CN103905928A - Network voice intercom method, device and system - Google Patents
Network voice intercom method, device and system Download PDFInfo
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Abstract
The invention discloses a network voice intercom method, device and system. The method comprises the following steps: data synchronization is performed on an audio stream recorded locally and an audio stream sent by the opposite end; the echo in the audio stream recorded locally is filtered by taking the audio stream sent by the opposite end as the reference audio stream; and the filtered audio stream is encoded and compressed and sent through a network. The method, device and system of the embodiments of the invention are used to perform data synchronization on the audio stream recorded locally and the audio stream sent by the opposite end and take the audio stream sent by the opposite end as the reference audio for filtering the echo, the encoding and decoding can be optimized, the echo can be eliminated, and the network voice intercom call quality can be improved.
Description
Technical field
The present invention relates to communication technical field, relate in particular to a kind of voice-over-net intercommunication method, Apparatus and system.
Background technology
Along with scientific and technical development, people are more and more higher to the requirement of life, contact is each other frequent especially, be accompanied by the extensive use of Android system and 3G network, voice-over-net call on Android system is more and more general, people can carry out normal speech talkback by Ethernet, and free, convenient, clear, smooth phone system is a direction of current voice-over-net call development.On Android system, there are some more ripe application at present, can meet intercommunication requirement, but substantially all charge, and seldom can accomplish in real time, noiseless, smooth call, it is not to be satisfied with very much that user experiences, can meet generally processing by hardware of these requirements, but this mode is not general, has increased cost yet.
Voice-over-net intercom system can be carried out normal speech talkback by Ethernet between two or multiple devices.Voice-over-net intercom system is divided into transmitting terminal and receiving terminal, and at transmitting terminal, audio recording module transfers sound to pcm audio stream by MIC, sends into coding module audio compressed data, finally by network, the data after coding is sent; At receiving terminal, obtain the voice data of coding by receiver module, send into decoder module, reduction audio stream data, finally sends into playing module the audio stream receiving is played back.In actual use procedure, voice-over-net is to lecture's echogenicity, and has time delay, causes speech quality not high.
Summary of the invention
In view of this, the technical problem to be solved in the present invention is to provide a kind of voice-over-net intercommunication method, Apparatus and system, causes with the echo solving in voice-over-net intercommunication the problem that speech quality is not high.
It is as follows that the present invention solves the problems of the technologies described above adopted technical scheme:
According to an aspect of the present invention, a kind of voice-over-net intercommunication method providing, comprises the following steps:
It is synchronous that the audio stream that the audio stream that this locality is recorded and opposite end send carries out data;
The echo of the audio stream that opposite end is sent in the audio stream of this locality being recorded with reference to audio stream filtered;
Audio stream after filtering is carried out compression coding and sent by network.
Preferably, the audio stream that the audio stream that this locality is recorded and opposite end send carries out the synchronous step of data, specifically comprises:
The audio stream that is provided for this locality to record carries out the buffering area of recording of buffer memory, and reference audio is flow to the reference buffer district of row cache; In the time that reference buffer district receives data, to the local audio stream of recording and reference audio to flow to row data synchronous.
Further, record the audio stream data that can record N frame buffering area and carry out buffer memory, reference buffer district can carry out buffer memory to the data of M frame reference audio stream; Wherein M>N, and M and N are natural number.
Preferably, the step that the echo of the audio stream that opposite end is sent in the audio stream of this locality being recorded with reference to audio stream filtered, specifically comprises:
According to the time period of default fixed intervals, check the frame data of audio stream that opposite end sends, in the time that frame data are less than the predetermined buffer value of speex algorithm, the echo in the audio stream of this locality being recorded by speex algorithm is filtered; When the frame data of audio stream that send when opposite end are greater than the predetermined buffer value of speex algorithm, discarded part downlink data upon handover.
Preferably, the audio stream after filtering being carried out to compression coding comprises: the audio stream after filtering is carried out to buffer memory and compression coding frame by frame.
