的中国专利申请的优先权，以及于2018年02月28日提交中国专利局的申请号为201810165962.0、名称为“一种智能语音处理系统”的中国专利申请的优先权，其全部内容通过引用结合在本申请中。
技术领域
本公开涉及音频调节领域，尤其涉及一种智能声场适配系统、智能声场校准系统及智能语音处理系统。
背景技术
常规的声音播放装置主要包括音箱和音箱适配装置，音箱与音箱适配装置连接，输入音箱适配装置的源数据或者信号经过数字信号处理得到数字音频信号，最终该数字音频信号被转化为模拟音频信号并被输出至音箱，实现电信号到声音信号的转换。衡量声音播放装置的音响效果的重要指标包括立体感、定位感、空间感、失真度、重低音和高音等。其中立体声是指具有立体感的声音，如果从记录到重放，整个系统能够在一定程度上恢复原发声的空间感，那么这种具有一定程度的方位层次等空间分布特性的重放声，称为音响技术中的立体声。一个立体声好的声音播放装置可以提高信息的清晰度和可懂度，提高听者的临场感。
安装一套音响系统时，不免遇到功放与音箱的配接问题。在音色方面，会注意其搭配上是否冷暖相宜、软硬适中，最终使整套器材还原音色呈中性，这仅是从艺术方面考虑。从技术方面考虑功放与音箱配接的要素有：1.功率匹配，2.阻抗匹配，3.阻尼系数的匹配，4.灵敏度匹配，5.音色匹配。如果我们在配接时认识到上述五点，可使所用器材的性能得到较大、最充分的发挥。
但是，现有音箱播放系统，一旦安装后，整个声场系统即被确定了，不能根据听众人数、听众在声场中所处的位置和音频信息类型进行声场再适配，从而降低了音箱系统对音频信息的表现张力，造成了播放资源的浪费。因此，亟需一种个性化和智能化的声场适配系统实时进行声场调整。
发明内容
本公开提供一种智能声场适配系统，所述声场适配系统至少包括自检部、音频信息采 集部、数据处理部和至少一个音箱；所述数据处理部基于自检部采集音箱数量和位置信息完成对音频信息采集部采集的音频信息频段分配，并基于各音箱的分配的频段信息和听众的位置关系完成对至少一个音箱的角度进行调整，并将音箱数量和位置参数、音箱频段分配参数、听众位置参数和音箱角度调整参数作为一个声场适配模型储存于数据处理部。
根据一个可选的实施方式，所述自检部配置成基于采集的至少一个音箱标识信息和环境信息对至少一个音箱进行名称和位置的标注和统计，并对所述至少一个音箱进行等效分组得到至少一个等效组。
根据一个可选的实施方式，所述自检部具体配置成将在声场中相同位置或在听众左右两侧的对称位置的至少一个音箱划分为一个等效组。
根据一个可选的实施方式，所述数据处理部中的音频分类单元配置成基于自检部统计的等效组的数量进行频段分配。
根据一个可选的实施方式，所述数据处理部中的音频分类单元配置成：根据所述自检部划分的等效组的数量以及每个等效组中的音箱与听众的位置关系，对所述音频信息采集部采集的音频信息进行频段分配，得到至少一个频段；针对得到的每个频段息，分别确定该频段对应的音频信息的目标播放位置，并将该频段分配给处于所述目标播放位置的音箱。
根据一个可选的实施方式，所述数据处理部中的音箱调节单元配置成基于各等效组的音箱设备与听众的位置关系以及各等效组的音箱设备分配的频段信息对各等效组的音箱设备的角度进行调整。
根据一个可选的实施方式，所述音频信息采集部包括话筒和音频获取器；所述数据处理部中的音箱调节单元配置成：
基于所述自检部采集的音箱数量和位置信息对所述音频获取器采集的音频信息进行频段分配，并基于所述至少一个音箱分配到的频段信息以及所述至少一个音箱与听众的位置关系对所述至少一个音箱的角度进行初步调整；
