CN110177176B - Method and device for improving conversation tone quality and mobile terminal - Google Patents
Method and device for improving conversation tone quality and mobile terminal Download PDFInfo
- Publication number
- CN110177176B CN110177176B CN201910444890.8A CN201910444890A CN110177176B CN 110177176 B CN110177176 B CN 110177176B CN 201910444890 A CN201910444890 A CN 201910444890A CN 110177176 B CN110177176 B CN 110177176B
- Authority
- CN
- China
- Prior art keywords
- signal
- channel
- mobile terminal
- voice
- audio
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Active
Links
Images
Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
- H04M1/724—User interfaces specially adapted for cordless or mobile telephones
- H04M1/72448—User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions
- H04M1/72454—User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions according to context-related or environment-related conditions
Abstract
The invention discloses a method, a device and a mobile terminal for improving conversation tone quality, wherein the method comprises the following steps: after establishing voice communication connection, acquiring an audio signal sent by a mobile terminal where a called user is located and answering time of the called user for responding to voice call; determining the type of the audio signal according to the listening time, wherein the type of the audio signal comprises a ring-back signal and a voice signal; if the audio signal is a ring-back signal, calling a first channel to debug the ring-back signal; and if the audio signal is a voice signal, calling a second channel to debug the voice signal. According to the technical scheme, the ring-back signal and the voice signal are distinguished through software, different audio configuration parameters are used for debugging the corresponding signals in different channels, and the voice communication quality is improved.
Description
Technical Field
The invention relates to the technical field of mobile terminals, in particular to a method and a device for improving conversation tone quality and a mobile terminal.
Background
With the development of mobile communication technology and the popularization of intelligent mobile terminals, mobile terminals play an increasingly important role in user life.
As shown in fig. 1, a calling party performs a voice call with a called party in a dial-up call mode, the calling party receives a color ring signal from the called party while the calling party waits for the called party to be connected, and the calling party receives the voice signal during the call after the called party connects the voice call.
However, since the color ring signal and the voice signal have frequency difference, when the color ring signal and the voice signal are modulated through the same channel, the color ring signal generates noise or the quality of the voice signal is sacrificed for the color ring signal.
Disclosure of Invention
In view of the foregoing problems, an object of the embodiments of the present invention is to provide a method, an apparatus and a mobile terminal for improving call tone quality, so as to solve the deficiencies of the prior art.
According to an embodiment of the present invention, a method for improving call quality is provided, the method including:
after establishing voice communication connection, acquiring an audio signal sent by a mobile terminal where a called user is located and answering time of the called user for responding to voice call;
determining the type of the audio signal according to the listening time, wherein the type of the audio signal comprises a ring-back signal and a voice signal;
if the audio signal is a ring-back signal, calling a first channel to debug the ring-back signal;
and if the audio signal is a voice signal, calling a second channel to debug the voice signal.
In the method for improving the call tone quality, the answering time is obtained in the following manner:
receiving an answering signaling sent by the mobile terminal where the called user is located when the mobile terminal answers the voice call;
and analyzing the answering signaling according to the frame structure of the answering signaling to obtain the sending time of the answering signaling, and taking the sending time of the answering signaling as the answering time.
In the method for improving the call tone quality, the answering time is obtained in the following manner:
and acquiring the signaling receiving time when the answering signaling is received, and taking the signaling receiving time as the answering time.
In the method for improving call quality, the determining the type of the audio signal according to the listening time includes:
and taking the audio signal of a first time interval before the answering time as a ring-back signal and taking the audio signal of a second time interval after the answering time as a voice signal.
In the method for improving the call tone quality, a corresponding relationship between the channel and the audio configuration parameter is pre-stored in the mobile terminal where the calling user is located, and the method further includes:
the first channel and the second channel are pre-established before voice communication connection is established;
and configuring the corresponding channels according to the audio configuration parameters corresponding to the channels.
In the method for improving the call tone quality, the audio configuration parameter includes amplitude information and frequency information;
invoking the first channel or the second channel to debug the corresponding audio signal comprises:
and debugging the amplitude of the audio signal entering the channel corresponding to the audio configuration parameter according to the amplitude information and/or debugging the frequency of the audio signal entering the channel corresponding to the audio configuration parameter according to the frequency information.
