CN110910896A - Method, device, equipment and computer readable medium for adjusting call quality - Google Patents
Method, device, equipment and computer readable medium for adjusting call quality Download PDFInfo
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- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0316—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude
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Abstract
The embodiment of the invention discloses a method, a device, equipment and a computer readable medium for adjusting call quality, which relate to the field of communication, wherein the method comprises the following steps: determining a first voice adjustment strategy for improving the quality of a subsequent voice signal sent from an opposite terminal to a local terminal according to a received current voice signal; and sending the first voice adjustment strategy to the opposite terminal so as to receive a subsequent voice signal which is adjusted and responded by the opposite terminal according to the first voice adjustment strategy. The embodiment of the invention can improve the communication quality.
Description
Technical Field
The present invention relates to the field of communications, and in particular, to a method, an apparatus, a device, and a computer readable medium for adjusting call quality.
Background
The related art provides the following two methods to improve call quality. One method is that the receiver analyzes the voice signal of the other party to obtain the noise signal intensity, and sends the adjusted voice signal to the other party according to the corresponding relation between the noise signal intensity and the voice signal value preset in the terminal, so as to improve the receiving effect of the other party. This method of determining the condition based on the intensity of the ambient noise of the opposite party has a limitation, and cannot achieve the improvement purpose when the ambient noise of the opposite party is small. The other method is that the answering party analyzes the influence of the current environment noise volume on the call to adjust the call volume of the local terminal so as to improve the local answering effect, and if the volume sent by the opposite party is very small, the answering loudness is only improved from the local terminal, so that the effect is limited.
Disclosure of Invention
The method, the device, the equipment and the computer readable medium for adjusting the call quality provided by the embodiment of the invention realize a new mode for improving the call quality.
The method for adjusting the call quality provided by the embodiment of the invention comprises the following steps:
determining a first voice adjustment strategy for improving the quality of a subsequent voice signal sent from an opposite terminal to a local terminal according to a received current voice signal;
and sending the first voice adjustment strategy to the opposite terminal so as to receive a subsequent voice signal which is adjusted and responded by the opposite terminal according to the first voice adjustment strategy.
The device for adjusting the call quality provided by the embodiment of the invention comprises:
the processing module is used for determining a first voice adjustment strategy for improving the quality of a subsequent voice signal sent from the opposite terminal to the local terminal according to the received current voice signal;
and the transceiver module is used for sending the first voice adjustment strategy to the opposite terminal so as to receive a subsequent voice signal which is adjusted and responded by the opposite terminal according to the first voice adjustment strategy.
The device for adjusting the call quality provided by the embodiment of the invention comprises: the processor and the memory, the memory stores a program for adjusting the call quality, the program can be run on the processor, and the program for adjusting the call quality realizes the steps of the method for adjusting the call quality when being executed by the processor.
An embodiment of the present invention provides a computer-readable medium, on which a program for adjusting call quality is stored, where the program for adjusting call quality is executed by a processor to implement the steps of the method for adjusting call quality.
The embodiment of the invention transmits the message to the other party according to the self answering effect by the answering party through the message feedback and response mechanism no matter what scene the two parties of the call are in, and requires the other party to transmit the voice signal after the audio parameter adjustment according to the corresponding relation, namely, the voice effect required by the answering party is realized by adjusting the audio parameter of the sound source, and the call quality is improved.
Drawings
Fig. 1 is a first flowchart of a method for adjusting call quality according to an embodiment of the present invention;
fig. 2 is a second flowchart of a method for adjusting call quality according to an embodiment of the present invention;
fig. 3 is a schematic diagram of a general idea flow for adjusting call quality according to an embodiment of the present invention;
fig. 4 is a schematic working block diagram of a feedback triggering process for adjusting call quality according to an embodiment of the present invention;
FIG. 5 is a diagram of an AMR-WB speech frame structure;
FIG. 6 is a schematic diagram of AMR-WB speech frame types;
FIG. 7 is a schematic diagram of a message feedback response flow based on AMR-WB voice frames according to an embodiment of the present invention;
FIG. 8 is a schematic operational block diagram of a voice adjustment interface provided by an embodiment of the present invention;
fig. 9 is a schematic block diagram of an apparatus for adjusting call quality according to an embodiment of the present invention;
fig. 10 is a schematic block diagram of an apparatus for adjusting call quality according to an embodiment of the present invention.