According to another aspect of the present invention, a kind of voice-over-net talkback unit providing, this device comprises synchronization module, echo cancellation module, audio coding module and sending module, wherein: synchronization module, it is synchronous that the audio stream sending for audio stream that this locality is recorded and opposite end carries out data; Echo cancellation module, the echo of the audio stream that opposite end is sent in the audio stream of this locality being recorded with reference to audio stream filtered; Audio coding module, for carrying out compression coding to the audio stream after filtering; Sending module, for sending the audio stream of compression coding by network.
Preferably, synchronization module comprises records buffering area, reference buffer district and data synchronisation unit, wherein: record buffering area, carry out buffer memory for the audio stream that this locality is recorded; Reference buffer district, for flowing to row cache to reference audio; Data synchronisation unit, in the time that reference buffer district receives data, it is synchronous that the audio stream that this locality is recorded and reference audio flow to row data.
Preferably, echo cancellation module specifically for: according to the time period of default fixed intervals, check the frame data of audio stream that opposite end sends, in the time that frame data are less than the predetermined buffer value of speex algorithm, the echo in the audio stream of this locality being recorded by speex algorithm is filtered; When the frame data of audio stream that send when opposite end are greater than the predetermined buffer value of speex algorithm, discarded part downlink data upon handover.
Preferably, audio coding module specifically for: to filter after audio stream carry out frame by frame buffer memory and compression coding.
According to a further aspect of the invention, a kind of voice-over-net intercom system providing, this system comprises receiving system and above-mentioned voice-over-net talkback unit, receiving system comprises receiver module, audio decoder module and audio playing module, wherein: receiver module, for the audio stream after received code compression; Audio decoder module, for the audio stream reducing after the compression coding receiving; Audio playing module, for playing the audio stream of reduction.
The methods, devices and systems of the embodiment of the present invention, it is synchronous that the audio stream that the audio stream that this locality is recorded and opposite end send carries out data, the audio stream that opposite end is sent, as the reference audio of filtering echo, has been optimized encoding and decoding and has been eliminated echo, has improved the intercommunication speech quality of voice-over-net.
Brief description of the drawings
Fig. 1 is the principle schematic that in correlation technique, echo produces;
Fig. 2 is a kind of voice-over-net intercommunication method flow diagram that the embodiment of the present invention provides.
Fig. 3 is that the embodiment of the present invention is called speex algorithm interface function schematic diagram in application layer;
Fig. 4 is the particular flow sheet of a kind of voice-over-net intercommunication method of providing of the embodiment of the present invention.
Fig. 5 is the modular structure figure of a kind of voice-over-net talkback unit of providing of the embodiment of the present invention.
Fig. 6 is the structural representation of a kind of voice-over-net intercom system of providing of the embodiment of the present invention.
Embodiment
In order to make technical problem to be solved by this invention, technical scheme and beneficial effect clearer, clear, below in conjunction with drawings and Examples, the present invention is further elaborated.Should be appreciated that specific embodiment described herein, only in order to explain the present invention, is not intended to limit the present invention.
In voice-over-net intercommunication, what affect speech quality is mainly echo and time delay.Please participate in Fig. 1, at A end, voice a is by microphone b typing, and loudspeaker c plays B and holds the sound transmitting, a part of sound e that c plays is also by microphone b typing, sound e is exactly echo, and sound e and sound a have been transferred to B end like this, and the sound that loudspeaker j plays has just comprised voice a and sound e, and sound e is transmitted by B end, so the sound that B end is play can be heard the sound e of oneself, in like manner, also can hear the sound h of oneself at A end.Around this principle known, echo cancellation is exactly sound e and the sound h filtering out after microphone records in Fig. 1.The principle producing according to echo, eliminates echo using the opposite end audio stream of playing as with reference to audio frequency, and echo part in the audio stream of microphone records for example, is filtered out by certain algorithm (speex algorithm).If but the audio frequency of reference is incorrect, be just difficult to filtering echo.
Be the method for a kind of voice-over-net intercommunication of providing of the embodiment of the present invention as shown in Figure 2, the method comprises:
It is synchronous that the audio stream that S201, the audio stream that this locality is recorded and opposite end send carries out data;
Preferably, this step can realize in the following manner:
The audio stream that is provided for this locality to record carries out the buffering area of recording of buffer memory, and reference audio is flow to the reference buffer district of row cache; In the time that reference buffer district receives data, to the local audio stream of recording and reference audio to flow to row data synchronous.Record the audio stream data that can record N frame buffering area and carry out buffer memory, reference buffer district can carry out buffer memory to the data of M frame reference audio stream; Wherein M>N, and M and N are natural number.