从所述话筒采集的音频信息中确定与所述音频获取器采集的音频信息相同的目标信息，根据所述目标信息、所述频段分配的结果以及所述至少一个音箱与听众的位置关系对所述至少一个音箱的角度进行二次调整。
根据一个可选的实施方式，所述音频信息采集部祸害包括音色调整单元；
所述话筒配置成采集原唱声音数据，所述音频获取器配置成采集背景音乐数据；
所述音色调整单元配置成基于所述背景音乐数据和预设的调音函数对所述原唱声音数据经过实时调整，将调整后的原唱声音数据与所述背景音乐数据混合成一混音数据，并将所述混音数据发送至所述数据处理部进行处理。
根据一个可选的实施方式，所述数据处理部至少还设有DSP处理器，所述DSP处理器配置成基于所述音频分类单元的频段分配结果为所述音频信息采集部采集的音频数据添加频段标记和位置信息标记，并将标记完成的音频数据发送至添加的位置信息标记所对应的音箱设备，以完成对应频段的音频信息的播放。
根据一个可选的实施方式，所述数据处理部至少还设有储存单元，所述储存单元配置成对包括音箱数量、位置参数、音箱频段分配参数、听众位置参数和音箱角度调整参数的声场适配模型的数据进行采集和记录。
根据一个可选的实施方式，所述数据处理部至少还设有数据触发单元，所述数据触发单元配置成基于所述自检部采集的音箱数量参数、音箱位置参数、听众位置参数触发所述储存单元中储存的至少一个声场适配模型，并按照所触发的声场适配模型对声场内音箱的角度进行快速调整。
根据一个可选的实施方式，所述环境信息包括各音箱的位置信息、及听众位置信息以及音箱所处空间的长度、宽度和高度信息。
根据一个可选的实施方式，所述环境信息通过智能终端上传至所述自检部。
本公开还提供一种智能声场校准系统，所述声场校准系统至少包括自检部、音频信息采集部、数据处理部和至少一个音箱，所述音频信息采集部中至少设有音频获取器和话筒；
所述数据处理部基于自检部采集音箱数量和位置信息完成对音频获取器采集的音频信息数据的频段分配，并基于各音箱的分配的频段信息和听众的位置关系完成对至少一个音箱的角度初步调整；
所述数据处理部基于话筒采集的与音频获取器获取的音频信息数据相同的声音信息数据、频段分配数据和听众的位置关系完成对至少一个音箱的角度的二次调整；
并且，所述数据处理部将音箱数量和位置参数、音箱频段分配参数、听众位置参数和音箱角度调整参数作为一个声场适配模型储存于数据处理部。
本公开还提供一种智能语音处理系统，所述语音处理系统至少包括自检部、音频信息采集部、数据处理部和至少一个音箱；所述音频信息采集部中至少设有音频获取器、话筒和音色调整单元，所述音色调整单元基于预设的调音函数和音频获取器获取的音频信息中的原声频率完成对经话筒采集的声音信息的实时调整，并将调整后的话筒采集的声音数据和音频获取器采集的音频数据形成混音发送至所述数据处理部，所述数据处理部基于自检部采集音箱数量和位置信息完成对音频信息采集部采集的音频信息的频段分配。
附图说明
图1为本公开提供的一种智能声场适配系统的功能模块示意图；
图2为本公开提供的一种音频信息采集部的功能模块示意图；
图3为本公开提供的一种数据处理部的功能模块示意图。
具体实施方式
下面进一步详细描述本公开的技术方案，但本公开的保护范围不局限于以下所述。
本实施例提供一种智能声场适配系统，如图1所示。所述智能声场适配系统至少包括自检部、音频信息采集部、数据处理部和至少一个音箱。
可选地，所述自检部可以为一种数据采集器，配置成实现音箱信息的采集和分类，以及环境信息的采集。所述音频信息采集部配置成实现音频信息获取，例如通过网络完成音频信息下周或直接从硬盘中完成特定音频信息获取。
