CN102915736B - Mixed audio processing method and stereo process system - Google Patents
Mixed audio processing method and stereo process system Download PDFInfo
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Abstract
The invention provides a kind of mixed audio processing method and system, its method comprises step: the coded audio data and the audio encoding type that receive client transmission; Decoded data is obtained by the audio decoder encode audio decoding data corresponding with audio encoding type; The relatively sampling rate of decoded data and the size of default server sampling rate; If the sampling rate of decoded data is greater than default server sampling rate, then down-sampled process is carried out to decoded data; If the sampling rate of decoded data is less than default server sampling rate, then carry out rising sampling processing to decoded data; Decoded data after decoded data/down-sampled process time identical with the server sampling rate preset to the sampling rate of decoded data/rising the decoded data after sampling processing carries out mixing operation acquisition audio mixing data.The present invention can allow the terminal device of participant each side to use different audio codecs, considerably increases the range of application that video conferencing technology interconnects.
Description
Technical field
The present invention relates to video conference field, particularly relate to a kind of mixed audio processing method and stereo process system.
Background technology
At present, along with developing rapidly of network technology and video conference, the application technology relative maturity of Audio and Video.In actual applications, the complementation process of audio frequency is still in position that is the most basic, core, more harsh to the requirement of real-time of audio frequency, thus, multiple terminals of the different location in using for reality, to carry out the mutual of the real-time audio of multiple point, under the condition of the permission of the network bandwidth, need multichannel voice frequency to carry out audio mixing according to certain strategy, and final coding sends other terminal to.
But no matter be traditional video conference manufacturer or network teleconference manufacturer, their video conference all requires: each side of participant must use the audio codec ability participant identical with server.If the terminal device of participant side does not possess or does not use identical audio codec, just cannot participant.Not only limit the terminal device of participant like this, also limit the range of application that videoconference server interconnects.
Summary of the invention
The object of the present invention is to provide a kind of be simple and easy to execute mixed audio processing method and stereo process system, allow the terminal device of participant each side to use different audio codecs, considerably increase the range of application that video conferencing technology interconnects.
Object of the present invention is achieved through the following technical solutions:
A kind of mixed audio processing method, comprises the steps:
Receive coded audio data and the audio encoding type of client transmission;
By the audio decoder corresponding with described audio encoding type, decoding is carried out to described coded audio data and obtain decoded data;
The sampling rate of more described decoded data and the size of default server sampling rate;
If the sampling rate of described decoded data is greater than default server sampling rate, then down-sampled process is carried out to described decoded data;
If the sampling rate of described decoded data is less than default server sampling rate, then carry out rising sampling processing to described decoded data;
Decoded data after decoded data/down-sampled process time identical with the server sampling rate preset to the sampling rate of described decoded data/rising the decoded data after sampling processing carries out mixing operation acquisition audio mixing data;
The size of the sampling rate of the sampling rate of more described audio mixing data and audio codec corresponding to described client;
If the sampling rate of described audio mixing data is greater than the sampling rate of audio codec corresponding to described client, then down-sampled process is carried out to described audio mixing data;
If the sampling rate of described audio mixing data is less than the sampling rate of audio codec corresponding to described client, then falling-rising sampling processing is carried out to described audio mixing data;
Audio mixing data after audio mixing data/down-sampled process when adopting the sampling rate of audio coder to described audio mixing data corresponding with described client to equal the sampling rate of audio codec corresponding to described client/the rise audio mixing data after sampling processing are carried out coding and are obtained audio mixing coded data;
Described audio mixing coded data is sent to described client;
Judge whether the packet loss of the coded audio data received is greater than preset value;
If so, then produce the instruction of the audio codec adopting low bit-rate, and described instruction is sent to corresponding client.