S202, the echo of audio stream in the audio stream of this locality being recorded with reference to audio stream that opposite end is sent are filtered;
S203, to filter after audio stream carry out compression coding and send by network;
Preferably, this step can be carried out buffer memory and compression coding frame by frame to the audio stream after filtering.Audio stream after compression coding receives and reduces and play for opposite end.
Specifically, the buffering area of recording audio stream and the buffering area of depositing opposite end audio stream are set, while receiving data, carry out data synchronous simultaneously.The buffering area of the audio stream that ought record receives data, and the buffering area of depositing the opposite end audio stream of broadcasting is while receiving data, can think that data are now synchronous, adopts opposite end audio stream now as with reference to audio frequency.In actual applications, when the buffering area of the audio stream of recording receives N frame data, and the buffering area of opposite end audio stream of playing is while receiving N+1 frame data, and effect is better.Wherein, frame data are directly proportional to coded sample rate, for example: if coded sample rate is 8KB, a frame sign is 160 bytes; If sample rate is 16KB, a frame sign is 320 bytes; If sample rate is 32KB, a frame sign is 640 bytes.
Preferably, the echo in embodiment of the present invention elimination can adopt the filtration of certain algorithm, for example speex algorithm.The audio stream of recording due to this locality is uniform and stable, and the speed of the far-end audio that network sends stream is unsettled, can be at interval of regular time section, check the frame data size of the audio stream sending as the opposite end with reference to audio frequency, if be greater than the predetermined buffer value of speex algorithm, abandon part, better to select reference audio; If be less than the predetermined buffer value of speex algorithm, can adopt speex algorithm to carry out echo filter.
Because the coding of audio frequency needs a period of time, according to 8K sampling, every frame 160 bytes, coding needs the time of about 20ms left and right, and audio compressed data is synchronizeed and can be reduced delay with echo cancellation.Can be by the audio stream data after a filtering frames of buffer memory is completed after compression coding, then the mode of audio stream data after the new filtering frames of buffer memory reduces delay.
Below with based on android platform, adopt the speex algorithm of increasing income to do echo cancellation and audio coding decoding is further set forth the present invention.Speex algorithm is the algorithm of the special encoding and decoding for audio frequency, and this algorithm, also in continuous renewal, has had echo cancellation module now, and through facts have proved, speex can realize preferably audio coding decoding and eliminate echo.
Refer to Fig. 3, java application layer is obtained the method for speex built-in function, because speex algorithm is c language compilation, to call and just speex algorithm need to be made to JNI library file in java application layer, as shown in the figure, first from speex, extract interface function and be packaged into JNI function, echo cancellation is mainly speex_echo_cancellation function, the sound that this function is recorded and play by contrast, from the sound of recording, the sound of (echo) is play in filtering; Encoding and decoding are mainly speex_bits_write compression function and speex_bits_read_from decompression function.
Each layer of encapsulation function refers to table 1, and JNI package file is compiled into .so library file, loads this storehouse in application layer, package application interface function, and in application layer call list 1, java application layer encapsulation function carries out echo cancellation and audio coding decoding.
Each layer of encapsulation function in the speex algorithm that table 1 is increased income
Refer to Fig. 4, the particular flow sheet of embodiment of the present invention voice-over-net intercommunication.Network receives and sends operation simultaneously.At transmitting terminal, by audio recording module from microphone records audio stream, be that a frame is stored in and records buffering area according to predetermined buffer value, open echo cancellation thread lock, audio stream is sent in echo cancellation thread, in echo cancellation thread, receive audio stream that far-end transmits as the reference voice that filters echo simultaneously, when the buffering area of depositing the audio stream of recording has received data, and when the buffering area of depositing the audio stream of broadcasting has received data, send into filtering echo in echo cancellation SpeexDspAEC function, through test of many times, receive 6 frame data when recording buffering area, and when play buffer receives 7 frame data, eradicating efficacy the best, the clean sound obtaining is sent into coding module, in coding module, because the compression time of audio frequency is long, so in order to prevent data jamming, while receiving new voice data, all guarantee current voice data of encoding encoded completing at every turn, namely receive data and coding and carry out synchronously, the voice data after coding sends by network.
