CN101309331A - Voice communication method, apparatus, system thereof and data card, terminal - Google Patents

Voice communication method, apparatus, system thereof and data card, terminal Download PDF

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Publication number
CN101309331A
CN101309331A CNA2008101271077A CN200810127107A CN101309331A CN 101309331 A CN101309331 A CN 101309331A CN A2008101271077 A CNA2008101271077 A CN A2008101271077A CN 200810127107 A CN200810127107 A CN 200810127107A CN 101309331 A CN101309331 A CN 101309331A
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China
Prior art keywords
speech data
buffering
quality testing
data
speech
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CNA2008101271077A
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Chinese (zh)
Inventor
谢葱茏
郭业辉
白春荣
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Huawei Device Shenzhen Co Ltd
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Shenzhen Huawei Communication Technologies Co Ltd
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Priority to CNA2008101271077A priority Critical patent/CN101309331A/en
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Abstract

The invention discloses a speech communication method, a device and the system, the data card and the terminal of the speech communication method. The speech communication method in the invention includes that the speech data is received and the quality of the speech data is detected; the size of the buffer is adjusted according to the quality detection result; the buffer with the adjusted size is adopted to buffer the received speech data; the buffered speech data is sent. Based on the speech communication method, the device and the system, the data card and the terminal of the speech communication method, the communication requirements in different situations are adapted; meanwhile, the dynamic balance is realized between the quality of the speech and the conversation delay.

Description

The method of speech communication, device and system thereof and data card, terminal
Technical field
The present invention relates to communication technical field, particularly a kind of method of speech communication, device and system thereof and data card, terminal.
Background technology
Along with the development of wireless network, the application of carrying out on Radio Link is more and more, by wireless network, can carry out data service, also can carry out conventional speech business.The equipment of surfing the Net specially, for example data card (wireless modem) is when carrying out speech business, because be subjected to the restriction of network condition, the restriction of transmission and the restriction of data source itself, the data that send and receive jitter phenomenon can occur inevitably, therefore need make buffered to the data that send and receive.
In the existing technology, the buffering area of employing fixed size prevents the shake of data in transmitting audio data.As described in Figure 1, data card A receives after the data that terminal A issues, because terminal A issues the speed instability of data, and eat dishes without rice or wine to transmit need be according to fixing speed to DSP (Digital SignalProcessor, digital signal processor) writes data, encode to be sent to then by DSP then and eat dishes without rice or wine.A kind of common situation is 16bit, and the sampling of 8000Hz just needs the 20 milliseconds of speech datas that write 320 bytes to DSP in every interval.Therefore data card A need make buffered to the data that receive, and sends to air interface then.
Equally, after terminal B received the data that data card B reports, because need play voice to sound card according to fixing speed, and also there was the problem of shake in the data that data card reports, so terminal B also will make buffered to the data that receive.
In research and practice process to prior art, the inventor finds to exist at least in the prior art following problem: (1) buffer size is fixed, in case after being provided with, just be difficult to adjust, has no idea to accomplish the self adaptation size.But different network environments, the jitter amplitude of different data sources is different, and adopts the shake of the buffering area equilibrium criterion of fixed size, has the excessive or too small balance between conversation time delay and speech quality of may having no idea.(2) in order to increase the quality of voice, just must guarantee the continuity conversed, will increase the size of buffering area, so just increased delay, if buffer size has been fixed, when needs improve the quality or reduce delay, just be difficult to change, therefore, be difficult in balance between conversation time delay and the speech quality.
Summary of the invention
The technical problem that the embodiment of the invention will solve provides a kind of method, device and system thereof of speech communication and data card, terminal, can adapt to the conversation requirement under the different situations, reaches dynamic equilibrium between the delay of the quality of voice and conversation.
For solving the problems of the technologies described above, the embodiment of the invention is achieved through the following technical solutions:
One embodiment of the invention provides a kind of method of speech communication, comprising:
Receive speech data, described speech data is carried out quality testing;
According to the quality testing result size of buffering area is adjusted, the buffering area that employing is adjusted after the size cushions the described speech data that receives;
Speech data after the buffering is sent.