In the method for improving the call tone quality, the "debugging the frequency of the audio signal entering the channel corresponding to the audio configuration parameter according to the frequency information" includes:
and debugging the frequency of the preset resonance point in the audio signal entering the channel corresponding to the audio configuration parameter according to the frequency information.
According to another embodiment of the present invention, there is provided an apparatus for improving quality of a call sound, the apparatus including:
the system comprises an acquisition module, a voice communication module and a voice communication module, wherein the acquisition module is used for acquiring an audio signal sent by a mobile terminal where a called user is located and answering time of the called user responding to a voice call after establishing voice communication connection;
the determining module is used for determining the type of the audio signal according to the answering time, wherein the type of the audio signal comprises a ring-back signal and a voice signal;
the first debugging module is used for calling a first channel to debug the ring-back signal when the audio signal is the ring-back signal;
and the second debugging module is used for calling a second channel to debug the voice signal when the audio signal is the voice signal.
According to another embodiment of the present invention, a mobile terminal is provided, which includes a memory for storing a computer program and a processor for executing the computer program to make the mobile terminal execute the above method for improving the quality of the call sound.
According to still another embodiment of the present invention, there is provided a computer-readable storage medium storing the computer program used in the mobile terminal described above.
The technical scheme provided by the embodiment of the disclosure can have the following beneficial effects:
according to the method, the device and the mobile terminal for improving the call tone quality, the ring-back signal and the voice signal are distinguished through the answering time, and are respectively and independently debugged in different channels, so that the debugged ring-back signal and the debugged voice signal are not distorted, an earphone can play the audio signal without damage, and the voice call quality is improved.
In order to make the aforementioned and other objects, features and advantages of the present invention comprehensible, preferred embodiments accompanied with figures are described in detail below.
Drawings
In order to more clearly illustrate the technical solution of the present invention, the drawings needed in the embodiments will be briefly described below, it should be understood that the following drawings only illustrate some embodiments of the present invention, and therefore should not be considered as limiting the scope of the present invention, and for those skilled in the art, other related drawings can be obtained according to the drawings without inventive efforts.
Fig. 1 is a flow chart illustrating a conventional voice call process.
Fig. 2 shows a frequency diagram of a ring-back signal and a speech signal over a time interval.
Fig. 3 shows a frequency response graph of a ring-back signal and a speech signal over a time interval.
Fig. 4 is a flowchart illustrating a method for improving call quality according to a first embodiment of the present invention.
Fig. 5 is a signaling diagram illustrating a voice call process according to a first embodiment of the present invention.
Fig. 6 shows a frequency diagram of the debugged ring-back signal and the voice signal in a time interval provided by the first embodiment of the present invention.
Fig. 7 is a flowchart illustrating a method for improving call quality according to a second embodiment of the present invention.
Fig. 8 is a schematic structural diagram illustrating an apparatus for improving quality of call sound according to a third embodiment of the present invention.
Description of the main element symbols:
400-a device for improving the tone quality of the call; 410-an obtaining module; 420-a determination module; 430-a first debug module; 440-second debug module.
Detailed Description
The technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are only a part of the embodiments of the present invention, and not all of the embodiments. The components of embodiments of the present invention generally described and illustrated in the figures herein may be arranged and designed in a wide variety of different configurations. Thus, the following detailed description of the embodiments of the present invention, presented in the figures, is not intended to limit the scope of the invention, as claimed, but is merely representative of selected embodiments of the invention. All other embodiments, which can be derived by a person skilled in the art from the embodiments of the present invention without making any creative effort, shall fall within the protection scope of the present invention.
In the embodiment of the present invention, the method for improving call tone quality is applied to various voice calls that need to be performed through a mobile terminal, including but not limited to the following application scenarios: various video and voice chat calls such as dial-up voice call, video call, WeChat and the like.
