CN107547984A - A kind of audio-frequency inputting method and audio output system based on intelligent terminal - Google Patents
A kind of audio-frequency inputting method and audio output system based on intelligent terminal Download PDFInfo
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Abstract
The invention provides a kind of audio-frequency inputting method based on intelligent terminal, comprise the following steps：An OTG lines are obtained, are connected to the connector of the intelligent terminal；An external audio output device is connected to the USB interface of the OTG lines；In running an audio file in the intelligent terminal, to form an audio stream；The audio stream is decoded, to resolve at least two voice data streams；The loudspeaker module that the voice data flow to the intelligent terminal is exported respectively, and is exported by connector, the OTG lines of the intelligent terminal to the external audio output device；Control loudspeaker module and external audio output device the playing audio data stream.After adopting the above technical scheme, it can simultaneously meet using earphone, put outside, the audio of the user of sound equipment connection intelligent terminal listens to demand.
Description
Technical field
The present invention relates to field of intelligent control, more particularly to a kind of audio-frequency inputting method and audio based on intelligent terminal are defeated
Go out system.
Background technology
With the increase of intelligent terminal frequency of use in life, the user of intelligent terminal increasingly payes attention to the use of individual
Experience, and usage experience also determine the pouplarity of intelligent terminal.Have in usage experience and relate in one aspect to very much intelligence greatly
The control aspect of terminal.When user needs the broadcasting to the audio in intelligent terminal, generally select and use the outer loudspeaker put to go out
Sound mouth or connection earphone listen to the sound of audio.
Due to an only passage for audio output, when user is multiple, multi-user can not be met while listen to audio
Requirement.
Therefore, it is necessary to which a kind of audio-frequency inputting method and output system that multi output can be carried out to single audio frequency, meets multi-purpose
Listen to a demand during audio in family.
The content of the invention
In order to overcome above-mentioned technological deficiency, it is an object of the invention to provide a kind of audio output side based on intelligent terminal
Method and audio output system, it can simultaneously meet using earphone, put outside, the audio of the user of sound equipment connection intelligent terminal listens to need
Ask.
The invention discloses a kind of audio-frequency inputting method based on intelligent terminal, comprise the following steps：
An OTG lines are obtained, are connected to the connector of the intelligent terminal；
An external audio output device is connected to the USB interface of the OTG lines；
In running an audio file in the intelligent terminal, to form an audio stream；
The audio stream is decoded, to resolve at least two voice data streams；
The loudspeaker module that the voice data flow to the intelligent terminal is exported respectively, and passes through the intelligent terminal
Connector, OTG lines are exported to the external audio output device；
Control loudspeaker module and external audio output device the playing audio data stream.
Preferably, the audio stream is decoded, is included the step of to resolve at least two voice data streams：
The audio stream is changed into frequency domain by time-domain；
The audio stream based on the frequency domain, several sub-bands are divided into using bandpass filter.
Preferably, the step of controlling loudspeaker module and external audio output device the playing audio data stream is wrapped
Include：
Digital demodulation is carried out to the voice data stream for being configured to the sub-band；
Using low pass filter, the voice data stream is reverted to the distribution of the audio stream；
By the audio streams to a synchronous adder, the audio file is reverted to by addition.
Preferably, in running an audio file in the intelligent terminal, to form an audio stream with decoding the audio stream,
Also include between the step of to resolve into audio output described at least two voice data streams：
The waveform of the audio stream is sampled, quantifies the amplitude sample layering of the audio stream, is translated into digital code；
The digital code is reconstructed, to compress the audio stream.
The invention also discloses a kind of audio output system based on intelligent terminal, the audio output system includes：
Intelligent terminal, operation has an audio file, to form an audio stream；
OTG lines, it is connected to the connector of the intelligent terminal；
External audio output device, it is connected with the USB interface of the OTG lines；
The intelligent terminal includes processor, and the processor includes：
Decoder module, the audio stream is decoded, to resolve at least two voice data streams；
Output module, export the voice data and flow to a loudspeaker module of the intelligent terminal, and pass through institute
State the connector of intelligent terminal, OTG lines are exported to the external audio output device；
Loudspeaker module and external audio output device the playing audio data stream.