请参照图2，图2为本实施例提供的一种音频信息采集部的功能模块框图。可选地，所述音频信息采集部可以包括话筒和音频获取器。其中，所述话筒配置成采集音频信息，例如原唱声音数据；所述音频获取器配置成采集音频信息，例如背景音乐数据。
可选地，所述音频信息采集部还可以包括音色调整单元。所述音色调整单元配置成基于所述背景音乐数据中的原声频率以及预设的调音函数对所述原唱声音数据进行实时调整，将调整后的原唱声音数据与所述背景音乐数据混合成一混音数据，并将所述混音数据发送至所述数据处理部进行下级数据处理。值得说明的是，后文提及的所述音频信息采集部采集的音频信息可以是指经过所述音色调整单元处理的所述混音数据，也可以是指未经所述音色调整单元处理的音频信息，本实施例不以此为限制。
通过所述音色调整单元实现了对话筒采集的声音信息的实时调整，使得话筒采集的声音的频率更趋向于原声数据，从而使得用户通过话筒唱歌时声音更加悦耳更贴合乐调，进一步提高了用户使用音箱播放系统的友好度。
所述自检部配置成基于采集的至少一个音箱标识信息和环境信息完成对至少一个音箱进行名称和位置的标注和统计，并对所述至少一个音箱进行等效分组以得到至少一个等效组(也可以称为“音效组”)。具体地，对音箱进行等效分组为将在声场中相同位置或在听众左右两侧对称位置的至少一个音箱等效划分为同一个音效组。
可选地，所述环境信息可以包括音箱所处空间的长度、宽度和高度信息。所述环境信息还可以包括各音箱的位置信息以及听众位置信息。进一步地，所述环境信息可以由用户经智能终端上传至所述自检部，也可以通过相应的环境信息采集器件自动采集获得。
请参照图3，图3是本公开提供的一种数据处理部的功能模块框图。可选地，所述数据处理部至少包括音频分类单元、音箱调节单元、DSP处理器、储存单元和数据触发单元， 所述数据处理部配置成对音频信息进行处理并将处理后的音频信息发送至对应的音箱设备(即：音箱)。可选地，所述数据处理部还配置成基于自检部采集的音箱数量和位置信息对音频信息采集部采集的音频信息进行频段分配。
其中，所述音频分类单元配置成对所述音频信息采集部采集的音频信息进行频段信息分配和播放位置分配。可选地，所述音频分类单元配置成基于所述自检部统计的等效组的数量完成频段分配。
详细地，所述音频分类单元具体可以配置成：根据所述自检部划分的等效组的数量以及每个等效组中的音箱与听众的位置关系，对所述音频信息采集部采集的音频信息进行频段分配，以将所述音频采集部采集的音频信息划分成至少一个频段；针对得到的每个频段，确定该频段对应的音频信息的目标播放位置，并将该频段分配给处于所述目标播放位置的音箱。
所述音箱调节单元配置成实现对各个音箱进行角度调整。可选地，所述音箱调节单元具体可以配置成基于各音箱分配到的频段信息以及各音箱与听众的位置关系对至少一个音箱的角度进行调整。进一步地，所述音箱调节单元配置成基于各等效组的音箱设备与听众的位置关系以及各等效组的音箱设备分配到的频段信息对各等效组的音箱设备的角度进行调整。
可选地，在所述音频信息采集部包括话筒和音频获取器的情况下，所述数据处理部具体可以配置成通过所述音箱调节单元来进行角度调整。详细地，所述音箱调节单元可以配置成：基于所述自检部采集的音箱数量和位置信息对所述音频获取器采集的音频信息进行频段分配，并基于所述至少一个音箱分配到的频段信息以及所述至少一个音箱与听众的位置关系对所述至少一个音箱的角度进行初步调整；从所述话筒采集的音频信息中确定与所述音频获取器采集的音频信息相同的目标信息，根据所述目标信息、所述频段分配的结果以及所述至少一个音箱与听众的位置关系对所述至少一个音箱的角度进行二次调整。
