CN106653028A - Voice communication method and system - Google Patents
Voice communication method and system Download PDFInfo
- Publication number
- CN106653028A CN106653028A CN201611011846.0A CN201611011846A CN106653028A CN 106653028 A CN106653028 A CN 106653028A CN 201611011846 A CN201611011846 A CN 201611011846A CN 106653028 A CN106653028 A CN 106653028A
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- packet
- call terminal
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- oscillogram
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- 238000004891 communication Methods 0.000 title claims abstract description 32
- 238000000034 method Methods 0.000 title claims abstract description 22
- 238000005070 sampling Methods 0.000 claims description 31
- 238000012795 verification Methods 0.000 claims description 30
- 238000013500 data storage Methods 0.000 claims description 5
- 238000011084 recovery Methods 0.000 claims description 5
- 238000006243 chemical reaction Methods 0.000 claims description 4
- 239000000284 extract Substances 0.000 claims description 4
- 230000005540 biological transmission Effects 0.000 abstract description 6
- 238000005516 engineering process Methods 0.000 description 4
- 229910002056 binary alloy Inorganic materials 0.000 description 2
- 238000012546 transfer Methods 0.000 description 2
- 230000002159 abnormal effect Effects 0.000 description 1
- 230000009286 beneficial effect Effects 0.000 description 1
- 230000015572 biosynthetic process Effects 0.000 description 1
- 238000013461 design Methods 0.000 description 1
- 238000001514 detection method Methods 0.000 description 1
- 238000011161 development Methods 0.000 description 1
- 230000000694 effects Effects 0.000 description 1
- 238000005457 optimization Methods 0.000 description 1
- 238000003786 synthesis reaction Methods 0.000 description 1
- 238000012360 testing method Methods 0.000 description 1
Classifications
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L15/00—Speech recognition
- G10L15/26—Speech to text systems
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L15/00—Speech recognition
- G10L15/28—Constructional details of speech recognition systems
- G10L15/30—Distributed recognition, e.g. in client-server systems, for mobile phones or network applications
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/0018—Speech coding using phonetic or linguistical decoding of the source; Reconstruction using text-to-speech synthesis
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/16—Vocoder architecture
- G10L19/167—Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/26—Devices for calling a subscriber
- H04M1/27—Devices whereby a plurality of signals may be stored simultaneously
- H04M1/271—Devices whereby a plurality of signals may be stored simultaneously controlled by voice recognition
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/487—Arrangements for providing information services, e.g. recorded voice services or time announcements
- H04M3/493—Interactive information services, e.g. directory enquiries ; Arrangements therefor, e.g. interactive voice response [IVR] systems or voice portals
- H04M3/4936—Speech interaction details
Landscapes
- Engineering & Computer Science (AREA)
- Human Computer Interaction (AREA)
- Signal Processing (AREA)
- Computational Linguistics (AREA)
- Health & Medical Sciences (AREA)
- Audiology, Speech & Language Pathology (AREA)
- Physics & Mathematics (AREA)
- Acoustics & Sound (AREA)
- Multimedia (AREA)
- Data Exchanges In Wide-Area Networks (AREA)
Abstract
The invention provides a voice communication method and system. The method includes that a first communication terminal sends voice packages to a server; the server converts the voice packages to words; encoding the words and thus obtaining data packages; sending the data packages to a second communication terminal; restoring the data packages to voice by the second communication terminal. The transmission data volume is reduced substantially after voice packages sent by the first terminal are converted to words through the server. The words are encoded and then sent to the second communication terminal. The second terminal restores the received data to voice. Voice transmission can be realized even in a condition with poor communication quality.
Description
Technical field
The present invention relates to voice transmission technology field, more particularly to a kind of audio communication method and system.
Background technology
With the continuous development of the communication technology, voice call becomes the main way that people remotely link up.But adopt language
When sound is conversed, if other side's speech signal is unstable, it is easy to voice packet packet loss phenomenon occur, causes call abnormal, user listens
Unclear sound, affects the experience of user.
The Chinese patent of the A of Publication No. CN 102883392 proposes a kind of multi-mode and multi-standby mobile terminal, including detection list
Unit, detects the parameter value of the service quality of current talking network；
Judging unit, is connected to the detector unit, judges the parameter value of the service quality of the current talking network and is
It is no less than or equal to threshold value, when the parameter value of the service quality is less than or equal to the threshold value, to communication unit control letter is sent
Number；The communication unit, when receiving from the control signal of the judging unit, by the multi-mode and multi-standby mobile terminal
The information of backup network is sent to domain server, and sets up standby connection with another call terminal by the domain server.
Such scheme is prevented from voice interruption when speech path network is of poor quality, but it needs to use standby net
Network, is also accomplished by building backup network in addition, increased call cost.