Preferably, the decoder module includes：
Domain conversion unit, the audio stream is changed into frequency domain by time-domain；
Bandpass filter, it is several sub-bands by the finite state Automat based on the frequency domain.
Preferably, loudspeaker module and the external audio output device includes：
Demodulating unit, digital demodulation is carried out to the voice data stream for being configured to the sub-band；
Low pass filter, the voice data stream is reverted to the distribution of the audio stream；
Synchronous adder, the audio stream is added to revert to the audio file.
Preferably, the audio output system also includes：
Compression module, in the processor, it is connected with the decoder module, wherein the compression module includes：
Conversion unit, the waveform of the audio stream is sampled, quantify the amplitude sample layering of the audio stream, be translated into
Digital code；
Reconfiguration unit, the digital code is reconstructed, to compress the audio stream.
After employing above-mentioned technical proposal, compared with prior art, have the advantages that：
1. intelligent terminal can export multi-channel audio simultaneously by earpiece holes, USB interface etc., meet the use of different user
It is required that；
2. distortion and the reduction of discrimination degree even will not also be caused to audio to the decompression of audio stream.
Brief description of the drawings
Fig. 1 is the schematic flow sheet for meeting the audio-frequency inputting method based on intelligent terminal in one embodiment of the present invention；
Fig. 2 is to meet the schematic flow sheet that voice data stream is decomposed in one embodiment of the present invention；
Fig. 3 is the schematic flow sheet for meeting audio stream plays in one embodiment of the present invention；
Fig. 4 is the flow signal for meeting the audio-frequency inputting method based on intelligent terminal in another preferred embodiment of the present invention
Figure；
Fig. 5 is the structural representation for meeting the audio output system based on intelligent terminal in one embodiment of the present invention；
Fig. 6 is the structural representation for meeting decoder module in one embodiment of the present invention；
Fig. 7 is to meet the structural representation of loudspeaker and external audio output device in one embodiment of the present invention；
Fig. 8 is the structural representation for meeting the audio output system based on intelligent terminal in another preferred embodiment of the present invention
Figure.
Embodiment
Advantages of the present invention is expanded on further below in conjunction with accompanying drawing and specific embodiment.
Here exemplary embodiment will be illustrated in detail, its example is illustrated in the accompanying drawings.Following description is related to
During accompanying drawing, unless otherwise indicated, the same numbers in different accompanying drawings represent same or analogous key element.Following exemplary embodiment
Described in embodiment do not represent all embodiments consistent with the disclosure.On the contrary, they be only with it is such as appended
The example of the consistent apparatus and method of some aspects be described in detail in claims, the disclosure.
It is only merely for the purpose of description specific embodiment in the term that the disclosure uses, and is not intended to be limiting the disclosure.
" one kind " of singulative used in disclosure and the accompanying claims book, " described " and "the" are also intended to including majority
Form, unless context clearly shows that other implications.It is also understood that term "and/or" used herein refers to and wrapped
Containing the associated list items purpose of one or more, any or all may be combined.
It will be appreciated that though various information, but this may be described using term first, second, third, etc. in the disclosure
A little information should not necessarily be limited by these terms.These terms are only used for same type of information being distinguished from each other out.For example, do not departing from
In the case of disclosure scope, the first information can also be referred to as the second information, and similarly, the second information can also be referred to as
One information.Depending on linguistic context, word as used in this " if " can be construed to " ... when " or " when ...
When " or " in response to determining ".
In the description of the invention, it is to be understood that term " longitudinal direction ", " transverse direction ", " on ", " under ", "front", "rear",
The orientation or position relationship of the instruction such as "left", "right", " vertical ", " level ", " top ", " bottom " " interior ", " outer " is based on accompanying drawing institutes
The orientation or position relationship shown, it is for only for ease of the description present invention and simplifies description, rather than instruction or the dress for implying meaning
Put or element there must be specific orientation, with specific azimuth configuration and operation, therefore it is not intended that to limit of the invention
System.