Detailed Description
The embodiments of the present invention will be described in detail below with reference to the accompanying drawings, and it should be understood that the embodiments described below are only for illustrating and explaining the present invention and are not intended to limit the present invention. The use of "first" and "second" in embodiments of the invention is not intended to imply any particular order or sequence or to limit the invention, but is merely intended to distinguish one element from another element described using the same technique. The terms "comprising," "including," and the like, as used in connection with the embodiments of the present invention, are open-ended terms, i.e., meaning including, but not limited to. As used in embodiments of the present invention, "and/or" refers to any or all combinations.
The embodiment of the invention does not require the environment of the two parties in communication, and the audio parameters of the sound source can be adjusted by the message feedback and response mechanism of the embodiment of the invention to realize the voice effect required by the receiver no matter whether the two parties in communication are in a noise environment or in a quiet environment or only one party is in a noise environment. The adjustment of the audio parameters is not limited to the adjustment of the audio parameters of the call sound, but may be performed by adjusting the parameters of noise suppression, or both.
Fig. 1 is a schematic flowchart of a method for adjusting call quality according to an embodiment of the present invention, and as shown in fig. 1, the method may include:
step S101: and determining a first voice adjustment strategy for improving the quality of a subsequent voice signal sent from the opposite terminal to the local terminal according to the received current voice signal.
Step S102: and sending the first voice adjustment strategy to the opposite terminal so as to receive a subsequent voice signal which is adjusted and responded by the opposite terminal according to the first voice adjustment strategy.
In one embodiment, the step S101 includes: the local terminal analyzes the current voice signal from the opposite terminal to obtain the audio parameter of the current voice signal, and then the local terminal determines the first voice adjustment strategy according to the comparison result of the audio parameter of the current voice signal and the audio parameter preset by the local terminal.
In this embodiment, the audio parameter may be a parameter of the call sound, such as loudness, a parameter of noise mixed with the call sound, such as intensity of the noise signal, or a combination of the loudness of the call sound and the intensity of the noise signal.
In this embodiment, the comparison result may be a difference between the audio parameter of the current voice signal and the audio parameter preset at the local terminal, which is obtained by comparing the audio parameter of the current voice signal with the audio parameter preset at the local terminal, after the local terminal obtains the difference, the opposite-terminal to-be-adjusted range to which the difference belongs is determined, and the opposite-terminal to-be-adjusted range to which the difference belongs is determined as the first voice adjustment policy. The range to be adjusted of the opposite end is preset by the local end, for example, the range to be adjusted of the opposite end is less than or equal to minus 6dB, and is greater than minus 6dB and less than minus 3 dB.
In another embodiment, the first voice adjustment strategy comprises a volume adjustment strategy and/or a noise reduction strategy for improving the quality of a subsequent voice signal sent by an opposite end to a local end, and the first voice adjustment strategy is obtained from a trigger control for improving the voice quality. The triggering control can be added to the call interface, and the triggering control is triggered by the user according to the answering effect of the current voice signal and generates the first voice adjustment strategy.
Fig. 2 is a second flow chart of a method for adjusting call quality according to an embodiment of the present invention, and as shown in fig. 2, the method may further include:
step S201: and the local terminal receives a second voice adjustment strategy which is sent by the opposite terminal and used for improving the quality of a subsequent voice signal sent by the local terminal to the opposite terminal.
Step S202: and the local terminal adjusts the subsequent voice signals sent to the opposite terminal by the local terminal according to the second voice adjustment strategy and sends the subsequent voice signals to the opposite terminal.
In one embodiment, the second voice adjustment policy is a local to-be-adjusted range. In this case, the step S201 of this embodiment includes: and the local terminal determines the audio parameters corresponding to the local terminal to-be-adjusted range and adjusts the subsequent voice signals sent to the opposite terminal by the local terminal according to the audio parameters corresponding to the local terminal to-be-adjusted range.
In another embodiment, the second voice adjustment policy includes a volume adjustment policy and/or a noise reduction policy, in this case, the step S201 of this embodiment includes: and increasing or decreasing the volume of the subsequent voice signal sent by the local terminal to the opposite terminal, and/or performing noise reduction processing on the subsequent voice signal sent by the local terminal to the opposite terminal.
In order to describe the interaction process of message feedback and response between the two parties, the following description is made in detail with reference to fig. 3 and 4.
Fig. 3 is a schematic flowchart of a general idea of adjusting call quality according to an embodiment of the present invention, and as shown in fig. 3, the process may include:
step S301: when the two parties are in a conversation, the sound analysis module of the receiver (or the local terminal) analyzes the received sound. And the comparison module of the receiver (or the local terminal) compares the collected audio parameters of the call sound with the audio parameters of the typical call sound (or the audio parameters preset by the local terminal) stored in the terminal in advance.