The content of the invention
The technical problem to be solved is：There is provided one kind can be in the case where speech signal be unstable, it is ensured that
Voice call is normally carried out, and the audio communication method and system of low cost.
In order to solve above-mentioned technical problem, the technical solution used in the present invention is：
A kind of audio communication method, including：
First call terminal sends voice packet to server；
The voice packet is converted into word by server；
Packet is obtained to the literal code；
The packet is sent to the second call terminal；
The packet is reduced into voice by the second call terminal.
The present invention provide another technical scheme be：
A kind of audio communication system, including：First call terminal, server and the second call terminal；
First call terminal includes the first sending module, for voice packet to be sent to server；
The server includes modular converter, for the voice packet to be converted into into word；
Coding module, for obtaining packet to the literal code；
Second sending module, for the packet to be sent to the second call terminal；
Second call terminal includes recovery module, for the packet to be reduced into into voice.
The beneficial effects of the present invention is：The voice packet that first terminal sends is greatly reduced Jing after server is converted into word
The data volume of transmission, sends to the second call terminal after encoding to word, the data convert that second terminal will be received again
Into voice.Voice transfer also can be realized in the case of speech quality difference.
Description of the drawings
Fig. 1 is the flow chart of the audio communication method of the embodiment of the present invention；
Fig. 2 is the structural representation of the audio communication system of the embodiment of the present invention.
Label declaration：
1st, the first call terminal；11st, the first sending module；12nd, the first sampling module；13rd, memory module；2nd, service end；
21st, modular converter；22nd, coding module；23rd, the second sending module；3rd, the second call terminal；31st, recovery module；32nd, second adopt
Egf block；33rd, comparing module；34th, the 3rd sending module.
Specific embodiment
To describe the technology contents of the present invention in detail, being realized purpose and effect, below in conjunction with embodiment and coordinate attached
Figure is explained.
The design of most critical of the present invention is：First terminal sends voice packet to server, and server turns voice packet
Change word into and encoded.
Fig. 1 is refer to, the present invention is provided：
A kind of audio communication method, including：
First call terminal sends voice packet to server；
The voice packet is converted into word by server；
Packet is obtained to the literal code；
The packet is sent to the second call terminal；
The packet is reduced into voice by the second call terminal.
Further, the first call terminal sends voice packet to before server, further includes：
First call terminal is sampled to sent the waveform of voice packet；
It is verification bag by the data storage of sampling；
The verification bag and voice packet are sent to the second call terminal in the form of oscillogram；
Second call terminal is sampled to the oscillogram for receiving；
The data sampled to the oscillogram are compared with the data in the verification bag；
Comparison result is sent to the first call terminal；
If comparison result be the corresponding sampling of the oscillogram data and verification bag in data difference reach it is default
Value, then the first call terminal sends voice packet to server.
Knowable to foregoing description, send first speech quality is analyzed before voice packet, speech quality difference is then adopted will
Voice is converted into the mode of word and sends, and speech quality is well then sent using normal sending method.Specifically, by comparing transmission
It is good that the waveform sample values that the waveform sample values and receiving terminal (second terminal) that end (first terminal) sends are received carry out speech quality
Bad judgement, it would however also be possible to employ other modes are analyzing speech quality.
Further, the data of the sampling include the sampled peak in Preset Time, sampling average and fluctuation number of times.
Knowable to foregoing description, peak value, average and fluctuation number of times are three and can more effectively reflect speech quality
Parameter.
Further, the difference of the data in the data of the corresponding sampling of the oscillogram and verification bag reaches preset value,
Specially：The difference of any one in the oscillogram and the corresponding sampled peak of verification bag, sampling average and fluctuation number of times
Reach preset value.
Knowable to foregoing description, generally above-mentioned preset value is 50%.Because application scenarios are different, some scenes need accurate
Realize voice call, some scenes then need to reduce time delay as far as possible, thus can also arrange other conditions come judge converse matter
Amount quality, such as respectively compares the difference of sampled peak, sampling average and each of fluctuation time with a threshold value, if respectively reaching threshold
Value, then illustrate that speech quality is poor.
Further, it is described that the literal code is obtained after packet, the packet is sent to the second call
Before terminal, further include：
The packet is verified using Hamming checking code.
Knowable to foregoing description, carry out verifying the accuracy rate that can improve verification using Hamming checking code.
Further, the described packet is sent to the second call terminal is specially：
Addition mark is to the packet；
The packet that with the addition of mark is sent to the second call terminal in the form of oscillogram.
Further, the packet is reduced into voice by second call terminal, is further included：
Second call terminal extracts the packet according to the mark from the oscillogram for receiving；
The packet is reduced into into word；
By the text conversion for restoring into voice.
Knowable to foregoing description, second terminal is received after oscillogram, can rapidly be positioned to carrying out text according to mark
The voice of word conversion, improves reduction rate, reduces time delay.