In the description of the invention, unless otherwise prescribed with limit, it is necessary to explanation, term " installation ", " connected ",
" connection " should be interpreted broadly, for example, it may be mechanical connection or electrical connection or the connection of two element internals, can
To be to be joined directly together, can also be indirectly connected by intermediary, for the ordinary skill in the art, can basis
Concrete condition understands the concrete meaning of above-mentioned term.
In follow-up description, the suffix using such as " module ", " part " or " unit " for representing element is only
Be advantageous to the explanation of the present invention, itself do not have a specific meaning.Therefore, " module " can mixedly make with " part "
With.
Refering to Fig. 1, show to meet the flow of the audio-frequency inputting method based on intelligent terminal in one embodiment of the present invention
It is intended to.In this embodiment, to realize the multi output to audio, comprise the following steps：
S100：An OTG lines are obtained, are connected to the connector of intelligent terminal
OTG lines are based on, and OTG technologies are On-The-Go, be exactly it is real there are currently no Host in the case of, realize between slave unit
Data transmission.Such as digital camera is directly connected on printer, by OTG technologies, the USB port that two equipment is shown in is connected,
The photograph taken is printed immediately；Data in digital camera can also be sent to the shifting of USB interface by OTG
On dynamic hard disk, field operation is just not necessarily to carry expensive storage card, or one portable computer of the back of the body.Pass through OTG skills
Art, can give intelligent terminal connector connect with enrich intelligent terminal function, such as extension remote control accessory, mobile phone,
Flat board becomes universal remote control use.
S200：An external audio output device is connected to the USB interface of OTG lines
In the other end of OTG lines, an external audio output device is connected with, for example, the playable audio such as sound equipment, earphone
Device.
S300：In running an audio file in intelligent terminal, to form an audio stream
An application program that can be used for parsing and playing audio file can be opened in intelligent terminal in this step, by user
It is selected that operation broadcasting is carried out to some audio files, so as to form an association to the audio stream of the audio file.
S400：Decoded audio stream, to resolve at least two voice data streams
To realize the multi output of audio stream, the audio stream need to be decomposed at least two voice data streams, each audio number
Can be the content of the sound channel in identical, or multi-channel audio stream according to stream.
S500：Output voice data flow to the loudspeaker module of intelligent terminal respectively, and by the connector of intelligent terminal,
OTG lines are exported to the external audio output device
After being provided with multitone frequency data stream, selection is therein all or two export to the loudspeaker module of intelligent terminal,
And by the connector such as USB port of intelligent terminal, then exported by OTG lines to external audio output device.
S600：After loudspeaker module and external audio output device receive voice data stream, by broadcasting user controlled, or
It is arranged to once receive voice data stream, voice data stream will be played out.
Refering to Fig. 2, above-mentioned decoded audio stream, need to carry out audio stream at least two voice data flow steps to resolve into
Following processing：
S401：Audio stream is changed into frequency domain by time-domain；
S402：Audio stream based on frequency domain, several sub-bands are divided into using bandpass filter
DTS (decoded time stamp) and PTS (Presentation Time Stamp) be respectively when decoder is decoded and shown frame relative to
SCR (system reference) timestamp.SCR can be understood as decoder should start from disk read data when time.
By taking the video file of mpeg forms as an example, each bag in mpeg files have a SCR timestamp and this
Individual timestamp is exactly system time when reading this packet.Under normal circumstances, decoder can start to read mpeg streams at it
When activation system clock (initial value of system clock is the SCR values of first packet, 0 but can not also usually be opened from 0
Begin).DTS timestamps determine that decoder is decoded when the SCR times being equal to the DTS times, and PTS timestamps are also similar.
Generally, what DTS/PTS timestamps indicated is the time for the SCR being later than in audio frequency and video bag.An if for example, video data
The SCR of bag is 100ms (meaning that this bag is to play 100ms to read from disk later), then DTS/PTS values are just similar
200/280ms, show as SCR to 200ms this video data should be decoded and be revealed after 80ms (depending on
Frequency evidence is saved in always in a buffer to be started to decode) underflow generally video data stream correlation mux of the generation in setting
Rate is too high.If mux rates, which are 1000000bits/sec, (means that decoder will read text with 1000000bits/sec speed
Part), but video rate is 2000000bits/sec (is intended to show video counts with 2000000bits/sec speed
According to), speed is not fast so that can not read enough video datas in 1 second enough during reading video data from disk.This
DTS/PTS timestamps will indicate that video is decoded or shown before being read out from hard disk in the case of kind.