One embodiment of the invention provides a kind of device of speech communication, comprising:
The quality testing unit is used for the speech data that receives is carried out quality testing;
Buffer cell is used for the speech data that receives is cushioned;
Adjustment unit is used for according to the quality testing result of quality testing unit output the size of described buffer cell being adjusted;
Transmitting element is used for the speech data after the described buffer cell reading and sending buffering.
One embodiment of the invention provides a kind of system of speech communication, comprising: transmitting terminal, receiving terminal;
Described transmitting terminal comprises:
First terminal is used for receiving and transmitting speech data;
First data card, be used for the speech data that receives from described first terminal is carried out quality testing, adjust the size of buffering area according to the quality testing result, that adopts that the buffering area adjusted after the size receives cushions the speech data after air interface sends buffering to described speech data;
Described receiving terminal comprises:
Second data card is used for receiving speech data from air interface;
Second terminal, be used for the speech data that receives from described second data card is carried out quality testing, adjust the size of buffering area according to the quality testing result, adopt the buffering area of adjusting after the size that the described speech data that receives is cushioned, the speech data after the buffering is write sound card.
Above technical scheme as can be seen, according to the speech data quality that receives the size of buffering area is adjusted, therefore the speech data that requires for the different quality that receives, the size of can self adaptation adjusting buffering area to be satisfying the conversation requirement under the different situations, thereby reaches dynamic equilibrium between the time delay of speech data quality and conversation.
Description of drawings
The system schematic that Fig. 1 provides for prior art.
The method flow diagram that Fig. 2 one embodiment of the invention provides;
The method flow diagram that another embodiment of Fig. 3 the present invention provides;
The method flow diagram that Fig. 4 another embodiment of the present invention provides;
The device schematic diagram that Fig. 5 provides for one embodiment of the invention;
The data card schematic diagram that Fig. 6 provides for one embodiment of the invention;
The terminal schematic diagram that Fig. 7 provides for one embodiment of the invention;
The system schematic that Fig. 8 provides for one embodiment of the invention.
Embodiment
The embodiment of the invention provides a kind of method, device and system thereof of speech communication and data card, terminal, be used for when speech communication, size according to the adjustment buffering area of the speech data quality adaptation that receives, thereby make receiving terminal adapt to conversation requirement under the different situations, between the delay of the quality of speech data and conversation, reach dynamic equilibrium.In order to make technical scheme of the present invention clearer, enumerate embodiment below and be elaborated.
Referring to Fig. 2, the method flow diagram of a kind of speech communication that provides for one embodiment of the invention comprises:
S101: receive speech data, described speech data is carried out quality testing, judge after current buffer size is to the speech data buffering that receives whether satisfy default voice quality,, then carry out S102 if do not satisfy.
S102: according to the quality testing result size of buffering area is adjusted, so that satisfy default voice quality through the speech data after the buffering area buffering of adjusting size.
S103: the buffering area that employing is adjusted after the size cushions the described speech data that receives.
S104: the speech data after the buffering is sent.
This embodiment as can be seen, according to the speech data quality that receives the size of buffering area is adjusted, therefore the speech data that requires for the different quality that receives, the size of can self adaptation adjusting buffering area can reach dynamic equilibrium to satisfy the conversation requirement under the different situations between the time delay of speech data quality and conversation.
Referring to Fig. 3, the method flow diagram of a kind of speech communication that provides for one embodiment of the invention comprises:
S201: data card receives speech data from terminal.
S202: the speech data that receives is carried out quality testing.For example, data card a is after receiving the speech data that terminal a issues, this data card a detects the speech data that receives by built-in voice monitoring module, judge whether the speech data after the buffering can reach default voice quality under the situation that current buffering area cushions the data that receive.
S203: the size of buffering area is adjusted according to the quality testing result.Concrete, can generate one according to the testing result that step S202 obtains and adjust coefficient, carry out the adjustment of buffer size according to this adjustment coefficient, generally speaking, when making speech data after the buffering satisfy default voice quality, as much as possible the size of buffering area is adjusted to minimum.
S204: the buffering area that employing is adjusted after the size cushions the speech data that receives.