At receiving terminal, receive the voice data that network sends, send decoder module that audio stream is reduced, then send playing module, and a sound part of playing is by typing again, be exactly echo by this part of typing, namely reference voice, deposit in cache list as a frame according to buffer value predetermined in table 2 with reference to sound at every turn, through test of many times, in the time having 7 frame data in list, just start to send into echo cancellation thread best results, ensure to have the reference voice of a frame (predetermined buffer value) at every turn.So repeatedly carry out, realized network talkback.
Predetermined buffer value is as shown in table 2, is the frame data size in the present embodiment, is also the each treatable size of data of echo cancellation and encoding and decoding, and buffer memory is also that a frame carries out according to this size.At transmitting terminal and receiving terminal, all need voice data to carry out buffer memory, in native system, while receiving audio stream size for predetermined buffer value, it is a frame, receive that putting into list after frame data carries out buffer memory, and start to take out from the first frame of list in the time of processing audio data, until the data in list all take.
Table 2
The modular structure figure of a kind of voice-over-net talkback unit of providing of the embodiment of the present invention as shown in Figure 5.Voice-over-net talkback unit 10 comprises synchronization module 110, echo cancellation module 120, coding module 130 and sending module 140.It is synchronous that the audio stream that synchronization module 110 sends for audio stream that this locality is recorded and opposite end carries out data; Echo cancellation module 120 is filtered as the echo of the audio stream of this locality being recorded with reference to audio stream for the audio stream that opposite end is sent; Audio coding module 130 is for carrying out compression coding to the audio stream after filtering; Sending module 140 is for sending the audio stream of compression coding by network.
Voice-over-net talkback unit 10 also comprises records module 100, for recording local audio stream.
Record buffering area 1101, carry out buffer memory for the audio stream that this locality is recorded;
Be the structural representation of a kind of voice-over-net intercom system of providing of the embodiment of the present invention as shown in Figure 6, this system comprises voice-over-net talkback unit 10 and receiving system 20 above.Wherein, receiving system 20 comprises receiver module 210, audio decoder module 220 and audio playing module 230, audio stream after receiver module 210 compresses for received code, the audio stream of audio decoder module 220 for reducing after the compression coding receiving, audio playing module 230 is play the audio stream of reduction.Voice-over-net talkback unit 10 is same as the previously described embodiments, no longer repeats here.
It is synchronous that the embodiment of the present invention is carried out data by the opposite end audio stream to the audio stream of recording and reception, using the opposite end audio stream receiving as the reference audio of filtering echo, optimized encoding and decoding and eliminated echo, improved the intercommunication speech quality of voice-over-net.
With reference to the accompanying drawings of the preferred embodiments of the present invention, not thereby limit to interest field of the present invention above.Those skilled in the art do not depart from the scope and spirit of the present invention, and can have multiple flexible program to realize the present invention, such as can be used for another embodiment and obtain another embodiment as the feature of an embodiment.Allly using any amendment of doing within technical conceive of the present invention, be equal to and replace and improve, all should be within interest field of the present invention.
Claims (10)
1. a method for voice-over-net intercommunication, is characterized in that, the method comprises:
It is synchronous that the audio stream that the audio stream that this locality is recorded and opposite end send carries out data;
The echo of the audio stream that described opposite end is sent in the audio stream of described this locality being recorded with reference to audio stream filtered;
Audio stream after filtering is carried out compression coding and sent by network.
2. method according to claim 1, is characterized in that, the audio stream that the described audio stream that this locality is recorded and opposite end send carries out the synchronous step of data, specifically comprises:
The audio stream that is provided for that described this locality is recorded carries out the buffering area of recording of buffer memory, and described reference audio is flow to the reference buffer district of row cache; In the time that described reference buffer district receives data, to the local audio stream of recording and described reference audio to flow to row data synchronous.