A kind of stereo process system, comprising:
Receiving element, for receiving coded audio data and the audio encoding type information of client transmission;
Decoding unit, obtains decoded data for carrying out decoding by the audio decoder corresponding with described audio encoding type to described coded audio data;
First comparing unit, for the sampling rate of more described decoded data and the size of default server sampling rate;
First down-sampled unit, during for being greater than default server sampling rate in the sampling rate of described decoded data, carries out down-sampled process to described decoded data;
First liter of sampling unit, for being less than default server sampling rate in the sampling rate of described decoded data, then carries out rising sampling processing to described decoded data;
Downmixing unit, for the decoded data after decoded data/down-sampled process time identical with the server sampling rate preset to the sampling rate of described decoded data/rising the decoded data after sampling processing carries out mixing operation acquisition audio mixing data;
Second comparing unit, for the size of the sampling rate of the sampling rate of more described audio mixing data and audio codec corresponding to described client;
Second down-sampled unit, for when the sampling rate of described audio mixing data is greater than the sampling rate of audio codec corresponding to described client, then carries out down-sampled process to described audio mixing data;
Second liter of sampling unit, for when the sampling rate of described audio mixing data is less than the sampling rate of audio codec corresponding to described client, then carries out rising sampling processing to described audio mixing data;
Audio mixing data/the second liter sampling unit after the down-sampled process of the down-sampled unit of audio mixing data/the second when coding unit adopts the sampling rate of audio coder to described audio mixing data corresponding with described client to equal the sampling rate of audio codec corresponding to described client rises the audio mixing data after sampling processing to carry out coding and obtains audio mixing coded data;
Transmitting element, for sending to described client by described audio mixing coded data;
Also comprise judging unit and control module, wherein:
Described judging unit is for judging whether the packet loss of the coded audio data that described receiving element receives is greater than preset value;
Described control module is used for when the result of determination of described judging unit is for being, produces the instruction of the audio codec adopting low bit-rate;
Described transmitting element is also for being sent to corresponding client by described instruction.
According to the scheme of the invention described above, after the coded audio data receiving client transmission and audio encoding type, by the audio decoder corresponding with described audio encoding type, decoding is carried out to described coded audio data and obtain decoded data, according to the sampling rate of decoded data and the magnitude relationship of server sampling rate, corresponding sample rate conversion is done (when the sampling rate of decoded data is equal with server sampling rate to decoded data again, do not change sampling rate), then carry out mixing operation.The present invention can allow the terminal device of participant each side to use different audio codecs (comprising audio coder and audio decoder), considerably increases the range of application that video conferencing technology interconnects.
Accompanying drawing explanation
Fig. 1 is the schematic flow sheet of the mixed audio processing method of the embodiment of the present invention;
Fig. 2 is the schematic flow sheet of the stereo process system of one embodiment of the invention;
Fig. 3 is the schematic flow sheet of the stereo process system of another embodiment of the present invention;
Fig. 4 is the schematic flow sheet of the stereo process system of third embodiment of the invention.
Embodiment
Below in conjunction with embodiment and accompanying drawing, the present invention is further elaborated, but embodiments of the present invention are not limited thereto.
Embodiment 1
Shown in Figure 1, be the schematic flow sheet of the mixed audio processing method of the embodiment of the present invention.As shown in Figure 1, the mixed audio processing method in this embodiment comprises step:
Step S101: the coded audio data and the audio encoding type that receive client transmission, enter step S102;
Step S102: by the audio decoder corresponding with described audio encoding type, decoding is carried out to described coded audio data and obtain decoded data, enter step S103, wherein, audio encoding type comprise G.711, G.723.1, G.722, G.722.1 Annex C, AAC-LD ... etc.;
Step S103: the sampling rate of more described decoded data and the size of default server sampling rate, if the sampling rate of described decoded data is greater than default server sampling rate, then enter step S104, if the sampling rate of described decoded data is less than default server sampling rate, then enter step S105, if the sampling rate of described decoded data is identical with the server sampling rate preset, then enter step S106, wherein, server sampling rate can be determined as the case may be;
Step S104: down-sampled process is carried out to described decoded data, enters step S106, wherein, down-sampled process can adopt mode of the prior art, does not repeat them here;
Step S105: carry out rising sampling processing to described decoded data, enter step S106, wherein, rises sampling processing and can adopt mode of the prior art, do not repeat them here.
Step S106: the decoded data after decoded data/down-sampled process time identical with the server sampling rate preset to the sampling rate of described decoded data/rising the decoded data after sampling processing carries out mixing operation acquisition audio mixing data, that is: when the sampling rate of described decoded data is identical with the server sampling rate preset, directly mixing operation is carried out to decoded data, when the sampling rate of described decoded data is different with the server sampling rate preset, the sample rate conversion performing correspondence is needed (to comprise and rise sampling processing, down-sampled process) after carry out mixing operation again.