It is the change made based on sub-band coding technology in above-mentioned steps S401, it is by the primary signal of audio stream
Frequency domain is changed into by time-domain, is then divided into several sub-bands, and digitally coded technology is carried out respectively to it.
It is that primary signal is divided into some (such as m) sub-bands (abbreviation subband) using bandpass filter (BPF) group.
Refering to Fig. 3, it is controlling loudspeaker module and external audio output device playing audio-fequency data stream, following step need to be performed
Suddenly：
S601：In loudspeaker module or external audio output device, the voice data stream for being configured to sub-band is carried out
Digital demodulation；
S602：Using low pass filter, voice data stream is reverted to the distribution of audio stream；
S603：By audio streams to a synchronous adder, audio file is reverted to by addition.
The voice data stream of input subband coding, corresponding digital decoding circuit (common m is sent to by each subband signal respectively
It is individual) digital demodulation is carried out, new demodulation of laying equal stress on, can be former originally each subband frequency restoration by all road low pass filters (m roads)
The distribution of beginning signal.Finally, each way band output signal is sent to synchronous adder, original letter is reverted to by addition
Number, the signal and primary signal of the recovery are quite similar.
There is the advantages of prominent by means of above-mentioned sub-band coding technology.First, the range value of each frequency component of sound spectrum
It is different, if distributing coefficient in proper proportions to different sub-band, the quantization electricity of each subband can be more reasonably controlled respectively
Flat number and corresponding reconstruction error, signal source characteristic of the code check more accurately with each subband is set to match.Generally, in low frequency base
Near sound, sampling value is represented using larger bit number, and can then be distributed with less coded-bit in high band.Its
It is secondary, by the bit number of reasonable distribution different sub-band, can control total reconstruction error spectral shape, by with psychoacoustic model
It is combined, noise spectrum can be perceived characteristic to be formed by human ear subjective noise.Then, can be saved using human auditory system masking effect
Save a large amount of bit numbers.
When using sub-band coding, the masking effect of the sense of hearing make use of to be handled.It is deleted to some subband signals
Remove or largely reduce bit number, can obvious compressed transmission data total amount.Such as the subband in the absence of signal frequency component, quilt
The subband of the signal frequency of masking by noise, by signal frequency component subband of neighbouring strong signal masking etc., it can all carry out at deletion
Reason.In addition, the frequency band range of system-wide transmission information content and signal, dynamic range etc. have relation, and dynamic range is then determined
Due to quantizing bit number, if introducing rational bit number to signal, can make to give different bit numbers in different sub-band on demand,
Also its compressible information content.
Refering to Fig. 4, pressure is played in the audio of loudspeaker module side and external audio output device side to reduce, need to be to sound
Frequency stream compression transmission, specifically, by the waveform of sampled audio stream, the amplitude sample of quantization audio stream is layered, and is translated into number
After word code, reconstructed number code, to compress audio stream.
In fact, in the above-described embodiments, audio stream initially enters digital filter group, it is divided into 32 of equiband
Sub-band, 32 subband data-signals can be exported by digital filter.This processing method carries out DCT changes with image coding signal
The effect changed is similar, but is not to be divided into 64 kinds of cosine frequency informations as picture signal, is only divided into 32 subbands here, will
Voice data stream is changed to the combination of 32 kinds of frequencies.The decomposing force of sound is less than image, and this processing method is feasible.Then,
Re-quantization is carried out to the sound accompaniment data of 32 subbands, to recompress data volume.For the quantization step not phase of each sub-band
Together, quantization step is determined according to the threshold of audibility and masking effect of human ear.By the compressed data of quantification treatment, protect
The main part of sound information has been stayed, and having given up auditory effect influences less sound information.Into the input of coded system
Signal, shunting part signal are sent to 1024 fast fourier transform devices (FFT) arranged side by side and enter line translation, and it detects input signal
For each moment sample point in the intensity of the distribution of dominant frequency spectral component frequency domain, transformed signal is sent to psychoacoustic model control
Unit.According to auditory psychology acoustic measurement statistical result, a psychologic acoustics control control form can be summarized, and according to this
Control unit is made in form, and element circuit can intensively reflect the threshold property and masking characteristics of human ear.By quantifying
32 subband datas compressed, also to add the auxiliary information such as scale factor, allocation information, be added to 1 jointly
Stream formatted unit, coding turn into the sound accompaniment encoded signal of two levels.Both the sound accompaniment containing 32 sub-bands was digital for it, and band
There is the strong and weak scale factor of the bit allocation data and different frequency bands data corresponding to these numbers., can when data decoding in future
According to the data recovery voice signal of each sub-band, and compression time-code bit allocation and strong and weak ratio situation, inverse quantization is being carried out
When, reduced with reference to program when compressing.