所述DSP处理器配置成将采集音频信息分配至特定的音箱设备中。可选地，所述DSP处理器配置成基于所述音频分类单元的频段分配结果为所述音频信息采集部采集的音频数据添加频段标记和位置信息标记。具体地，所述DSP处理器将频段分配信息和位置分配信息作为标注数据标注于采集的音频信息中。同时，所述DSP处理器将标记完成的音频数据发送至位置信息(即：添加的所述位置信息标记所指示的位置信息)对应音的箱设备，以完成对应频段的音频信息的播放。
至少一个音箱设备接收所述DSP处理器发送的音频数据，并根据接收的音频数据中的频段标注信息(即：前述的频段标记)播放对应频段的音频数据。
所述数据处理部中的储存单元配置成对包括音箱数量、位置参数、音箱频段分配参数、听众位置参数和音箱角度调整参数的声场适配模型的数据进行采集和记录。如此，可以将至少一个声场适配模型储存于数据处理部的所述储存单元中，以便后续使用过程中能够快速地触发和调用所述储存单元中记录的声场适配模型。
所述数据处理部中的数据触发单元配置成基于所述自检部实时采集的音箱数量参数、音箱位置参数、听众位置参数触发所述储存单元中记录的至少一个声场适配模型，并按照所触发的声场适配模型来对声场内音箱的角度进行快速调整。如此，实现了在音箱数量参数、音箱位置参数、听众位置参数发生改变时，整个音箱播放系统能够基于储存的声场适配模型快速地完成音箱播放系统的实时和智能调节，为用户带来更好听觉体验。
本实施例还提供一种智能声场校准系统，如图1所示。所述声场校准系统至少包括自检部、音频信息采集部、数据处理部和至少一个音箱。
可选地，所述自检部为一种数据采集器，配置成实现音箱信息的采集和分类，以及环境信息的采集。所述音频信息采集部配置成实现音频信息获取，例如通过网络完成音频信息下载或直接从硬盘中完成特定音频信息获取。
所述自检部基于采集的至少一个音箱标识信息和环境信息完成对至少一个音箱进行名称和位置的标注和统计，并进行等效分组。具体地，对音箱进行等效分组为将在声场中相同位置或在听众左右两侧对称位置的至少一个音箱等效分为同一个音效组。
可选地，所述环境信息为音箱所处空间的长度、宽度和高度信息。所述环境信息还包括各音箱的位置信息以及听众位置信息。进一步地，所述环境信息可以由用户经智能终端上传至自检部。
可选地，请参照图2，所述音频信息采集部中至少设有音频获取器和话筒，分别配置成完成背景音乐数据和原唱声音数据的采集。
可选地，请参照图3，所述数据处理部至少包括音频分类单元、音箱调节单元、DSP处理器、储存单元和触发单元，所述数据处理部配置成完成音频信息处理并发送至对应的音箱设备。可选地，所述数据处理部基于自检部采集音箱数量和位置信息完成对音频信息采集部采集的音频信息进行频段分配。
其中，所述音频分类单元配置成完成对音频信息采集部采集的音频信息进行频段信息分配和播放位置分配。可选地，所述音频分类单元基于自检部统计的音箱设备的等效组数完成频段分配。
所述音箱调节单元配置成实现对各个音箱进行角度调整。所述音箱调节单元基于各等 效组音箱设备与听众的位置关系以及各等效组音箱设备分配的频段信息完成对各等效组音箱设备的角度调整。
可选地，所述音箱调节单元基于自检部采集音箱数量和位置信息完成对音频获取器采集的音频信息数据的频段分配，并基于各音箱的分配的频段信息和听众的位置关系完成对至少一个音箱的角度初步调整。
进一步地，所述音箱调节单元基于话筒采集的与音频获取器获取的音频信息数据相同的声音信息数据、频段分配数据和听众的位置关系完成对至少一个音箱的角度的二次调整。
所述DSP处理器配置成实现将采集音频信息分配至特定的音箱设备中。可选地，所述DSP处理器配置成基于所述音频分类单元的频段分配结果对音频信息采集部采集的音频数据进行频段标记和位置信息标记。具体地为所述DSP处理器将频段分配信息和位置分配信息作为标注数据标注于采集的音频信息中。同时，所述DSP处理器将标记完成的音频数据发送至位置信息对应音箱设备处以对对应频段的音频信息进行播放。