S205: read the speech data after the buffering, and the speech data after air interface sends this buffering.So far, realized that data card to receiving the buffering and the transmission of speech data, can also comprise the steps: after S204
Whether judgement sends through the speech data that cushions finishes, if, return step S202, the speech data that sends through receiving once more behind the speech data that cushions is carried out quality testing.Otherwise continue to detect, finish up to speech data transmission through buffering.
This embodiment as can be seen, data card is according to the speech data quality that receives, the size of adaptive adjustment buffering area, make when the speech data that air interface sends is satisfying default speech data quality, reduce the time delay of conversing as much as possible, thereby between conversation time delay and speech quality, reach balance.
Referring to Fig. 4, the method flow diagram of a kind of speech communication that provides for one embodiment of the invention comprises:
S301: terminal receives speech data from data card.
S302: the speech data that receives is carried out quality testing.For example, terminal b is not after receiving the speech data that data card b issues, this terminal b detects the speech data that receives by built-in voice monitoring module, judge whether the speech data after the buffering can reach default voice quality under the situation that current buffering area cushions the data that receive.
S303: the size of buffering area is adjusted according to the quality testing result.Concrete, can generate one according to the testing result that step S302 obtains and adjust coefficient, carry out the adjustment of buffer size according to this adjustment coefficient, generally speaking, when making speech data after the buffering satisfy default voice quality, as much as possible the size of buffering area is adjusted to minimum.
S304: the buffering area that employing is adjusted after the size cushions the speech data that receives.
S305: read the speech data after the buffering, and the speech data after will cushioning writes sound card.
So far, realized that data card to receiving the buffering and the transmission of speech data, can also comprise the steps: after S304
Whether judgement sends through the speech data that cushions finishes, if, return step S302, the speech data that sends through receiving once more behind the speech data that cushions is carried out quality testing.Otherwise continue to detect, finish up to speech data transmission through buffering.
This embodiment as can be seen, terminal is according to the speech data quality that receives, the size of adaptive adjustment buffering area, make the speech data that writes sound card when satisfying default speech data quality, reduce the time delay of conversing as much as possible, thereby between conversation time delay and speech quality, reach balance.
Referring to Fig. 5, the device schematic diagram of a kind of speech communication that provides for one embodiment of the invention comprises:
Quality testing unit 401 is used for the speech data that receives is carried out quality testing, thereby judges whether can reach given voice quality under current buffering area quantity.
Buffer cell 402 is used for the speech data that receives is cushioned.
Adjustment unit 403 is used for according to the quality testing result of quality testing unit 201 outputs the size of buffer cell 402 being adjusted.Concrete, can generate one according to the testing result of the 201 pairs of voice qualities in quality testing unit and adjust coefficient, carry out the adjustment of buffer size according to this adjustment coefficient, generally speaking, when making speech data after the buffering satisfy default voice quality, as much as possible the size of buffering area is adjusted to minimum.
Transmitting element 404 is used for reading through the speech data after buffer cell 203 bufferings from buffering unit 402, and sends the speech data after this buffering.
After the speech data of the 401 pairs of receptions in quality testing unit carries out quality testing, the quality testing result is sent to adjustment unit 403, adjustment unit 403 is according to the 401 quality testing results that receive adjust the size of buffer cell 402 from the quality testing unit, the speech data of the 402 pairs of receptions of buffer cell after the adjustment size cushions, and by the speech data after the transmitting element 404 transmission bufferings.
In another embodiment of the present invention, described device also comprises:
Feedback unit, the speech data that is used for described transmitting element 404 sends when finishing, to the message of described quality testing unit 401 transmission quality testings, so that the speech data that receives once more behind the speech data of described quality testing unit 401 pairs of transmissions processes buffering carries out quality testing.
Present embodiment has increased feedback unit, can be so that this quality testing unit 401 carries out quality testing to the speech data that transmitting element transmission finishing back receives timely.
Referring to Fig. 6, the schematic diagram of a kind of data card that provides for one embodiment of the invention comprises:
Quality testing unit 401 is used for the speech data that receives is carried out quality testing, thereby judges whether can reach given voice quality under current buffering area quantity.
Buffer cell 402 is used for the speech data that receives is cushioned.