It can be seen that as image procossing, it is treated will also to pass through conversion, quantization, code bit compression etc. for the compressed encoding of sound accompaniment
Journey, it has used the statistics that many mathematical modelings and psycho acoustic measure, to 32 sub-bands and signal at all levels
Processing also respectively has the sampling rate differed.Actual psycho-acoustic model and in good time processing control process are sufficiently complex.These
Algorithm details has all been cured in decoding chip by hardware mode, and these contents can not change again.
Other compression methods are such as：The coding form that parameter coding and multiple technologies mutually merge is also applicable.Parameter is compiled
Code is first according to different signal sources, and such as speech signal, natural sound form establishes characteristic model, by extracting characteristic parameter
And coded treatment, try hard to the meaning of one's words of holding original sound for making the voice signal of reconstruction as high as possible, but the waveform of reconstruction signal is same
The waveform of primary sound sound signal might have sizable difference.Conventional characteristic parameter has formant, linear predictor coefficient, frequency band
The parametric coding techniques such as division filters can realize the sound signal encoding of low rate, and bit rate is compressible to 2Kbit/s-
4.8Kbit/s, but the quality of sound can only achieve medium, and particularly naturalness is relatively low, is suitable only for transmission and the table of language
Reach.The coding form that waveform coding and parameter coding combine is overcome original waveform coding and parameter coding by hybrid coding
Weakness, try hard to keep waveform coding high quality and parameter coding low rate, can be obtained in 4-16Kbit/s speed
The synthetic video signal of high quality.The basis of hybrid coding is linear predictive coding (LPC), commonly uses pulse excitation linear prediction and compiles
The coding staffs such as code (MPLPC), planning pulse excitation linear predictive coding (KPELPC) codebook excited linear predictive coding (CELPC)
Formula.
In above-mentioned any embodiment, the form of audio file is not restricted by, and for any kind of form, is only needed
Decoding is coordinated to complete the output of Multi-audio-frequency in above-described embodiment.Such as：
CD forms, WAVE (* .WAV), AIFF, AU, MP3, MIDI, WMA, RealAudio, VQF, OggVorbis, AAC,
APE。
It is common to MP3, WMA, OGG be referred to as lossy compression method, lossy compression method is exactly to reduce audio sample frequency as its name suggests
Rate and bit rate, the audio file of output can be smaller than original.Another audio compression is referred to as Lossless Compression, Neng Gou
On the premise of 100% preserves all data of original, by the smaller of the volume compression of audio file, and by the audio after compression
After file reduction, it can realize and source file identical size, identical code check.At present nondestructive compression type have APE, FLAC,
WAV, WavPack, LPAC, WMALossless, AppleLossless, TTA, Tak, La, OptimFROG, Shorten, and it is normal
Nondestructive compression type see, main flow has APE, FLAC, TTA, TAK, WAV at present.
APE forms
APE is one of currently a popular digital music file form.Different from this kind of lossy compression method modes of MP3, APE is one
Kind Lossless Compression Audiotechnica, that is to say, that after the audio data file read from audio CD is compressed into APE forms, also
The file of APE forms can be reduced again, and the audio file after reducing and the striking resemblances before compression, without any loss.
APE file size is about CD half, but as the popularization in broadband, APE forms receive many music-lovers'
Like, especially for for the friend for wishing to pass through network transmission audio CD, APE can help them to save substantial amounts of money
Source.
