CN206293160U - A kind of speech processing system based on digital DSP algorithm - Google Patents
A kind of speech processing system based on digital DSP algorithm Download PDFInfo
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- CN206293160U CN206293160U CN201621368484.6U CN201621368484U CN206293160U CN 206293160 U CN206293160 U CN 206293160U CN 201621368484 U CN201621368484 U CN 201621368484U CN 206293160 U CN206293160 U CN 206293160U
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Abstract
The utility model provides a kind of speech processing system based on digital DSP algorithm, including：Audio frequency collecting device, AD conversion unit, DSP digital processing units, D/A conversion unit, signal amplification unit, audio processing modules；Audio frequency collecting device is connected with AD conversion unit, and AD conversion unit is connected by DSP digital processing units with audio processing modules, and D/A conversion unit is connected to the output end of audio processing modules, and signal amplification unit is connected with D/A conversion unit.The utility model solves sound feedback and utters long and high-pitched sounds in real time, improves output audio quality.
Description
Technical field
The utility model is related to speech processing device field, and in particular to a kind of speech processes based on digital DSP algorithm
System.
Background technology
At present in the application of multi-media voice processing system, conventional sound feedback processing mode of uttering long and high-pitched sounds has two kinds：1、
Wave filter；2nd, frequency shifter.Main audio frequency process is filtering, shift frequency, and the feedback that decayed by wave filter is uttered long and high-pitched sounds frequency, frequency shifter
Uttered long and high-pitched sounds a little come feedback of staggering, the generation for preventing sound feedback from uttering long and high-pitched sounds by both technologies.But by both mode pair
The infringement of sound signal is very huge, situations such as cause audio distortions, the demand that influence user collects to high definition sound signal.
Simultaneously site environment ambient noise cannot be filtered in traditional multi-media voice system so that use environment on a large scale
When using, often being influenceed the reduction degree of sound by ambient noise.And the equipment that system is mainly by difference in functionality is stacked
Form, system compatibility, scalability are relatively poor, while also bringing quite big to user in system debug and regular maintenance
A part of cost.
It is not difficult to find out, prior art also has certain defect.
Utility model content
Technical problem to be solved in the utility model is to provide a kind of speech processing system based on digital DSP algorithm,
Sound feedback is solved in real time to utter long and high-pitched sounds, and improves output audio quality.
To achieve the above object, the utility model uses following technical scheme：
A kind of speech processing system based on digital DSP algorithm, including：Audio frequency collecting device, AD conversion unit, DSP
Digital processing unit, D/A conversion unit, signal amplification unit, audio processing modules；Audio frequency collecting device and AD conversion unit
Connection, AD conversion unit is connected by DSP digital processing units with audio processing modules, and D/A conversion unit is connected at audio
The output end of module is managed, signal amplification unit is connected with D/A conversion unit.
Further, the audio processing modules include a data storage cell, at data storage cell and DSP numerals
Connection carries out data interaction between reason device.
Further, the audio processing modules include a wave filter, and wave filter is connected to D/A conversion unit and DSP
Treatment is filtered between digital processing unit.
Further, the audio processing modules include that local side pick up facility, local side sound amplifier, distal end pickup set
Standby and distal end sound amplifier；Local side pick up facility, local side sound amplifier, distal end pick up facility and distal end sound amplifier difference
It is connected with DSP digital processing units, DSP digital processing units is received signal from local side pick up facility and distal end pick up facility, and
To distal end sound amplifier and local side sound amplifier output signal.
A kind of speech processing system based on digital DSP algorithm that the utility model is provided, with advantages below：
Solve the problems, such as that current multimedia-audio systems are complicated, be conducive to operator to use；
User is allowed to make a speech freer, the audio frequency feedback based on DSP self adaptations is uttered long and high-pitched sounds control, system need not arrange many
Microphone carries out voice pickup to spokesman, need to only configure 1-2 and only lift by microphone to pickup on a large scale in environment, makes hair
Speaker breaks away from the constraint of microphone；
Ambient noise is big during solving the problems, such as current multimedia system audio recording, is dropped by DSP adaptive backgrounds
After making an uproar so that the audio signal of recording is apparent；
Eliminated by DSP adaptive echos, solve the Network echo produced in remote audio communication.
Brief description of the drawings
In order to illustrate more clearly of the utility model embodiment or technical scheme of the prior art, below will be to embodiment
Or the accompanying drawing to be used needed for description of the prior art is briefly described, it should be apparent that, drawings in the following description are only
It is some embodiments of the present utility model, for those of ordinary skill in the art, is not paying the premise of creative work
Under, other accompanying drawings can also be obtained according to these accompanying drawings.
A kind of feedback of speech processing system based on digital DSP algorithm that Fig. 1 is provided for the utility model embodiment is maked a whistling sound
It is processing framework schematic diagram.