至少一个音箱设备完成对DSP处理器发送的音频数据进行接收，并根据接收的音频数据中的频段标注信息播放对应频段的音频数据。
所述数据处理部中储存单元配置成完成对包括音箱数量和位置参数、音箱频段分配参数、听众位置参数和音箱角度调整参数的声场适配模型的数据进行采集和记录，从而可以通过所述储存单元来存储至少一个声场适配模型，以便后续使用过程中能够完成声场适配模型的快速触发和调用。
所述数据处理部中数据触发单元基于自检部实时采集的音箱数量参数、音箱位置参数、听众位置参数实现与储存单元中储存的至少一个声场适配模型进行触发，完成声场内音箱角度的快速调整。如此，实现了在音箱数量参数、音箱位置参数、听众位置参数发生改变时，整个音箱播放系统能够基于储存的声场适配模型完成音箱播放系统的实时和智能调节，为用户带来更好听觉体验。
本实施例还提供一种智能语音处理系统，如图1所示。所述语音处理系统至少包括自检部、音频信息采集部、数据处理部和至少一个音箱。
可选地，所述自检部为一种数据采集器，配置成实现音箱信息的采集和分类，以及环境信息的采集。所述音频信息采集部配置成实现音频信息获取，例如通过网络完成音频信息下载或直接从硬盘中完成特定音频信息获取。
所述自检部基于采集的至少一个音箱标识信息和环境信息完成对至少一个音箱进行名称和位置的标注和统计，并进行等效分组。具体地，对音箱进行等效分组为将在声场中处 于相同位置或处于听众左右两侧的对称位置的至少一个音箱等效分为同一个音效组(即：前述的等效组)。
可选地，所述环境信息包括音箱所处空间的长度、宽度和高度信息。所述环境信息还可以包括各音箱的位置信息以及听众位置信息。进一步地，所述环境信息可以由用户经智能终端上传至自检部，也可以通过预设的环境信息采集器件采集获得。
可选地，请参照图2，所述音频信息采集部中至少设有音频获取器、话筒和音色调整单元，配置成对整个语音处理系统的所有声音信息数据进行采集。所述音色调整单元配置成基于预设的调音函数以及所述音频获取器获取的音频信息中的原声频率对所述话筒采集的声音信息进行实时调整。并且，所述音色调整单元可以将调整后的话筒采集的声音数据(例如，调整后的原唱声音数据)和音频获取器采集的音频数据(例如，所述背景声音数据)形成混音数据发送至所述数据处理部进行下级数据处理。进一步地，通过所述音色调整单元实现了对话筒采集的声音信息的实时调整，使得话筒采集的声音的频率数据更趋向于原声数据，从而使得用户通过话筒进行唱歌时，声音更加悦耳更贴合乐调，进一步提高了用户使用本音箱播放系统的友好度。
可选地，请参照图3，所述数据处理部至少包括音频分类单元、音箱调节单元、DSP处理器、储存单元和触发单元，所述数据处理部配置成完成音频信息处理并发送至对应的音箱设备。优选地，所述数据处理部基于自检部采集音箱数量和位置信息完成对音频信息采集部采集的音频信息频段分配。
其中，所述音频分类单元配置成完成对音频信息采集部采集的音频信息进行频段信息分配和播放位置分配。可选地，所述音频分类单元基于自检部统计的音箱设备的等效组的数量进行频段分配。
所述音箱调节单元配置成实现对各个音箱进行角度调整。可选地，所述音箱调节单元基于各音箱的分配的频段信息和听众的位置关系完成对至少一个音箱的角度进行调整。进一步地，所述音箱调整单元基于各等效组的音箱设备与听众的位置关系以及各等效组的音箱设备分配到的频段信息对各等效组的音箱设备的角度进行调整。