3. method according to claim 2, is characterized in that:
Buffer memory can be carried out to the audio stream data of recording described in N frame in the described buffering area of recording, and described reference buffer district can carry out buffer memory to the data of reference audio stream described in M frame; Wherein M>N, and M and N are natural number.
4. method according to claim 1 and 2, is characterized in that, the step that the echo of the described audio stream that described opposite end is sent in the audio stream of described this locality being recorded with reference to audio stream filtered, specifically comprises:
According to the time period of default fixed intervals, check the frame data of audio stream that opposite end sends, in the time that frame data are less than the predetermined buffer value of speex algorithm, the echo in the audio stream of described this locality being recorded by speex algorithm is filtered; When the frame data of audio stream that send when opposite end are greater than the predetermined buffer value of speex algorithm, discarded part downlink data upon handover.
5. method according to claim 1, is characterized in that, described audio stream after filtering is carried out compression coding and comprised: the audio stream after filtering is carried out to buffer memory and compression coding frame by frame.
6. a voice-over-net talkback unit, is characterized in that, this device comprises synchronization module, echo cancellation module, audio coding module and sending module, wherein:
Synchronization module, it is synchronous that the audio stream sending for audio stream that this locality is recorded and opposite end carries out data;
Echo cancellation module, the echo of the audio stream that described opposite end is sent in the audio stream of described this locality being recorded with reference to audio stream filtered;
Audio coding module, for carrying out compression coding to the audio stream after filtering;
Sending module, for sending the audio stream of compression coding by network.
7. talkback unit according to claim 6, is characterized in that, described synchronization module comprises records buffering area, reference buffer district and data synchronisation unit, wherein:
Record buffering area, carry out buffer memory for the audio stream that described this locality is recorded;
Reference buffer district, for flowing to row cache to described reference audio;
Data synchronisation unit, in the time that described reference buffer district receives data, it is synchronous that the audio stream that described this locality is recorded and described reference audio flow to row data.
8. talkback unit according to claim 6, it is characterized in that, described echo cancellation module specifically for: according to the time period of default fixed intervals, check the frame data of the audio stream of opposite end transmission, in the time that frame data are less than the predetermined buffer value of speex algorithm, the echo in the audio stream of described this locality being recorded by speex algorithm is filtered; When the frame data of audio stream that send when opposite end are greater than the predetermined buffer value of speex algorithm, discarded part downlink data upon handover.
9. talkback unit according to claim 7, is characterized in that, described audio coding module specifically for: to filter after audio stream carry out frame by frame buffer memory and compression coding.
10. a voice-over-net intercom system, is characterized in that, comprises receiving system and the voice-over-net talkback unit as described in claim 6-9 Arbitrary Term, and described receiving system comprises receiver module, audio decoder module and audio playing module, wherein:
Described receiver module, for the audio stream after received code compression;
Described audio decoder module, for the audio stream reducing after the compression coding receiving;
Described audio playing module, for playing the audio stream of reduction.