In this embodiment, the typical call sound means that the terminal can clearly hear the sound of the other party without any special processing in most scenes, and the audio parameter of this sound is preset in the terminal in advance.
Step S302: through comparison of the comparison value (or the comparison result) and the preset value (-6dB, -3dB, 0dB or 3dB), each preset value corresponds to one group of audio parameters, and the receiver (or the home terminal) sends a message (or a feedback message or a voice message packet carrying a first voice adjustment strategy) matched with the preset value so as to inform the receiver to adjust the audio parameters for responding and improve the receiving effect.
In this embodiment, the predetermined value may form an interval range, taking the loudness of the call sound as an example, where the audio parameter is less than or equal to-6 dB represents that the loudness of the audio received by the receiving party (or the home terminal) is too small, and the receiving party (or the opposite terminal) needs to boost the loudness by 6dB and then sends the audio; the loudness of the audio received by the receiver (or the local terminal) is slightly small when the loudness is more than-6 dB and less than or equal to-3 dB, and the audio is sent after the loudness of the other party (or the opposite terminal) is increased by 3 dB; the loudness of the audio received by the receiver (or the local terminal) is moderate when the loudness is more than-3 dB and less than or equal to 3dB, and the receiver (or the opposite terminal) does not need to process; greater than 3dB means that the listening party (or the local terminal) feels that the loudness of the audio is too great, and the other party (or the opposite terminal) needs to send the audio after reducing the loudness by 3 dB.
In another embodiment, the adjusted audio parameters are not limited to the adjustment of the audio parameters (e.g., loudness) of the speech sounds, but may be the adjustment of the noise suppression parameters, or both. The specific requirements may be determined according to the definitions of the message formats by the two parties to the call.
Step S303: after receiving the feedback message, the opposite side (or the opposite side) analyzes the feedback voice message packet, adjusts the voice signal according to the audio parameter corresponding to the preset value and then responds back to realize the requirement of the receiver (or the local side) on the voice signal.
The embodiment of the invention feeds back the current conversation state of the opposite side through a message mechanism, and informs the opposite side of responding the voice after the audio parameter is adjusted, thereby achieving the purpose of improving the answering effect. That is to say, the audio parameters of typical call sound are preset in the terminal, the sound analysis module analyzes the received voice information, the comparison module compares the preset sound with the audio parameters of the received sound, according to the corresponding relation between the comparison result and the preset value, a corresponding voice message packet is sent to the opposite party, the opposite party responds the voice after the audio parameters are adjusted according to the analysis value, and therefore the purpose of improving the answering effect of the opposite party is achieved by adjusting the audio parameters of the source end.
Fig. 4 is a schematic working block diagram of a feedback triggering process for adjusting call quality according to an embodiment of the present invention, and as shown in fig. 4, the process may include:
step S401: the terminal is internally pre-stored with audio parameters of the prior call sound, wherein the typical call sound means that the terminal can clearly hear the sound of the other party without any special processing in most scenes.
Step S402: the sound analysis module analyzes the received sound to obtain the audio parameters of the call sound.
Steps S403 and S404: and the comparison module compares the collected audio parameters of the call sound with the audio parameters of the typical call sound stored in the terminal in advance. By comparing the comparison value with a predetermined value (e.g., -6dB, -3dB, 0dB, or 3dB), the listener sends a message matching the predetermined value to inform the other party to adjust the audio parameters in response. Wherein each predetermined value corresponds to a set of audio parameters.
The voice analysis module of the receiver analyzes the received voice, the comparison module compares the received voice with the audio parameters of the typical voice preset in the terminal in advance, and corresponding voice packet messages are sent to the receiver according to the corresponding relation between the comparison result and the preset value. The opposite side analyzes the voice packet message (or voice message) and responds the voice response after correspondingly adjusting the audio parameters. Therefore, through a message feedback mechanism of the system, the opposite side adaptively adjusts the audio parameters to respond according to the message request of the opposite side, and the conversation effect is improved. In an extensible manner, the adjustment of the audio parameters is not limited to the adjustment of the audio parameters of the call sound, but may be performed on the parameters of noise suppression, or both. The receiver and the opposite party are a relative concept, roles can be exchanged in the call system, and the call effect of the two parties is improved by adaptively adjusting the frequency parameters through a message feedback mechanism.