A kind of background drop of speech processing system based on digital DSP algorithm that Fig. 2 is provided for the utility model embodiment
Processing framework of making an uproar schematic diagram.
A kind of elimination net of speech processing system based on digital DSP algorithm that Fig. 3 is provided for the utility model embodiment
Network echo configuration diagram.
Specific embodiment
It is new below in conjunction with this practicality to make the purpose, technical scheme and advantage of the utility model embodiment clearer
Type embodiment and accompanying drawing, are clearly and completely described to the technical scheme in the utility model embodiment.Need explanation
It is that described embodiment is only a part of embodiment of the utility model, rather than whole embodiments.It is new based on this practicality
Embodiment in type, the every other implementation that those of ordinary skill in the art are obtained under the premise of creative work is not made
Example, belongs to the scope of the utility model protection.
Embodiment
Fig. 1 to Fig. 3 is referred to, present embodiment discloses a kind of speech processing system based on digital DSP algorithm, including：
Audio frequency collecting device, AD conversion unit, DSP digital processing units, D/A conversion unit, signal amplification unit, audio frequency process mould
Block；Audio frequency collecting device is connected with AD conversion unit, and AD conversion unit passes through DSP digital processing units and audio processing modules
Connection, D/A conversion unit is connected to the output end of audio processing modules, and signal amplification unit is connected with D/A conversion unit.
The utility model is based on DSP digital processing units, solves the problems, such as that current multimedia-audio systems are complicated.Current many matchmakers
Body audio system is more by sound console, microphone, feedback suppressor, Echo Canceller, denoiser, audio process etc..This is caused
System is installed, debugging, using and safeguard all sufficiently complex, be unfavorable for that operator uses, and in system debug and regular maintenance
In also bring sizable a part of cost to user.
Specifically, the compatible feedback of the system is uttered long and high-pitched sounds, background noise reduction, eliminates Network echo three zones.
Preferably, the audio processing modules include a data storage cell, at data storage cell and DSP numerals
Connection carries out data interaction between reason device.
Initial audio signal sample is carried out to live sound source by audio frequency collecting device, after the completion of system to simulate sound
Frequency signal is sampled, and realizes analog-to-digital conversion, and simulated audio signal is converted into digital audio and video signals.Data signal after conversion
Available digital audio and video signals are transferred to data storage cell and do feedback sound by feeding DSP digital processing units, DSP digital processing units
Contrast is used, and digital audio and video signals are delivered to next stage D/A conversion unit and processed by another aspect DSP digital processing units, number
After digital audio and video signals are converted into simulated audio signal by mould converting unit, then rear class signal amplification unit is transferred to analogue audio frequency
Signal is amplified output.Voice signal output after can be acquired by audio frequency collecting device again, when the signal for collecting again
During secondary arrival DSP digital processing units, it is right that the former sound data signal in the automatic called data memory cell of DSP digital processing units is carried out
Than contrast identical frequency is no longer exported, and so entirely the feedback that is produced by sound feedback is uttered long and high-pitched sounds phenomenon for cut-out.
Preferably, the audio processing modules include a wave filter, wave filter is connected to D/A conversion unit and DSP
Treatment is filtered between digital processing unit.
The audio signal that audio frequency collecting device is collected is sampled, by AD conversion unit by simulated audio signal
After being converted to digital audio and video signals, the intensity of signal and the frequency spectrum of signal are analyzed by DSP digital processing units.Work as someone
During speech, do the analysis of signal in real time, so we just can analysis background noise and spokesman intensity and spectrum distribution, then
According to this analysis result, the two signals are contrasted in real time by wave filter, allow talker's sound spectrum to pass through, allowed
The frequency spectrum of ambient noise is filtered or reduced, and thus reaches the effect of background noise reduction.Finally by D/A conversion unit by numeral
Audio signal is converted to simulated audio signal, then exports signal amplification unit and carry out signal and amplify output.
Preferably, the audio processing modules include that local side pick up facility, local side sound amplifier, distal end pickup set
Standby and distal end sound amplifier；Local side pick up facility, local side sound amplifier, distal end pick up facility and distal end sound amplifier difference
It is connected with DSP digital processing units, DSP digital processing units is received signal from local side pick up facility and distal end pick up facility, and
To distal end sound amplifier and local side sound amplifier output signal.
The sound signal that the distal end picked up by distal end pick up facility is come, being transferred to local side sound amplifier carries out public address
Before, can be to a reference signal to DSP digital processing units.DSP digital processing units utilize this reference signal, it is possible to set up
One signal model for being used to eliminate echo., when local side sound amplifier carries out public address, voice signal can be in wall for remote signaling
Wall, ceiling, carry out on floor after multiple reflections and picked up by local side pick up facility.Pick up facility picks up these reflected sounds
Afterwards, this echo signal is sent into DSP digital processing units, by after calculating process, by this echo signal and the ginseng collected before this
Examine signal to be contrasted, so as to filter and reduce echo signal.So the pronunciation primary sound of local side pick up facility pickup is with regard to energy
The interference of echo is enough avoided, passing to distal end sound amplifier carries out public address.