When a calling party and a called party carry out voice communication, a mobile terminal where the calling party is located first initiates a voice communication connection request to a mobile terminal where the called party is located, after a voice communication connection is initiated, the calling party can also hear a certain sound in the process that the calling party waits for the called party to respond to the voice communication, and the sound heard by the calling party in the waiting process is the ring-back signal in the embodiment of the invention. The ring-back signal has different expressions in different operators, such as mobile operators, and is called a color ring signal; at the Unicom operator, the ring-back signal is called a ring-back signal; in telecommunication operators, the ring-back signal is called a melody signal, and with the rapid development of mobile communication technology, the ring-back signal can also have more expressions.
In the voice communication process, the ring-back signal is usually a personalized service provided by an operator to a called user, after the called user opens the ring-back personalized service, the ring-back signal is generated at a local exchange of the called user, and then the ring-back signal is transmitted to a terminal where a calling user is located through an established voice communication connection, and the ring-back signal is played to the calling user by a playing device of the terminal where the calling user is located. In addition, in addition to the above-described color ring, dazzle ring, and melody ring, a sound of "beep" while waiting for a voice call to be connected may be included in the ringback signal. All the embodiments of the present invention take the case that the called subscriber is provided with a ring-back as an example for explanation.
In order to better embody the technical scheme of the present application, the prior art is briefly described. The process of voice communication includes: the mobile terminal of the calling user can establish voice communication connection with the mobile terminal of the called user in a dial-up calling mode, in the process of waiting for the mobile terminal of the called user to respond to the voice call, the audio signal received by the mobile terminal of the calling user is a ring-back signal, and after the mobile terminal of the called user responds to the voice communication connection sent by the mobile terminal of the calling user, the mobile terminal of the calling user and the mobile terminal of the called user can mutually transmit the voice signal in the call process (the voice signal refers to the voice signal for voice interaction between the calling user and the called user after the call is connected). Therefore, in the mobile terminal where the calling subscriber is located, all the received audio signals include the ring-back signal and the voice signal. As shown in fig. 2, the S1 part signal is a ring-back signal, the S2 part signal is a voice signal, and the frequency response of the ring-back signal S1 is not consistent with the frequency response of the voice signal S2, as shown in fig. 3, the frequency response of the ring-back signal S1 is 3-5dB higher than that of the voice signal S2 at low frequency (200-500Hz), so when the ring-back signal and the voice signal are debugged in the same channel, in order to be compatible with the audio signals with two different frequency responses, the problem that the ring-back signal generates noise or the voice signal quality is sacrificed for the compatibility of the ring-back signal is often caused. Therefore, it is a difficult problem to solve the problem of noise generation or poor quality of voice signals in the ring back signal.
In the scheme, the voice signal and the ring-back signal are distinguished by the called user responding to the turn-on time of the voice call, and the voice signal and the ring-back signal are respectively debugged by using different channels, so that the optimized call voice quality is achieved.
Example 1
Fig. 4 is a flowchart illustrating a method for improving call quality according to a first embodiment of the present invention.
The method for improving the conversation tone quality comprises the following steps:
in step S110, after the voice communication connection is established, an audio signal sent from the mobile terminal where the called user is located and the answering time of the called user for responding to the voice call are obtained.
When a calling party and a called party carry out voice communication, a mobile terminal where the calling party is located first establishes voice communication connection with a mobile terminal where the called party is located, and the audio signals are all sound information received after the mobile terminal where the calling party is located establishes voice communication connection with the mobile terminal where the called party is located in a calling mode or the like.
Fig. 5 is a simplified signaling diagram of a voice communication process. The voice communication process is illustrated only by simple steps, and it should be noted that the voice communication process further includes more detailed and complicated steps, and for convenience of describing the embodiment, more detailed and complicated steps are not described in detail here.