Further, the word is encoded by the way of UTF-8 and obtains the packet.
Fig. 2 is refer to, another technical scheme of the present invention is：
A kind of audio communication system, including：First call terminal 1, the call terminal 3 of server 2 and second；
First call terminal 1 includes the first sending module 11, for voice packet to be sent to server；
The server 2 includes modular converter 21, for the voice packet to be converted into into word；
Coding module 22, for obtaining packet to the literal code；
Second sending module 23, for the packet to be sent to the second call terminal；
Second call terminal 3 includes recovery module 31, for the packet to be reduced into into voice.
Further, first call terminal 1 also includes：
First sampling module 12, is sampled for waveform of first call terminal to sent voice packet；
Memory module 13, for the data storage of the sampling to be wrapped for verification；
First sending module 11 is additionally operable to send the verification bag and voice packet in the form of oscillogram to second
Call terminal；
Second call terminal 3 also includes：
Second sampling module 32, for sampling to the oscillogram for receiving；
Comparing module 33, for the data sampled to the oscillogram to be compared with the data in the verification bag；
3rd sending module 34, for comparison result to be sent to the first call terminal；
If comparison result be the corresponding sampling of the oscillogram data and verification bag in data difference reach it is default
Value, then into first sending module 11.
Embodiments of the invention one are：
A kind of audio communication method, including：
First call terminal is sampled to sent the waveform of voice packet；
Peak value, average and the fluctuation number of times that the data of sampling are sampled in Preset Time is stored as verification bag；
To be sent in the form of oscillogram to the second call terminal with voice packet after the verification bag addition mark；
Second call terminal is sampled to the oscillogram for receiving；
By the peak value sampled in Preset Time to the data that the oscillogram is sampled, average and fluctuation number of times and the school
The data tested in bag are compared；
Comparison result is sent to the first call terminal；
If comparison result be the corresponding sampling of the oscillogram data and verification bag in data difference reach it is default
Value, then the first call terminal sends voice packet to server.
The voice packet is converted into word by server；
The word is encoded by the way of UTF-8 and obtains packet；
The packet is verified using Hamming checking code；
Addition mark is to the packet；
The packet that with the addition of mark is sent to the second call terminal in the form of oscillogram.
Second call terminal extracts the packet according to the mark from the oscillogram for receiving；
The packet is reduced into into word；
By the text conversion for restoring into voice.
Specifically, the scheme of above-described embodiment one is illustrated with an example：
Mobile phone A is sampled to sent the waveform of voice packet, calculates peak value a1, average a2 and the ripple sampled in 1 second
Dynamic number of times a3, by sampled peak a1, average a2 and fluctuation number of times a3 the storage of 20 bytes, as verification bag are converted into, by verification bag
It is put in oscillogram to be sent, by voice channel transmission to mobile phone B in the form of the markd waveform of band；Mobile phone B is docked
The oscillogram of receipts is sampled, calculate 1 second in sample peak value b1, average b2 and fluctuation number of times b3, by peak value a1, average a2,
Fluctuation number of times a3 compares respectively with peak value b1, average b2, fluctuation number of times b3, if the difference of one of which reaches 50%
(difference such as peak value a1 and peak value b1 is 60%), then illustrate dtr signal, enables following method optimization tonequality：
By voice packet by WiFi channel transfers to speech recognition server, speech recognition server turns voice to mobile phone A
For word, then word is encoded to into binary system byte according to UTF-8, the binary system byte is adopted into Hamming check code check；School
Test rear output waveform figure M；
Mobile phone B is transferred to by voice channel after oscillogram M is marked, mobile phone B is recognized with markd waveform, the Duan Bo
Shape is oscillogram M, extracts the literal code packet of this section of waveform, restores the word of UTF-8 codings, then by TTS
Phonetic synthesis API, restores voice and plays out.
Fig. 2 is refer to, embodiments of the invention two are：
A kind of system corresponding with the audio communication method of embodiment one, including：First call terminal 1, service end 2 and
Two call terminals 3；
First call terminal 1 includes：
First sampling module 12, is sampled for waveform of first call terminal to sent voice packet；
Memory module 13, for the data storage of the sampling to be wrapped for verification；
First sending module 11, for the verification bag and voice packet to be sent to the second call eventually in the form of oscillogram
End；
Second call terminal 3 includes：
Second sampling module 32, for sampling to the oscillogram for receiving；
Comparing module 33, for the data sampled to the oscillogram to be compared with the data in the verification bag
It is right；
3rd sending module 34, for comparison result to be sent to the first call terminal；
If comparison result be the corresponding sampling of the oscillogram data and verification bag in data difference reach it is default
Value, then into first sending module 11, first sending module 11 sends voice packet to server；
The server 2 includes modular converter 21, for the voice packet to be converted into into word；
Coding module 22, for obtaining packet to the literal code；
Second sending module 23, for the packet to be sent to the second call terminal；
Second call terminal 3 also includes recovery module 31, for the packet to be reduced into into voice.