The following takes an AMR-WB (Adaptive Multi-Rate-Wideband) speech frame structure as an example to illustrate the format of a speech packet.
As shown in fig. 5 and fig. 6, AMR-WB includes 16 encoding modes, where 10 to 13 in the Frame Type Index are For future use fields, and 10, 11, 12, and 13 may be respectively defined to represent voice message packets with predetermined values of-6 dB, -3dB, 0dB, and 3dB, and the voice message packets are packed and then sent to the other party. As shown in fig. 7, the process may include:
step S701: the opposite party (or the opposite end) receives the voice message packet.
Step S702: the opposite side analyzes the received voice message packet.
Step S703: and judging whether the Frame Type Index is 10, 11, 12 or 13, if so, executing step S704, otherwise, executing step S705.
Step S704: if the Frame Type Index is 10, 11, 12 or 13, determining that the voice packet is a message packet requiring to adjust the audio parameter, adjusting the audio parameter of the response according to the value of the Frame Type Index, and then performing voice response, that is, feeding back the sound which increases or decreases the audio parameter according to the predefined.
For example, the Frame Type Index is 10, and since 10 represents a predetermined value of-6 dB, it indicates that the range to be adjusted of the opposite end to which the comparison value of the audio parameter of the call sound received by the receiving party and the audio parameter of the preset typical call sound belongs is less than or equal to-6 dB, that is, the loudness of the voice received by the receiving party (or the local end) is too small, the opposite party (or the opposite end) should increase the loudness of the subsequent voice signal sent to the receiving party (or the local end) by 6dB, and then send the subsequent voice signal. Similarly, if the Frame Type Index is 11, since 11 represents a predetermined value of-3 dB, the range to be adjusted of the opposite end to which the comparison value between the audio parameter of the call sound received by the receiving party and the audio parameter of the preset typical call sound belongs is greater than-6 dB and less than or equal to-3 dB, that is, the loudness of the voice received by the receiving party (or the local end) is small, and therefore the opposite party (or the opposite end) should increase the loudness by 3dB for the subsequent voice signal sent to the receiving party (or the local end) and then send the voice.
Step S705: if the Frame Type Index is not 10, 11, 12, or 13, the processing may be performed according to a normal voice packet.
Therefore, the corresponding audio parameter voice is responded to according to the request of the opposite side through a message response mechanism so as to improve the conversation effect.
As described above, the adjustment of the audio parameters is not limited to the adjustment of the call sound audio parameters, and may be performed by adjusting the noise suppression parameters, or both. The specific requirements may be determined according to the definitions of the message formats by the two parties to the call.
It should be noted that the format of the voice packet is not limited to the AMR-WB voice frame structure, but may be other voice frame formats, and the AMR-WB voice frame format is only used as an example here.
As an extended embodiment that can improve user experience, the embodiment of the present invention may add a trigger control, such as a button or a pull-down menu, to the call interface, as shown in the working block diagram of the voice adjustment interface shown in fig. 8. If the receiver feels that the sound of the opposite side is relatively small (or the noise of the environment of the opposite side is relatively large), the receiver can trigger a corresponding button or select a corresponding menu item to send a message to the opposite side to request the opposite side to send the adjusted sound volume or/and the voice subjected to noise reduction processing, and the message format and the feedback response mechanism are as described above and are not repeated.
Fig. 9 is a schematic block diagram of an apparatus for adjusting call quality according to an embodiment of the present invention, where the apparatus may be disposed at a local end, i.e., a listening side, or disposed at an opposite end, i.e., a sound source side. As shown in fig. 9, the following description is made by taking the local terminal as an example, and the apparatus may include:
the processing module 91 is configured to determine, according to the received current voice signal, a first voice adjustment policy for improving quality of a subsequent voice signal sent from the opposite end to the home end, so as to implement functions of the sound analysis module and the comparison module in the foregoing embodiment;
a transceiver module 92, configured to send the first voice adjustment policy to the opposite end, so as to receive a subsequent voice signal that is adjusted and responded by the opposite end according to the first voice adjustment policy.
In an embodiment, the processing module 91 analyzes the current voice signal from the opposite terminal to obtain an audio parameter of the current voice signal, and then determines the first voice adjustment policy according to a comparison result between the audio parameter of the current voice signal and an audio parameter preset at the local terminal.
In this embodiment, the audio parameter may be a parameter of the call sound, such as loudness, a parameter of noise mixed with the call sound, such as intensity of the noise signal, or a combination of the loudness of the call sound and the intensity of the noise signal.