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Cited By (13)
| Publication number | Priority date | Publication date | Assignee | Title |
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| CN104468146A (en) * | 2014-12-10 | 2015-03-25 | 深圳市来邦实业有限公司 | Broadcast intercom system based on IP network |
| CN105516524A (en) * | 2015-12-18 | 2016-04-20 | 合肥寰景信息技术有限公司 | Network voice synchronous de-noising communication method |
| CN105592240A (en) * | 2015-12-18 | 2016-05-18 | 合肥寰景信息技术有限公司 | Network voice synchronous de-noising communication device |
| CN105611222A (en) * | 2015-12-25 | 2016-05-25 | 北京紫荆视通科技有限公司 | Voice data processing method, device and system and controlled device |
| CN105611092A (en) * | 2015-12-18 | 2016-05-25 | 合肥寰景信息技术有限公司 | Network voice synchronization and denoising communication system |
| CN105915738A (en) * | 2016-05-30 | 2016-08-31 | 宇龙计算机通信科技(深圳)有限公司 | Echo cancellation method, echo cancellation device and terminal |
| CN106488348A (en) * | 2016-08-31 | 2017-03-08 | 西南大学 | Wireless mic software |
| CN106792277A (en) * | 2015-11-25 | 2017-05-31 | 北京国基科技股份有限公司 | A kind of speech talkback method and device under unstable network |
| CN107426708A (en) * | 2016-05-23 | 2017-12-01 | 石家庄市善理通益科技有限公司 | The switching method and apparatus of intercommunication terminal, call model |
| CN108847863A (en) * | 2018-04-19 | 2018-11-20 | 力同科技股份有限公司 | A kind of integrated talkback chip, integrated talkback terminal and signal processing method |
| CN110473563A (en) * | 2019-08-19 | 2019-11-19 | 山东省计算中心（国家超级计算济南中心） | Breathing detection method, system, equipment and medium based on time-frequency characteristics |
| CN112788263A (en) * | 2019-11-04 | 2021-05-11 | 成都鼎桥通信技术有限公司 | Method and equipment for controlling multifunctional recorder to record voice |
| CN114760389A (en) * | 2022-06-16 | 2022-07-15 | 腾讯科技（深圳）有限公司 | Voice communication method and device, computer storage medium and electronic equipment |
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| Publication number | Priority date | Publication date | Assignee | Title |
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| CN104468146A (en) * | 2014-12-10 | 2015-03-25 | 深圳市来邦实业有限公司 | Broadcast intercom system based on IP network |
| CN104468146B (en) * | 2014-12-10 | 2018-03-16 | 深圳来邦科技有限公司 | The broadcast intercom system of IP based network |
| CN106792277A (en) * | 2015-11-25 | 2017-05-31 | 北京国基科技股份有限公司 | A kind of speech talkback method and device under unstable network |
| CN105516524A (en) * | 2015-12-18 | 2016-04-20 | 合肥寰景信息技术有限公司 | Network voice synchronous de-noising communication method |
| CN105592240A (en) * | 2015-12-18 | 2016-05-18 | 合肥寰景信息技术有限公司 | Network voice synchronous de-noising communication device |
| CN105611092A (en) * | 2015-12-18 | 2016-05-25 | 合肥寰景信息技术有限公司 | Network voice synchronization and denoising communication system |
| CN105611222A (en) * | 2015-12-25 | 2016-05-25 | 北京紫荆视通科技有限公司 | Voice data processing method, device and system and controlled device |
| CN105611222B (en) * | 2015-12-25 | 2019-03-15 | 北京紫荆视通科技有限公司 | Audio data processing method, device, controlled device and system |
| CN107426708A (en) * | 2016-05-23 | 2017-12-01 | 石家庄市善理通益科技有限公司 | The switching method and apparatus of intercommunication terminal, call model |
| CN105915738A (en) * | 2016-05-30 | 2016-08-31 | 宇龙计算机通信科技(深圳)有限公司 | Echo cancellation method, echo cancellation device and terminal |
| CN106488348A (en) * | 2016-08-31 | 2017-03-08 | 西南大学 | Wireless mic software |
| CN108847863A (en) * | 2018-04-19 | 2018-11-20 | 力同科技股份有限公司 | A kind of integrated talkback chip, integrated talkback terminal and signal processing method |
| CN110473563A (en) * | 2019-08-19 | 2019-11-19 | 山东省计算中心（国家超级计算济南中心） | Breathing detection method, system, equipment and medium based on time-frequency characteristics |
| CN112788263A (en) * | 2019-11-04 | 2021-05-11 | 成都鼎桥通信技术有限公司 | Method and equipment for controlling multifunctional recorder to record voice |
| CN112788263B (en) * | 2019-11-04 | 2023-06-23 | 成都鼎桥通信技术有限公司 | Method and equipment for controlling multifunctional recorder to record voice |
| CN114760389A (en) * | 2022-06-16 | 2022-07-15 | 腾讯科技（深圳）有限公司 | Voice communication method and device, computer storage medium and electronic equipment |
| CN114760389B (en) * | 2022-06-16 | 2022-09-02 | 腾讯科技（深圳）有限公司 | Voice communication method and device, computer storage medium and electronic equipment |
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Application publication date: 20140702 |