Accordingly, according to the scheme of the present embodiment, after the coded audio data receiving client transmission and audio encoding type, by the audio decoder corresponding with described audio encoding type, decoding is carried out to described coded audio data and obtain decoded data, according to the sampling rate of decoded data and the magnitude relationship of server sampling rate, corresponding sample rate conversion is done (when the sampling rate of decoded data is equal with server sampling rate to decoded data again, do not change sampling rate), then carry out mixing operation.The present invention can allow the terminal device of participant each side to use different audio codecs (comprising audio coder and audio decoder), considerably increases the range of application that video conferencing technology interconnects.
Wherein, when the audio mixing data after mixing operation being sent to corresponding client, also need to do corresponding sample rate conversion to these audio mixing data, and the audio coder adopting client corresponding is encoded.Therefore, step can also be comprised:
The size of the sampling rate of the sampling rate of more described audio mixing data and audio codec corresponding to described client, wherein, the general server sampling rate with presetting of sampling rate of audio mixing data is identical;
If the sampling rate of described audio mixing data is greater than the sampling rate of audio codec corresponding to described client, then down-sampled process is carried out to described audio mixing data;
If the sampling rate of described audio mixing data is less than the sampling rate of audio codec corresponding to described client, then falling-rising sampling processing is carried out to described audio mixing data;
Audio mixing data after audio mixing data/down-sampled process when adopting the sampling rate of audio coder to described audio mixing data corresponding with described client to equal the sampling rate of audio codec corresponding to described client/the rise audio mixing data after sampling processing are carried out coding and are obtained audio mixing coded data, in other words, when the sampling rate of audio codec to described audio mixing data equals the sampling rate of audio codec corresponding to described client, directly can carry out encoding operation to audio mixing data, when the sampling rate of described audio mixing data is not equal to the sampling rate of audio codec corresponding to described client, first should carry out sample rate conversion to audio mixing data and (comprise down-sampled process, down-sampled process) carry out encoding operation again,
Described audio mixing coded data is sent to described client.Thus, when client receives audio mixing coded data, can carry out decoding and then obtain audio mixing data with the audio decoder of its correspondence.
On the other hand, when holding HD video meeting, the parties network situation participating in video conference is different, in order to ensure voice quality, video conference producer adopt strategy otherwise be reduce video frame per second ensure transmission speech bandwidth, ensure audio priory transmission, again or use multi code Rate of Chinese character audio codec, at the situation decline low bit-rate that network is bad.Due to high definition audio codec, such as lowest bit rate G.719 still has 32kps, when network condition continues to degenerate, still may cannot ensure that voice are smooth, thus cause the voice quality because of a whole meeting-place of the bad impact of network, side meeting-place.And packet loss can embody the quality of network quality, for this reason, can also judge whether the packet loss of the coded audio data received is greater than preset value; If so, then produce the instruction of the audio codec adopting low bit-rate, and described instruction is sent to corresponding client, wherein, preset value can be determined according to actual requirement, as selected 3% or 5%.Thus client can use the audio codec relative to present video codec with more low bit-rate, and the audio codec that the present invention uses client is unrestricted, thus when network condition continues to degenerate, still can ensure that voice are smooth.
Wherein, the packet that coded audio data is corresponding comprises a timestamp sequence number temporally arranged, if there is packet loss, the timestamp sequence number of each packet then received is discontinuous, then can determine packet loss by the timestamp sequence number of packet corresponding to coded audio data, be packet in 001 to 100 as received timestamp sequence number within the unit interval, but during by query time stamp sequence number, two data-bag losts of discovery time stamp sequence number 026,075, then can determine that current packet loss is 2%.