所述DSP处理器配置成将采集音频信息分配至特定的音箱设备中。可选地，所述DSP处理器配置成基于音频分类单元的频段分配结果对所述音频信息采集部采集的音频数据进行频段标记和位置信息标记。具体地，所述DSP处理器将频段分配信息和位置分配信息作为标注数据标注于采集的音频信息中。同时，所述DSP处理器将标记完成的音频数据发送至位置信息对应音箱设备处完成对应频段音频信息的播放。
至少一个音箱设备完成对DSP处理器发送的音频数据进行接收，并根据接收的音频数 据中的频段标注信息完成对应频段音频数据的播放。
所述数据处理部中储存单元配置成完成对包括音箱数量和位置参数、音箱频段分配参数、听众位置参数和音箱角度调整参数的声场适配模型的数据采集和记录。从而可以通过所述储存单元完成将至少一个声场适配模型储存于数据处理部，以便后续使用过程中能够完成声场适配模型的快速触发和调用。
所述数据处理部中数据触发单元基于自检部实时采集的音箱数量参数、音箱位置参数、听众位置参数实现与储存单元中储存的至少一个声场适配模型进行触发，完成声场内音箱角度的快速调整。从而实现了在音箱数量参数、音箱位置参数、听众位置参数发生改变时，整个音箱播放系统能够基于储存的声场适配模型完成音箱播放系统的实时和智能调节，为用户带来更好听觉体验。
以上所述实施例仅表达了本公开的具体实施方式，其描述较为具体和详细，但并不能因此而理解为对本公开保护范围的限制。应当指出的是，对于本领域的普通技术人员来说，在不脱离本公开构思的前提下，还可以做出若干变形和改进，这些都属于本公开的保护范围。
工业实用性
本公开提供的智能声场适配系统、智能声场校准系统及智能语音处理系统，能够针对特定的声场环境进行实时的声场适配和声场调整，从而满足用户对声场的个性化需求。

Claims (15)

  1. 一种智能声场适配系统，其特征在于，所述声场适配系统至少包括自检部、音频信息采集部、数据处理部和至少一个音箱；
    所述数据处理部基于自检部采集音箱数量和位置信息完成对音频信息采集部采集的音频信息频段分配，并基于各音箱的分配的频段信息和听众的位置关系完成对至少一个音箱的角度进行调整，并将音箱数量和位置参数、音箱频段分配参数、听众位置参数和音箱角度调整参数作为一个声场适配模型储存于数据处理部。
  2. 如权利要求1所述的智能声场适配系统，其特征在于，所述自检部配置成基于采集的至少一个音箱标识信息和环境信息对至少一个音箱进行名称和位置的标注和统计，并对所述至少一个音箱进行等效分组得到至少一个等效组。
  3. 如权利要求2所述的智能声场适配系统，其特征在于，所述自检部具体配置成将在声场中相同位置或在听众左右两侧的对称位置的至少一个音箱划分为一个等效组。
  4. 如权利要求2或3所述的智能声场适配系统，其特征在于，所述数据处理部中的音频分类单元配置成基于所述自检部统计的等效组的数量进行频段分配。
  5. 如权利要求2或3所述的智能声场视频系统，其特征在于，所述数据处理部中的音频分类单元配置成：根据所述自检部划分的等效组的数量以及每个等效组中的音箱与听众的位置关系，对所述音频信息采集部采集的音频信息进行频段分配，以将所述音频采集部采集的音频信息划分成至少一个频段；针对每个频段，确定该频段对应的音频信息的目标播放位置，并将该频段分配给处于所述目标播放位置的音箱。
  6. 如权利要求4或5所述的智能声场适配系统，其特征在于，所述数据处理部中的音箱调节单元配置成基于各等效组的音箱设备与听众的位置关系以及各等效组的音箱设备分配的频段信息对各等效组的音箱设备的角度进行调整。
  7. 如权利要求1所述的智能声场适配系统，其特征在于，所述音频信息采集部包括话筒和音频获取器；所述数据处理部中的音箱调节单元配置成：