It according to user is with reference to comparatively that " distal end " and " local side " mentioned hereinabove is.The mistake of echo cancellor
For the pronunciation primary sound of distal end, principle is also the same to journey.
A kind of speech processing system based on digital DSP algorithm provided by the utility model, solves current multimedia sound
The complicated problem of display system.Current multimedia-audio systems are more by sound console, microphone, feedback suppressor, Echo Canceller, noise reduction
Device, audio process etc. so that system install, debugging, using and safeguard it is all sufficiently complex so that system install, debugging, should
With and safeguard it is all sufficiently complex, be unfavorable for that operator uses.It is freer that the system allows user to make a speech, based on DSP digital processings
The audio frequency feedback that device carries out self adaptation is uttered long and high-pitched sounds control, and system need not arrange that many microphones carry out voice pickup to spokesman,
1-2 need to only be configured only to lift by microphone to pickup on a large scale in environment, spokesman is broken away from the constraint of microphone.Solve simultaneously
The big problem of ambient noise during certainly current multimedia system audio recording, it is necessary to audio in meeting and education activities
When information is preserved, the audio-frequency information ambient noise for often recording out is excessive, it is impossible to clearly preserve conference content.Pass through
After DSP digital processing units carry out background noise reduction, automatic ambient noise can be filtered and be not output so that the audio letter of recording
It is number apparent, improve and record quality.The system carries out adaptive echo elimination by DSP digital processing units, solves remote audio
The Network echo produced in communication.
Embodiment described above only expresses a kind of implementation method of the present utility model, and its description is more specific and detailed,
But therefore can not be interpreted as the limitation to the utility model the scope of the claims.It should be pointed out that common for this area
For technical staff, without departing from the concept of the premise utility, various modifications and improvements can be made, these all belong to
In protection domain of the present utility model.Therefore, the protection domain of the utility model patent should be determined by the appended claims.
Claims (4)
1. a kind of speech processing system based on digital DSP algorithm, it is characterised in that including：Audio frequency collecting device, analog-to-digital conversion
Unit, DSP digital processing units, D/A conversion unit, signal amplification unit, audio processing modules；Audio frequency collecting device and modulus
Converting unit is connected, and AD conversion unit is connected by DSP digital processing units with audio processing modules, D/A conversion unit connection
In the output end of audio processing modules, signal amplification unit is connected with D/A conversion unit.
2. the speech processing system based on digital DSP algorithm according to claim 1, it is characterised in that：At the audio
Reason module includes a data storage cell, and connection carries out data interaction between data storage cell and DSP digital processing units.
3. the speech processing system based on digital DSP algorithm according to claim 1, it is characterised in that：At the audio
Reason module includes a wave filter, and wave filter is connected to and treatment is filtered between D/A conversion unit and DSP digital processing units.
4. the speech processing system based on digital DSP algorithm according to claim 1, it is characterised in that：At the audio
Reason module includes local side pick up facility, local side sound amplifier, distal end pick up facility and distal end sound amplifier；Local side pickup
Equipment, local side sound amplifier, distal end pick up facility and distal end sound amplifier are connected with DSP digital processing units respectively, make DSP numbers
Word processing device receives signal from local side pick up facility and distal end pick up facility, and distal end sound amplifier and local side public address are set
Standby output signal.
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| Application Number | Priority Date | Filing Date | Title |
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| CN201621368484.6U CN206293160U (en) | 2016-12-14 | 2016-12-14 | A kind of speech processing system based on digital DSP algorithm |
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| Application Number | Priority Date | Filing Date | Title |
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| CN201621368484.6U CN206293160U (en) | 2016-12-14 | 2016-12-14 | A kind of speech processing system based on digital DSP algorithm |
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Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN109040884A (en) * | 2018-08-31 | 2018-12-18 | 上海联影医疗科技有限公司 | Voice system based on Medical Devices |
| CN111586527A (en) * | 2020-04-28 | 2020-08-25 | 重庆西原楼宇自动化工程有限公司 | Intelligent voice processing system |
| CN112764535A (en) * | 2021-01-08 | 2021-05-07 | 温州职业技术学院 | System for realizing multi-language information exchange |
-
2016
- 2016-12-14 CN CN201621368484.6U patent/CN206293160U/en active Active
Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN109040884A (en) * | 2018-08-31 | 2018-12-18 | 上海联影医疗科技有限公司 | Voice system based on Medical Devices |
| CN111586527A (en) * | 2020-04-28 | 2020-08-25 | 重庆西原楼宇自动化工程有限公司 | Intelligent voice processing system |
| CN112764535A (en) * | 2021-01-08 | 2021-05-07 | 温州职业技术学院 | System for realizing multi-language information exchange |
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