Adjustment unit 403 is used for according to the quality testing result of quality testing unit 201 outputs the size of buffer cell 402 being adjusted.Concrete, can generate one according to the testing result of the 201 pairs of voice qualities in quality testing unit and adjust coefficient, carry out the adjustment of buffer size according to this adjustment coefficient, generally speaking, when making speech data after the buffering satisfy default voice quality, as much as possible the size of buffering area is adjusted to minimum.
First transmitting element 501 is used for reading speech data after the buffering, the speech data after air interface sends described buffering from described buffer cell 402.
After the speech data of the 401 pairs of receptions in quality testing unit carries out quality testing, the quality testing result is sent to adjustment unit 403, adjustment unit 403 is according to the 401 quality testing results that receive adjust the size of buffer cell 402 from the quality testing unit, the speech data of the 402 pairs of receptions of buffer cell after the adjustment size cushions, and by the speech data of transmitting element 501 after air interface sends described buffering.
In another embodiment of the present invention, described data card also comprises:
Feedback unit, the speech data that is used for described transmitting element 404 sends when finishing, to the message of described quality testing unit 401 transmission quality testings, so that the speech data that receives once more behind the speech data of described quality testing unit 401 pairs of transmissions processes buffering carries out quality testing.
Present embodiment has increased feedback unit, can be so that this quality testing unit 401 carries out quality testing to the speech data that transmitting element transmission finishing back receives timely.
Referring to Fig. 7, the schematic diagram of a kind of terminal that provides for one embodiment of the invention comprises:
Quality testing unit 401 is used for the speech data that receives is carried out quality testing, thereby judges whether can reach given voice quality under current buffering area quantity.
Buffer cell 402 is used for the speech data that receives is cushioned.
Adjustment unit 403 is used for according to the quality testing result of quality testing unit 201 outputs the size of buffer cell 402 being adjusted.Concrete, can generate one according to the testing result of the 201 pairs of voice qualities in quality testing unit and adjust coefficient, carry out the adjustment of buffer size according to this adjustment coefficient, generally speaking, when making speech data after the buffering satisfy default voice quality, as much as possible the size of buffering area is adjusted to minimum.
Second transmitting element 601 is used for reading speech data after the buffering from described buffer cell 402, and the speech data after the described buffering is write sound card.
After the speech data of the 401 pairs of receptions in quality testing unit carries out quality testing, the quality testing result is sent to adjustment unit 403, adjustment unit 403 is according to the 401 quality testing results that receive adjust the size of buffer cell 402 from the quality testing unit, the speech data of adjusting the 402 pairs of receptions of buffer cell after the size cushions, and the speech data after will cushioning by second transmitting element 601 writes sound card.
In another embodiment of the present invention, described terminal also comprises:
Feedback unit, the speech data that is used for described transmitting element 404 sends when finishing, to the message of described quality testing unit 401 transmission quality testings, so that the speech data that receives once more behind the speech data of described quality testing unit 401 pairs of transmissions processes buffering carries out quality testing.
Present embodiment has increased feedback unit, can be so that this quality testing unit 401 carries out quality testing to the speech data that transmitting element transmission finishing back receives timely.
Referring to Fig. 8, the system schematic of a kind of speech communication that provides for one embodiment of the invention comprises: transmitting terminal 701, receiving terminal 702;
Described transmitting terminal 701 comprises:
First terminal 703 is used for receiving and transmitting speech data.
First data card 704, be used for the speech data that receives from first terminal 703 is carried out quality testing, adjust the size of buffering area according to the quality testing result, adopt and adjust the speech data that the buffering area after the size receives and cushion the speech data after air interface sends buffering receiving.
For example, the speech data that receives from first terminal 703 is carried out quality testing, whether the size of judging current buffering area can satisfy default voice quality after to the speech data buffering that receives, if do not satisfy, then the testing result according to voice quality generates an adjustment coefficient, carries out the adjustment of buffer size according to this adjustment coefficient, generally speaking, when making speech data after the buffering satisfy default voice quality, as much as possible the size of buffering area is adjusted to minimum.Utilize the size of adjusted buffering area that the speech data that first terminal 503 receives is cushioned, and the speech data after air interface sends buffering.