Embodiment 2
According to the mixed audio processing method of the invention described above, the present invention also provides a kind of stereo process system, and just the embodiment of stereo process system of the present invention is described in detail below.In this embodiment, as shown in Figure 2, the stereo process system in the present embodiment, comprises receiving element 201, decoding unit 202, first comparing unit the 203, first down-sampled unit 204, first liter of sampling unit 205, downmixing unit 206, wherein:
Receiving element 201, for receiving the A and audio encoding type that client sends, wherein, G.711 audio encoding type comprises, G.723.1, G.722, G.722.1 Annex C, AAC-LD ... etc.;
Decoding unit 202, obtains decoded data for carrying out decoding by the audio decoder corresponding with described audio encoding type to described coded audio data;
First comparing unit 203, for the sampling rate of more described decoded data and the size of default server sampling rate, wherein, server sampling rate can be determined as the case may be;
First down-sampled unit 204, during for being greater than default server sampling rate in the sampling rate of described decoded data, carry out down-sampled process to described decoded data, wherein, down-sampled process can adopt mode of the prior art, does not repeat them here;
First liter of sampling unit 205, for being less than default server sampling rate in the sampling rate of described decoded data, then carries out rising sampling processing to described decoded data, wherein, rises sampling processing and can adopt mode of the prior art, do not repeat them here;
Downmixing unit 206, carry out mixing operation for the decoded data after the decoded data/the first liter sampling unit 205 liters of sampling processing after the down-sampled process of the down-sampled unit of decoded data/the first 204 time identical with the server sampling rate preset to the sampling rate of described decoded data and obtain audio mixing data, that is: when the sampling rate of described decoded data is identical with the server sampling rate preset, directly mixing operation is carried out to decoded data, when the sampling rate of described decoded data is different with the server sampling rate preset, need to perform corresponding sample rate conversion in the working cell of correspondence and (comprise down-sampled process, down-sampled process) after carry out mixing operation again.
Accordingly, according to the scheme of the present embodiment, receiving element 201 receive client send coded audio data and audio encoding type after, decoding unit 202 obtains decoded data by carrying out decoding with corresponding audio decoder to described coded audio data, the sampling rate of the more described decoded data of the first comparing unit 203 and the size of default server sampling rate, when the sampling rate of described decoded data is greater than default server sampling rate, first down-sampled unit 204 carries out down-sampled process to described decoded data, default server sampling rate is less than in the sampling rate of described decoded data, first liter of sampling unit 205 carries out rising sampling processing to described decoded data, finally, downmixing unit 206 is to having carried out sample rate conversion (when the sampling rate of decoded data is equal with server sampling rate, do not change sampling rate) data carry out mixing operation.The present invention can allow the terminal device of participant each side to use different audio codecs, considerably increases the range of application that video conferencing technology interconnects.
Wherein, when the audio mixing data after mixing operation being sent to corresponding client, need to do corresponding sample rate conversion to audio mixing data, and the audio codec adopting client corresponding is encoded.Therefore, shown in Figure 3, stereo process system of the present invention can also comprise second comparing unit the 207, second down-sampled unit 208, second liter of sampling unit 209, coding unit 210, transmitting element 211, wherein:
Second comparing unit 207, for the size of the sampling rate of the sampling rate of more described audio mixing data and audio codec corresponding to described client, wherein, the general server sampling rate with presetting of sampling rate of audio mixing data is identical;
Second down-sampled unit 208, for when the sampling rate of described audio mixing data is greater than the sampling rate of audio codec corresponding to described client, then carries out down-sampled process to described audio mixing data;
Second liter of sampling unit 209, for when the sampling rate of described audio mixing data is less than the sampling rate of audio codec corresponding to described client, then carries out rising sampling processing to described audio mixing data;
Coding unit 210, audio mixing data after audio mixing data/the second liter sampling unit 209 liters of sampling processing after the down-sampled process of the down-sampled unit of audio mixing data/the second 208 when equaling the sampling rate of audio codec corresponding to described client for adopting the sampling rate of the audio coder corresponding with described client to described audio mixing data are carried out coding and are obtained audio mixing coded data, in other words, when the sampling rate of audio codec to described audio mixing data equals the sampling rate of audio codec corresponding to described client, directly can carry out encoding operation to audio mixing data, when the sampling rate of described audio mixing data is not equal to the sampling rate of audio codec corresponding to described client, corresponding sample rate conversion should be carried out at the unit of correspondence to audio mixing data and (comprise down-sampled process, down-sampled process) after carry out encoding operation again,
Transmitting element 211, for sending to described client by described audio mixing coded data.