APE essence, it is a kind of Lossless Compression audio format in fact.Huge WAV audio files can pass through
This Software Compression of Monkey'sAudio is APE.Many times it is used as network audio file transmission, because after being compressed
APE file sizes it is more half as large than WAV source file more, the transmission time used can be saved.Importantly, pass through
Source file before the wav file obtained after Monkey'sAudio decompression reduction can be accomplished and compress is completely the same.So
APE is described as " Lossless Audio Compression form ", and Monkey'sAudio is described as " Lossless Audio Compression software ".With use
This kind of expert data compressed softwares of WinZip or WinRAR come compacted voice file difference, the APE audio files after compressing
Can directly it be played.Monkey'sAudio can install " in_APE.dll " plug-in unit into Winamp, so that
Winamp also possesses the ability for playing APE files.Same foobar2000, and 1,000 hark the broadcasting that can also support APE.
FLAC forms
FLAC is Free Lossless Audio Codec abbreviation, and Chinese can solve to be encoded for Lossless Audio Compression.
FLAC is a set of famous free compressed audio coding, is characterized in Lossless Compression.Different from other lossy compression methods coding such as MP3
And AAC, it will not destroy any original audio information, it is possible to reduce music CD tonequality.It is by many soft now
Part and hardware audio product are supported.FLAC is similar with MP3, but is Lossless Compression, that is to say, that audio is in a manner of FLAC
Compression will not lose any information.This compression is similar with Zip mode, but FLAC is by bigger compression ratio, because
FLAC is specific to the compress mode designed the characteristics of audio, and the file that player plays FLAC can be used to compress.
FLAC is free and supports most operating system, including Windows, " unix " (Linux, BSD, Solaris,
OSX, IRIX), BeOS, OS/2, and Amiga.And FLAC is provided in developing instrument autotools, MSVC, Watcom C,
With the build systems on ProjectBuilder.Audio (PCM) data being encoded do not have any information loss, decoding output
Audio and each byte of input of encoder be just as.Each data frame has the 16-bit of a present frame
CRC check code, for Monitoring Data error of transmission.To whole section audio data, one is also preserved in file header for original
The MD5 marks of uncompressed voice data, for being verified in decoding and test to data.
Refering to Fig. 5-8, the audio output system based on intelligent terminal in different embodiments or preferred embodiment is respectively illustrated
The structural representation of system.In a wherein embodiment, the processing to audio stream is completed by the processor in intelligent terminal, specifically
Ground, audio output system include：
Intelligent terminal, operation has an audio file, to form an audio stream；
OTG lines, it is connected to the connector of intelligent terminal；
External audio output device, it is connected with the USB interface of OTG lines；
Intelligent terminal includes processor, and processor includes：Decoder module, the audio stream is decoded, to resolve at least two
Individual voice data stream；Output module, output voice data flow to a loudspeaker module of intelligent terminal, and passes through intelligent terminal
Connector, OTG lines are exported to external audio output device；Loudspeaker module and external audio output device playing audio-fequency data
Stream.
Further, decoder module includes：Domain conversion unit, audio stream is changed into frequency domain by time-domain；Band logical is filtered
Ripple device, it is several sub-bands by the finite state Automat based on frequency domain.Domain is converted with the mode of finite state Automat hereinbefore
It has been be described in detail that, therefore not to repeat here.
And include in loudspeaker module and external audio output device side, both of which：Demodulating unit, to being configured to sub-band
Voice data stream carry out digital demodulation；Low pass filter, voice data stream is reverted to the distribution of audio stream；Synchronous phase
Add device, be added audio stream to revert to audio file, to complete the decoding to audio stream and transcoding, be more easy to play.
Further, audio output system also includes：
Compression module, in the processor, it is connected with decoder module, wherein compression module includes：Conversion unit, adopt
The waveform of audio stream described in sample, the amplitude sample layering of quantization audio stream, is translated into digital code；Reconfiguration unit, reconstruct
The digital code, to compress audio stream.