In sum, the present invention is provided audio communication method and system, can be in the feelings of distant terminal network signal difference
Voice is sent to distant terminal under condition.
Embodiments of the invention are the foregoing is only, the scope of the claims of the present invention is not thereby limited, it is every using this
The equivalents that bright specification and accompanying drawing content are made, or the technical field of correlation is directly or indirectly used in, include in the same manner
In the scope of patent protection of the present invention.
Claims (10)
1. a kind of audio communication method, it is characterised in that include：
First call terminal sends voice packet to server；
The voice packet is converted into word by server；
Packet is obtained to the literal code；
The packet is sent to the second call terminal；
The packet is reduced into voice by the second call terminal.
2. audio communication method according to claim 1, it is characterised in that the first call terminal sends voice packet to clothes
Before business device, further include：
First call terminal is sampled to sent the waveform of voice packet；
It is verification bag by the data storage of sampling；
The verification bag and voice packet are sent to the second call terminal in the form of oscillogram；
Second call terminal is sampled to the oscillogram for receiving；
The data sampled to the oscillogram are compared with the data in the verification bag；
Comparison result is sent to the first call terminal；
If comparison result is the difference of the data in the data of the corresponding sampling of the oscillogram and verification bag reaches preset value,
First call terminal sends voice packet to server.
3. audio communication method according to claim 2, it is characterised in that the data of the sampling are included in Preset Time
Sampled peak, sampling average and fluctuation number of times.
4. audio communication method according to claim 3, it is characterised in that the data of the corresponding sampling of the oscillogram and
The difference of the data in verification bag reaches preset value, specially：The oscillogram and the corresponding sampled peak of verification bag, sampling are
The difference of any one in value and fluctuation number of times reaches preset value.
5. audio communication method according to claim 1, it is characterised in that described that packet is obtained to the literal code
Afterwards, the packet is sent to before the second call terminal, is further included：
The packet is verified using Hamming checking code.
6. audio communication method according to claim 1, it is characterised in that the described packet is sent to second is led to
Telephone terminal is specially：
Addition mark is to the packet；
The packet that with the addition of mark is sent to the second call terminal in the form of oscillogram.
7. audio communication method according to claim 6, it is characterised in that second call terminal is by the packet
Voice is reduced into, is further included：
Second call terminal extracts the packet according to the mark from the oscillogram for receiving；
The packet is reduced into into word；
By the text conversion for restoring into voice.
8. audio communication method according to claim 1, it is characterised in that the word is compiled by the way of UTF-8
Code obtains the packet.
9. a kind of audio communication system, it is characterised in that include：First call terminal, server and the second call terminal；
First call terminal includes the first sending module, for voice packet to be sent to server；
The server includes modular converter, for the voice packet to be converted into into word；
Coding module, for obtaining packet to the literal code；
Second sending module, for the packet to be sent to the second call terminal；
Second call terminal includes recovery module, for the packet to be reduced into into voice.
10. audio communication system according to claim 9, it is characterised in that first call terminal also includes：
First sampling module, is sampled for waveform of first call terminal to sent voice packet；
Memory module, for the data storage of the sampling to be wrapped for verification；
First sending module is additionally operable to send the verification bag and voice packet in the form of oscillogram to the second call eventually
End；
Second call terminal also includes：
Second sampling module, for sampling to the oscillogram for receiving；
Comparing module, for the data sampled to the oscillogram to be compared with the data in the verification bag；
3rd sending module, for comparison result to be sent to the first call terminal；
If comparison result is the difference of the data in the data of the corresponding sampling of the oscillogram and verification bag reaches preset value,
Into first sending module.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201611011846.0A CN106653028A (en) | 2016-11-17 | 2016-11-17 | Voice communication method and system |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201611011846.0A CN106653028A (en) | 2016-11-17 | 2016-11-17 | Voice communication method and system |
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| CN106653028A true CN106653028A (en) | 2017-05-10 |
Family
ID=58807567
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| CN201611011846.0A Pending CN106653028A (en) | 2016-11-17 | 2016-11-17 | Voice communication method and system |
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Cited By (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN111199747A (en) * | 2020-03-05 | 2020-05-26 | 北京花兰德科技咨询服务有限公司 | Artificial intelligence communication system and communication method |
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2016
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| CN101415193A (en) * | 2008-11-17 | 2009-04-22 | 武汉虹信通信技术有限责任公司 | Method for evaluating mobile wireless network voice quality through oscillogram for road measurement system |
| CN102710539A (en) * | 2012-05-02 | 2012-10-03 | 中兴通讯股份有限公司 | Method and device for transferring voice messages |
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Application publication date: 20170510 |