Described receiving terminal 702 comprises:
Second data card 705 is used for receiving speech data from air interface;
Second terminal 706, be used for the speech data that receives from described second data card 705 is carried out quality testing, adjust the size of buffering area according to the quality testing result, adopt the buffering area of adjusting after the size that the speech data that receives is cushioned, the speech data after the buffering is write sound card.
For example, the speech data that receives from second data card 705 is carried out quality testing, whether the size of judging current buffering area can satisfy default voice quality after to the speech data buffering that receives, if do not satisfy, then the testing result according to voice quality generates an adjustment coefficient, carries out the adjustment of buffer size according to this adjustment coefficient, generally speaking, when making speech data after the buffering satisfy default voice quality, as much as possible the size of buffering area is adjusted to minimum.Utilize the size of adjusted buffering area that the speech data that second data card 705 receives is cushioned, and the speech data after will cushioning write sound card.
After first terminal 703 of transmitting terminal 701 receives speech data, send to first data card 704 of transmitting terminal 701, be sent to air interface after the speech data buffering of 704 pairs of receptions of first data card of transmitting terminal 701, after second data card 705 of receiving terminal 702 receives speech data from air interface, send to second terminal 706 of receiving terminal 702, write sound card after the speech data buffering of 706 pairs of receptions of second terminal of receiving terminal 702.
In another embodiment of the present invention, described system also comprises:
First feedback unit, the speech data that is used to detect after process cushions sends when finishing, to the message of first data card, 704 transmission quality testings, so that the speech data that receives once more behind the speech data of 704 pairs of transmissions processes of described first data card buffering carries out quality testing.
This embodiment has increased by first feedback unit, can be so that first data card 704 carries out quality testing to the speech data that sends through receiving once more behind the speech data of buffering timely.
Second feedback unit, the speech data that is used to detect after process cushions sends when finishing, to the message of described second terminal 706 transmission quality testings, so that the speech data that receives once more behind the speech data of 706 pairs of transmissions processes of described second terminal buffering carries out quality testing.
Present embodiment has increased by second feedback unit, can be so that second terminal 706 is carried out quality testing to the speech data that receives once more behind the speech data of process buffering timely.
Above embodiment adjusts the size of buffering area according to the speech data quality that receives as can be seen, and therefore for the speech data of the different quality requirement that receives, the size of self adaptation adjustment buffering area satisfies the conversation requirement under the different situations; And the speech data after the feasible buffering reduces the time delay of conversing as much as possible, thereby reach dynamic equilibrium between the delay of speech data quality and conversation when satisfying default speech data quality.
One of ordinary skill in the art will appreciate that all or part of step that realizes in the foregoing description implementation procedure is to instruct relevant hardware to finish by program, described program can be stored in a kind of computer-readable recording medium.
The above-mentioned storage medium of mentioning can be a read-only memory, disk or CD etc.
More than method, device and system thereof and data card, the terminal of a kind of speech communication provided by the present invention is described in detail, for one of ordinary skill in the art, thought according to the embodiment of the invention, part in specific embodiments and applications all can change, in sum, this description should not be construed as limitation of the present invention.

Claims (12)

1, a kind of method of speech communication is characterized in that, comprising:
Receive speech data, described speech data is carried out quality testing;
According to the quality testing result size of buffering area is adjusted, the buffering area that employing is adjusted after the size cushions the described speech data that receives;
Speech data after the buffering is sent.
2, method according to claim 1 is characterized in that, described method also comprises:
Whether judgement sends through the speech data that cushions finishes, if the speech data that sends through receiving once more behind the speech data that cushions is carried out quality testing.
3, method according to claim 1 is characterized in that, describedly according to the quality testing result size of buffering area is adjusted, and comprising:
Generate one according to the quality testing result and adjust coefficient, the size of buffering area is adjusted according to this adjustment coefficient.
4, method according to claim 1 is characterized in that, the described speech data that obtains and send after the buffering comprises:
Read the speech data after the buffering, the speech data after air interface sends described buffering; Perhaps,
Read the speech data after the buffering, the speech data after the described buffering is write sound card.