    基于所述自检部采集的音箱数量和位置信息对所述音频获取器采集的音频信息进行频段分配，并基于所述至少一个音箱分配到的频段信息以及所述至少一个音箱与听众的位置关系对所述至少一个音箱的角度进行初步调整；
    从所述话筒采集的音频信息中确定与所述音频获取器采集的音频信息相同的目标信息，根据所述目标信息、所述频段分配的结果以及所述至少一个音箱与听众的位置关系对所述至少一个音箱的角度进行二次调整。
  8. 如权利要求7所述的智能声场适配系统，其特征在于，所述音频信息采集部还包括 音色调整单元；
    所述话筒配置成采集原唱声音数据，所述音频获取器配置成采集背景音乐数据；
    所述音色调整单元配置成基于所述背景音乐数据中的原声频率和预设的调音函数对所述原唱声音数据进行实时调整，将调整后的原唱声音数据与所述背景音乐数据混合成一混音数据，并将所述混音数据发送至所述数据处理部进行处理。
  9. 如权利要求4-6中任意一项所述的智能声场适配系统，其特征在于，所述数据处理部至少还设有DSP处理器，所述DSP处理器配置成基于所述音频分类单元的频段分配结果为音频信息采集部采集的音频数据添加频段标记和位置信息标记，并将标记完成的音频数据发送至添加的位置信息标记所对应的音箱设备，以完成对应频段的音频信息的播放。
  10. 如权利要求1-9中任意一项所述的智能声场适配系统，其特征在于，所述数据处理部至少还设有储存单元，所述储存单元配置成对包括音箱数量、位置参数、音箱频段分配参数、听众位置参数和音箱角度调整参数的声场适配模型的数据进行采集和记录。
  11. 如权利要求10所述的智能声场适配系统，其特征在于，所述数据处理部至少还设有数据触发单元，所述数据触发单元基于所述自检部采集的音箱数量参数、音箱位置参数及听众位置参数触发所述储存单元中储存的至少一个声场适配模型，按照所触发的声场适配模型对声场内音箱的角度进行快速调整。
  12. 如权利要求2-11中任意一项所述的智能声场适配系统，其特征在于，所述环境信息包括各音箱的位置信息、听众位置信息以及音箱所处空间的长度、宽度和高度信息。
  13. 如权利要求2-12中任意一项所述的智能声场适配系统，其特征在于，所述环境信息通过智能终端上传至所述自检部。
  14. 一种智能声场校准系统，其特征在于，所述声场校准系统至少包括自检部、音频信息采集部、数据处理部和至少一个音箱，所述音频信息采集部中至少设有音频获取器和话筒；
    所述数据处理部基于自检部采集音箱数量和位置信息完成对音频获取器采集的音频信息数据的频段分配，并基于各音箱的分配的频段信息和听众的位置关系完成对至少一个音箱的角度初步调整；
    所述数据处理部基于话筒采集的与音频获取器获取的音频信息数据相同的声音信息数据、频段分配数据和听众的位置关系完成对至少一个音箱的角度的二次调整；
    并且，所述数据处理部将音箱数量和位置参数、音箱频段分配参数、听众位置参数和音箱角度调整参数作为一个声场适配模型储存于数据处理部。
  15. 一种智能语音处理系统，其特征在于，所述语音处理系统至少包括自检部、音频 信息采集部、数据处理部和至少一个音箱；所述音频信息采集部中至少设有音频获取器、话筒和音色调整单元，所述音色调整单元基于预设的调音函数和音频获取器获取的音频信息中的原声频率完成对经话筒采集的声音信息的实时调整，并将调整后的话筒采集的声音数据和音频获取器采集的音频数据形成混音发送至所述数据处理部，所述数据处理部基于自检部采集音箱数量和位置信息完成对音频信息采集部采集的音频信息的频段分配。
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