On the other hand, when holding HD video meeting, the parties network situation participating in video conference is different, in order to ensure voice quality, video conference producer adopt strategy otherwise be reduce video frame per second ensure transmission speech bandwidth, ensure audio priory transmission, again or use multi code Rate of Chinese character audio codec, at the situation decline low bit-rate that network is bad.Due to high definition audio codec, such as lowest bit rate G.719 still has 32kps, when network condition continues to degenerate, still may cannot ensure that voice are smooth, thus cause the voice quality because of a whole meeting-place of the bad impact of network, side meeting-place.And packet loss can embody the quality of network quality, for this reason, shown in Figure 4, stereo process system of the present invention can also comprise judging unit 212 and control module 213, and judging unit 212 is for judging whether the packet loss of the coded audio data that described receiving element receives is greater than preset value; Control module 213, for when the result of determination of judging unit 212 is for being, produces the instruction of the audio codec adopting low bit-rate; Correspondingly, described transmitting element 211 is also for being sent to corresponding client by described instruction, preset value can be determined according to actual requirement, as selected 3% or 5%.Thus client can use the audio codec relative to present video codec with more low bit-rate, and the audio codec that the present invention uses client is unrestricted, thus when network condition continues to degenerate, still can ensure that voice are smooth.
Wherein, the packet that coded audio data is corresponding comprises a timestamp sequence number temporally arranged, if there is packet loss, the timestamp sequence number of each packet then received is discontinuous, therefore, judging unit 212 can be corresponding according to the coded audio data received the timestamp sequence number of packet determine described packet loss, be packet in 001 to 100 as received timestamp sequence number within the unit interval, but during by query time stamp sequence number, two data-bag losts of discovery time stamp sequence number 026,075, then can determine that current packet loss is 2%.
Embodiment 3
For the ease of understanding the solution of the present invention, being originally in embodiment, give an embody rule example of the present invention, but following embody rule example not forming the restriction to the scope of the claims of the present invention.
Such as, meeting-place A and meeting-place B holds a HD video meeting, uses high definition audio codec AAC-LD (sampling rate 48kHz, code check 64kps) with server negotiate.In meeting, meeting-place C will add meeting, but its client device does not have high definition audio codec AAC-LD.G.722.1C (sampling rate 32kHz, code check 32kps) client C uses audio codec.
Now according to the present invention program, after the coded audio data of client and meeting-place C of receiving meeting-place C use audio codec information G.722.1C, use audio codec G.722.1C encode audio decoding data, then carry out rising sampling processing (rising sampling rate to 48Khz) to decoded data, then with the speech data audio mixing of meeting-place A and meeting-place B.By down-sampled for the data after audio mixing to 32kHz, after G.722.1C encoding with audio codec, send back to client corresponding to C meeting-place again.When being checked through, the data packetloss rate that client corresponding to C meeting-place send is too large, can notify that this client uses the audio codec that code check is lower, as audio codec iLBC (sampling rate 8kHz, code check 13.3kps).
The above embodiment only have expressed several embodiment of the present invention, and it describes comparatively concrete and detailed, but therefore can not be interpreted as the restriction to the scope of the claims of the present invention.It should be pointed out that for the person of ordinary skill of the art, without departing from the inventive concept of the premise, can also make some distortion and improvement, these all belong to protection scope of the present invention.Therefore, the protection domain of patent of the present invention should be as the criterion with claims.
Claims (4)
1. a mixed audio processing method, is characterized in that, comprises the steps:
Receive coded audio data and the audio encoding type of client transmission;
By the audio decoder corresponding with described audio encoding type, decoding is carried out to described coded audio data and obtain decoded data;
The sampling rate of more described decoded data and the size of default server sampling rate;
If the sampling rate of described decoded data is greater than default server sampling rate, then down-sampled process is carried out to described decoded data;
If the sampling rate of described decoded data is less than default server sampling rate, then carry out rising sampling processing to described decoded data;
Decoded data after decoded data/down-sampled process time identical with the server sampling rate preset to the sampling rate of described decoded data/rising the decoded data after sampling processing carries out mixing operation acquisition audio mixing data;
The size of the sampling rate of the sampling rate of more described audio mixing data and audio codec corresponding to described client;
If the sampling rate of described audio mixing data is greater than the sampling rate of audio codec corresponding to described client, then down-sampled process is carried out to described audio mixing data;
If the sampling rate of described audio mixing data is less than the sampling rate of audio codec corresponding to described client, then falling-rising sampling processing is carried out to described audio mixing data;
Audio mixing data after audio mixing data/down-sampled process when adopting the sampling rate of audio coder to described audio mixing data corresponding with described client to equal the sampling rate of audio codec corresponding to described client/the rise audio mixing data after sampling processing are carried out coding and are obtained audio mixing coded data;
Described audio mixing coded data is sent to described client;
Judge whether the packet loss of the coded audio data received is greater than preset value;
If so, then produce the instruction of the audio codec adopting low bit-rate, and described instruction is sent to corresponding client.