5, a kind of device of speech communication is characterized in that, comprising:
The quality testing unit is used for the speech data that receives is carried out quality testing;
Buffer cell is used for the speech data that receives is cushioned;
Adjustment unit is used for according to the quality testing result of quality testing unit output the size of described buffer cell being adjusted;
Transmitting element is used for the speech data after the described buffer cell reading and sending buffering.
6, device according to claim 5 is characterized in that, described transmitting element comprises:
First transmitting element is used for reading speech data after the buffering, the speech data after air interface sends described buffering from described buffer cell.
7, device according to claim 5 is characterized in that, described transmitting element comprises:
Second transmitting element is used for reading speech data after the buffering from described buffer cell, and the speech data after the described buffering is write sound card.
8, device according to claim 5 is characterized in that, described device also comprises:
Feedback unit, the speech data that is used for described transmitting element sends when finishing, to the message of described quality testing unit transmission quality testing, so that described quality testing unit carries out quality testing to the speech data that sends through receiving once more behind the speech data of buffering.
9, a kind of data card is characterized in that, comprise, and voice communication device, described voice communication device further comprises:
The quality testing unit is used for the speech data that receives is carried out quality testing;
Buffer cell is used for the speech data that receives is cushioned;
Adjustment unit is used for according to the quality testing result of quality testing unit output the size of described buffer cell being adjusted;
First transmitting element is used for reading speech data after the buffering, the speech data after air interface sends described buffering from described buffer cell.
10, a kind of terminal is characterized in that, comprise, and voice communication device, described voice communication device further comprises:
The quality testing unit is used for the speech data that receives is carried out quality testing;
Buffer cell is used for the speech data that receives is cushioned;
Adjustment unit is used for according to the quality testing result of quality testing unit output the size of described buffer cell being adjusted;
Second transmitting element is used for reading speech data after the buffering from described buffer cell, and the speech data after the described buffering is write sound card.
11, a kind of system of speech communication is characterized in that, comprising: transmitting terminal, receiving terminal;
Described transmitting terminal comprises:
First terminal is used for receiving and transmitting speech data;
First data card, be used for the speech data that receives from described first terminal is carried out quality testing, adjust the size of buffering area according to the quality testing result, that adopts that the buffering area adjusted after the size receives cushions the speech data after air interface sends buffering to described speech data;
Described receiving terminal comprises:
Second data card is used for receiving speech data from air interface;
Second terminal, be used for the speech data that receives from described second data card is carried out quality testing, adjust the size of buffering area according to the quality testing result, adopt the buffering area of adjusting after the size that the described speech data that receives is cushioned, the speech data after the buffering is write sound card.
12, system according to claim 11 is characterized in that, described system also comprises:
First feedback unit, the speech data that is used to detect after process cushions sends when finishing, to the message of described first data card transmission quality testing, so that described first data card carries out quality testing to the speech data that sends through receiving once more behind the speech data of buffering;
Second feedback unit is used to detect through the speech data after the buffering and sends when finishing, and sends the message of quality testing to described second terminal, so that described second terminal is carried out quality testing to the speech data that sends through receiving once more behind the speech data of buffering.
CNA2008101271077A 2008-06-19 2008-06-19 Voice communication method, apparatus, system thereof and data card, terminal Pending CN101309331A (en)

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Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2011035693A1 (en) * 2009-09-22 2011-03-31 中兴通讯股份有限公司 Wireless internet accessing device and method thereof for voice buffer
CN103067383A (en) * 2012-12-27 2013-04-24 华为终端有限公司 Wireless internet access device and multimedia communication method
US11343301B2 (en) * 2017-11-30 2022-05-24 Goto Group, Inc. Managing jitter buffer length for improved audio quality

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2011035693A1 (en) * 2009-09-22 2011-03-31 中兴通讯股份有限公司 Wireless internet accessing device and method thereof for voice buffer
CN103067383A (en) * 2012-12-27 2013-04-24 华为终端有限公司 Wireless internet access device and multimedia communication method
US11343301B2 (en) * 2017-11-30 2022-05-24 Goto Group, Inc. Managing jitter buffer length for improved audio quality

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Application publication date: 20081119