In this embodiment, the comparison result may be a difference between the audio parameter of the current voice signal and the audio parameter preset at the local terminal, which is obtained by comparing the audio parameter of the current voice signal with the audio parameter preset at the local terminal, after the processing module 91 obtains the difference, the processing module determines an opposite-terminal to-be-adjusted range to which the difference belongs, and determines the opposite-terminal to-be-adjusted range to which the difference belongs as the first voice adjustment policy. The range to be adjusted of the opposite end is preset by the local end, for example, greater than-6 dB and less than or equal to-3 dB, greater than-3 dB and less than or equal to 3dB, and the like.
In another embodiment, the first voice adjustment policy includes a volume adjustment policy and/or a noise reduction policy for improving the quality of a subsequent voice signal sent from the opposite end to the local end, and the first voice adjustment policy is obtained by the processing module 91 from a trigger control for improving the voice quality. The triggering control can be added to the call interface, and the triggering control is triggered by the user according to the answering effect of the current voice signal and generates the first voice adjustment strategy.
On the basis of the above embodiment, the transceiver module 92 may also be configured to receive a second voice adjustment policy sent by the opposite end to improve the quality of a subsequent voice signal sent by the local end to the opposite end. Correspondingly, the processing module 91 is further configured to adjust a subsequent voice signal sent by the local terminal to the opposite terminal according to the second voice adjustment policy, and send the subsequent voice signal to the opposite terminal.
In one embodiment, the second voice adjustment policy may be a local to-be-adjusted range. At this time, the processing module 91 determines the audio parameter corresponding to the local end to-be-adjusted range, and adjusts the subsequent voice signal sent by the local end to the opposite end according to the audio parameter corresponding to the local end to-be-adjusted range. For example, if the range to be adjusted by the local terminal is-6 dB and less than or equal to-3 dB, the processing module 91 finds a set of audio parameters corresponding to the range, and adjusts the subsequent voice signal sent by the local terminal to the opposite terminal according to the set of audio parameters.
In another embodiment, the second voice adjustment policy may include a volume adjustment policy and/or a noise reduction policy, where the processing module 91 increases or decreases the volume of the subsequent voice signal sent by the local terminal to the opposite terminal, and/or performs noise reduction processing on the subsequent voice signal sent by the local terminal to the opposite terminal. For example, if the second voice adjustment policy is a noise reduction policy, the processing module 91 performs noise reduction processing on a subsequent voice signal sent by the local terminal to the opposite terminal, and then sends the subsequent voice signal to the opposite terminal via the transceiver module 92.
Fig. 10 is a schematic block diagram of an apparatus for adjusting call quality according to an embodiment of the present invention, where as shown in fig. 10, the apparatus may include: a processor 11 and a memory 12, wherein the memory 12 stores a program capable of being executed on the processor 11 for adjusting the call quality, and the program for adjusting the call quality realizes the steps of the method for adjusting the call quality when being executed by the processor 11.
Embodiments of the present invention may also provide a computer readable medium, on which a program for adjusting call quality is stored, and when the program for adjusting call quality is executed by a processor, the steps of the method for adjusting call quality are implemented.
It will be understood by those of ordinary skill in the art that all or some of the steps of the methods, systems, functional modules/units in the devices disclosed above may be implemented as software, firmware, hardware, and suitable combinations thereof. In a hardware implementation, the division between functional modules/units mentioned in the above description does not necessarily correspond to the division of physical components; for example, one physical component may have multiple functions, or one function or step may be performed by several physical components in cooperation. Some or all of the physical components may be implemented as software executed by a processor, such as a central processing unit, digital signal processor, or microprocessor, or as hardware, or as an integrated circuit, such as an application specific integrated circuit. Such software may be distributed on computer readable media, which may include computer storage media (or non-transitory media) and communication media (or transitory media). The term computer storage media includes volatile and nonvolatile, removable and non-removable media implemented in any method or technology for storage of information such as computer readable instructions, data structures, program modules or other data, as is well known to those of ordinary skill in the art. Computer storage media includes, but is not limited to, RAM, ROM, EEPROM, flash memory or other memory technology, CD-ROM, Digital Versatile Disks (DVD) or other optical disk storage, magnetic cassettes, magnetic tape, magnetic disk storage or other magnetic storage devices, or any other medium which can be used to store the desired information and which can accessed by a computer. In addition, communication media typically embodies computer readable instructions, data structures, program modules or other data in a modulated data signal such as a carrier wave or other transport mechanism and includes any information delivery media as known to those skilled in the art.