2. mixed audio processing method according to claim 1, is characterized in that, the timestamp sequence number according to packet corresponding to the coded audio data received determines described packet loss.
3. a stereo process system, is characterized in that, comprising:
Receiving element, for receiving coded audio data and the audio encoding type information of client transmission;
Decoding unit, obtains decoded data for carrying out decoding by the audio decoder corresponding with described audio encoding type to described coded audio data;
First comparing unit, for the sampling rate of more described decoded data and the size of default server sampling rate;
First down-sampled unit, during for being greater than default server sampling rate in the sampling rate of described decoded data, carries out down-sampled process to described decoded data;
First liter of sampling unit, for being less than default server sampling rate in the sampling rate of described decoded data, then carries out rising sampling processing to described decoded data;
Downmixing unit, rises the decoded data after sampling processing for the decoded data/the first liter sampling unit after the down-sampled process of the down-sampled unit of decoded data/the first time identical with the server sampling rate preset to the sampling rate of described decoded data and carries out mixing operation acquisition audio mixing data;
Second comparing unit, for the size of the sampling rate of the sampling rate of more described audio mixing data and audio codec corresponding to described client;
Second down-sampled unit, for when the sampling rate of described audio mixing data is greater than the sampling rate of audio codec corresponding to described client, then carries out down-sampled process to described audio mixing data;
Second liter of sampling unit, for when the sampling rate of described audio mixing data is less than the sampling rate of audio codec corresponding to described client, then carries out rising sampling processing to described audio mixing data;
Audio mixing data/the second liter sampling unit after the down-sampled process of the down-sampled unit of audio mixing data/the second when coding unit adopts the sampling rate of audio coder to described audio mixing data corresponding with described client to equal the sampling rate of audio codec corresponding to described client rises the audio mixing data after sampling processing to carry out coding and obtains audio mixing coded data;
Transmitting element, for sending to described client by described audio mixing coded data;
Also comprise judging unit and control module, wherein:
Described judging unit is for judging whether the packet loss of the coded audio data that described receiving element receives is greater than preset value;
Described control module is used for when the result of determination of described judging unit is for being, produces the instruction of the audio codec adopting low bit-rate;
Described transmitting element is also for being sent to corresponding client by described instruction.
4. stereo process system according to claim 3, is characterized in that, described judging unit determines described packet loss according to the timestamp sequence number of packet corresponding to the coded audio data received.
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| CN103457888B (en) * | 2013-08-30 | 2016-06-08 | 深圳清华大学研究院 | Combine communicate signalling method and the device of liter sampling with digital fabrication |
| EP3155617B1 (en) * | 2014-06-10 | 2022-01-05 | MQA Limited | Digital encapsulation of audio signals |
| CN104469258A (en) * | 2014-12-04 | 2015-03-25 | 重庆晋才富熙科技有限公司 | Intelligent remote teaching video interaction mode adjusting method |
| CN108401263B (en) * | 2017-02-07 | 2021-09-17 | 大唐移动通信设备有限公司 | Voice quality assessment method and device |
| CN109524015B (en) * | 2017-09-18 | 2022-04-15 | 杭州海康威视数字技术股份有限公司 | Audio coding method, decoding method, device and audio coding and decoding system |
| CN110033781B (en) * | 2018-01-10 | 2021-06-01 | 盛微先进科技股份有限公司 | Audio processing method, apparatus and non-transitory computer readable medium |
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| EP2226995A4 (en) * | 2008-02-04 | 2012-01-11 | Nec Corp | Voice mixing device and method, and multipoint conference server |
| BRPI0905069A2 (en) * | 2008-07-29 | 2015-06-30 | Panasonic Corp | Audio coding apparatus, audio decoding apparatus, audio coding and decoding apparatus and teleconferencing system |
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