Mobile terminal can be implemented in a variety of manners.For example, the terminal described in the present invention can include such as moving
Phone, smart phone, notebook computer, PDA (personal digital assistant), PAD (tablet personal computer), PMP (put by portable multimedia broadcasting
Device), the fixed terminal of the mobile terminal of guider etc. and such as digital TV, desktop computer etc..Hereinafter it is assumed that eventually
End is mobile terminal.However, it will be understood by those skilled in the art that in addition to being used in particular for moving the element of purpose, root
The terminal of fixed type is can also apply to according to the construction of embodiments of the present invention.
It should be noted that embodiments of the invention have preferable implementation, and not the present invention is made any type of
Limitation, any one skilled in the art change or are modified to possibly also with the technology contents of the disclosure above equivalent effective
Embodiment, as long as without departing from the content of technical solution of the present invention, above example is made according to technical spirit of the invention
Any modification or equivalent variations and modification, in the range of still falling within technical solution of the present invention.
Claims (8)
1. a kind of audio-frequency inputting method based on intelligent terminal, it is characterised in that comprise the following steps：
An OTG lines are obtained, are connected to the connector of the intelligent terminal；
An external audio output device is connected to the USB interface of the OTG lines；
In running an audio file in the intelligent terminal, to form an audio stream；
The audio stream is decoded, to resolve at least two voice data streams；
The loudspeaker module that the voice data flow to the intelligent terminal, and the connection by the intelligent terminal are exported respectively
Mouth, OTG lines are exported to the external audio output device；
Control loudspeaker module and external audio output device the playing audio data stream.
2. audio-frequency inputting method as claimed in claim 1, it is characterised in that the audio stream is decoded, to resolve at least two
The step of individual voice data stream, includes：
The audio stream is changed into frequency domain by time-domain；
The audio stream based on the frequency domain, several sub-bands are divided into using bandpass filter.
3. audio-frequency inputting method as claimed in claim 2, it is characterised in that control the loudspeaker module and external audio defeated
The step of going out device playing audio data stream includes：
Digital demodulation is carried out to the voice data stream for being configured to the sub-band；
Using low pass filter, the voice data stream is reverted to the distribution of the audio stream；
By the audio streams to a synchronous adder, the audio file is reverted to by addition.
4. audio-frequency inputting method as claimed in claim 1, it is characterised in that literary in running an audio in the intelligent terminal
Part, to form an audio stream with decoding the audio stream, to resolve into the step of audio output described at least two voice data streams
Also include between rapid：
The waveform of the audio stream is sampled, quantifies the amplitude sample layering of the audio stream, is translated into digital code；
The digital code is reconstructed, to compress the audio stream.
5. a kind of audio output system based on intelligent terminal, it is characterised in that the audio output system includes：
Intelligent terminal, operation has an audio file, to form an audio stream；
OTG lines, it is connected to the connector of the intelligent terminal；
External audio output device, it is connected with the USB interface of the OTG lines；
The intelligent terminal includes processor, and the processor includes：
Decoder module, the audio stream is decoded, to resolve at least two voice data streams；
Output module, the loudspeaker module that the voice data flow to the intelligent terminal is exported, and by the intelligence eventually
Connector, the OTG lines at end are exported to the external audio output device；
Loudspeaker module and external audio output device the playing audio data stream.
6. audio output system as claimed in claim 5, it is characterised in that
The decoder module includes：
Domain conversion unit, the audio stream is changed into frequency domain by time-domain；
Bandpass filter, it is several sub-bands by the finite state Automat based on the frequency domain.
7. audio output system as claimed in claim 6, it is characterised in that
Loudspeaker module and the external audio output device includes：
Demodulating unit, digital demodulation is carried out to the voice data stream for being configured to the sub-band；
Low pass filter, the voice data stream is reverted to the distribution of the audio stream；
Synchronous adder, the audio stream is added to revert to the audio file.
8. audio output system as claimed in claim 5, it is characterised in that the audio output system also includes：
Compression module, in the processor, it is connected with the decoder module, wherein the compression module includes：
Conversion unit, the waveform of the audio stream is sampled, quantify the amplitude sample layering of the audio stream, be translated into numeral
Code；
Reconfiguration unit, the digital code is reconstructed, to compress the audio stream.
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