By the embodiment of the invention, the answering party can feed back the answering effect to the opposite party by the answering party under any scene, so that the opposite party responds to the improved voice signal, and the conversation quality of the two parties is improved.
Although the present invention has been described in detail hereinabove, the present invention is not limited thereto, and various modifications can be made by those skilled in the art in light of the principle of the present invention. Thus, modifications made in accordance with the principles of the present invention should be understood to fall within the scope of the present invention.
Claims (10)
1. A method for adjusting call quality, the method comprising:
determining a first voice adjustment strategy for improving the quality of a subsequent voice signal sent from an opposite terminal to a local terminal according to a received current voice signal;
and sending the first voice adjustment strategy to the opposite terminal so as to receive a subsequent voice signal which is adjusted and responded by the opposite terminal according to the first voice adjustment strategy.
2. The method of claim 1, wherein determining a first voice adjustment strategy for improving the quality of a subsequent voice signal from an opposite end to the home end according to the received current voice signal comprises:
analyzing the current voice signal to obtain an audio parameter of the current voice signal;
and determining the first voice adjustment strategy according to the comparison result of the audio parameter of the current voice signal and the audio parameter preset at the local terminal.
3. The method of claim 2, wherein the determining the first voice adjustment strategy according to the comparison result between the audio parameter of the current voice signal and the locally preset audio parameter comprises:
comparing the audio parameter of the current voice signal with the audio parameter preset at the home terminal to obtain a difference value between the audio parameter of the current voice signal and the audio parameter preset at the home terminal as a comparison result;
and determining an opposite terminal to-be-adjusted range preset by the local terminal to which the comparison result belongs, and determining the opposite terminal to-be-adjusted range as the first voice adjustment strategy.
4. The method according to claim 1, wherein the first voice adjustment strategy comprises a volume adjustment strategy and/or a noise reduction strategy for improving the quality of a subsequent voice signal sent from the opposite end to the local end, and the first voice adjustment strategy is obtained from a trigger control for improving the voice quality, and the trigger control is triggered by the user according to the listening effect of the current voice signal.
5. The method according to any one of claims 1-4, further comprising:
receiving a second voice adjustment strategy which is sent by the opposite terminal and used for improving the quality of a subsequent voice signal sent by the local terminal to the opposite terminal;
and adjusting subsequent voice signals sent to the opposite terminal by the local terminal according to the second voice adjustment strategy, and sending the subsequent voice signals to the opposite terminal.
6. The method according to claim 5, wherein the second voice adjustment policy is a range to be adjusted by the local terminal, and the adjusting the subsequent voice signal sent by the local terminal to the opposite terminal according to the second voice adjustment policy comprises:
determining audio parameters corresponding to the range to be adjusted of the local terminal;
and adjusting subsequent voice signals sent to the opposite terminal by the local terminal according to the audio parameters corresponding to the range to be adjusted by the local terminal.
7. The method according to claim 5, wherein the second voice adjustment strategy comprises a volume adjustment strategy and/or a noise reduction strategy, and the adjusting the subsequent voice signal sent by the local terminal to the opposite terminal according to the second voice adjustment strategy comprises:
and increasing or decreasing the volume of the subsequent voice signal sent by the local terminal to the opposite terminal, and/or performing noise reduction processing on the subsequent voice signal sent by the local terminal to the opposite terminal.
8. An apparatus for adjusting call quality, the apparatus comprising:
the processing module is used for determining a first voice adjustment strategy for improving the quality of a subsequent voice signal sent from the opposite terminal to the local terminal according to the received current voice signal;
and the transceiver module is used for sending the first voice adjustment strategy to the opposite terminal so as to receive a subsequent voice signal which is adjusted and responded by the opposite terminal according to the first voice adjustment strategy.
9. An apparatus for adjusting call quality, the apparatus comprising: a processor and a memory, wherein the memory stores a program for adjusting call quality, the program being executable on the processor and the program for adjusting call quality implementing the steps of the method for adjusting call quality as claimed in any one of claims 1 to 7 when the program for adjusting call quality is executed by the processor.
10. A computer-readable medium, on which a program for adjusting call quality is stored, which when executed by a processor implements the steps of the method for adjusting call quality according to any one of claims 1 to 7.
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| CN201810978640.8A CN110910896A (en) | 2018-08-27 | 2018-08-27 | Method, device, equipment and computer readable medium for adjusting call quality |
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Citations (8)
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