The voice communication process between the calling user and the called user comprises the following steps:
A1. a mobile terminal where a calling user is located initiates a voice communication connection request to a mobile terminal where a called user is located;
A2. after receiving the voice communication connection request, the mobile terminal of the called user sends an ACK (acknowledgement) message to the mobile terminal of the calling user;
A3. after knowing that the mobile terminal where the called user is located receives the voice communication connection request, the mobile terminal where the calling user is located establishes voice communication connection with the mobile terminal where the called user is located;
A4. after the voice communication connection is established, the mobile terminal where the called user is located transmits a ring-back signal to the mobile terminal where the calling user is located through the switch;
A5. after the called user answers the voice call, the mobile terminal of the called user sends an answering instruction to the mobile terminal of the calling user;
A6. after receiving the answering instruction, the mobile terminal of the calling user sends an ACK (acknowledgement) message to the mobile terminal of the called user;
A7. after the called user connects the voice call, the calling user and the called user start to carry out voice communication, and the mobile terminal where the calling user is located and the mobile terminal where the called user is located start to transmit voice signals mutually;
A8. after the voice call is finished, a calling user or a called user hangs up the voice call, and as an implementation mode, a mobile terminal where the calling user is located sends a hang-up instruction to a mobile terminal where the called user is located;
as another embodiment, the A8 step may further be: after the voice call is finished, the mobile terminal where the called user is located sends a hang-up instruction to the mobile terminal where the calling user is located.
A9. When the calling user hangs up the voice call, as an implementation mode, the mobile terminal where the called user is located receives the hang-up instruction and sends an ACK (acknowledgement) message to the mobile terminal where the calling user is located;
as another embodiment, the a9 step may further be: when the called user hangs up the voice call, the mobile terminal of the calling user receives the hang-up instruction and sends an ACK (acknowledgement) message to the mobile terminal of the called user;
A10. and releasing the voice communication connection after the mobile terminal where the calling user is located receives the ACK message of the hangup instruction from the mobile terminal where the called user is located.
In connection with the voice communication process in fig. 5, as an embodiment, the listening time is obtained by:
receiving an answering signaling sent by the mobile terminal where the called user is located when the mobile terminal answers the voice call; and analyzing the answering signaling according to the frame structure of the answering signaling to obtain the sending time of the answering signaling, and taking the sending time of the answering signaling as the answering time.
Specifically, in step a5, when the mobile terminal where the called user is located responds to the voice call sent by the mobile terminal where the calling user is located (for example, the mobile terminal where the called user is located answers the call of the mobile terminal where the calling user is located), the mobile terminal where the called user is located sends an answering signaling to the mobile terminal where the calling user is located at time t9 after answering the voice call, where the answering signaling at least includes sending time of the answering signaling. After the mobile terminal where the calling user is located receives the answering signaling at the time t10, the answering signaling is analyzed according to the frame format of the signaling to obtain information of each field in the answering signaling, the field where the sending time is located is determined according to the meaning of each field preset in the frame format, the sending time t9 of the answering signaling is obtained according to the content of the field where the sending time is located, and the sending time t9 is used as the answering time of the mobile terminal where the called user is located for responding to the voice call.
As another embodiment, the listening time may be obtained by:
and acquiring the signaling receiving time for receiving the answering signaling sent by the mobile terminal of the called user when responding to the voice call, and taking the signaling receiving time as the answering time.
Specifically, since the message transmission time between the mobile terminal where the called user is located and the mobile terminal where the calling user is located is relatively fast, in order to quickly obtain the receiving time, after the mobile terminal where the called user is located sends the answering command to the mobile terminal where the calling user is located in step a5, the mobile terminal where the calling user is located receives the answering command at time t10, and the mobile terminal where the calling user is located can directly use the signaling receiving time t10 when receiving the answering signaling sent by the mobile terminal where the called user is located as the answering time when the mobile terminal where the called user is located responds to the voice call.
As another embodiment, the listening time may be obtained by:
and determining the answering time according to the interruption time of the audio signal.
Specifically, in step a6, after receiving the answering command sent by the called user, the mobile terminal of the calling user returns an ACK confirmation message to the mobile terminal of the called user, and starts to perform voice communication with the called user. When the mobile terminal where the called user is located responds to the voice call sent by the mobile terminal where the calling user is located, the audio signal received by the mobile terminal where the calling user is located is temporarily interrupted, the mobile terminal where the calling user is located records the interruption starting time, timing is started from the interruption starting time until the next voice signal is received, timing is stopped, the timing is compared with a preset interruption threshold, and if the timing is greater than or equal to the interruption threshold, the interruption starting time corresponding to the timing is used as the answering time of the mobile terminal where the called user is located responding to the voice call.
In step S120, the type of the audio signal is determined according to the listening time.
The types of the audio signals comprise a ring-back signal in a waiting answering process and a voice signal in a voice communication process after answering.
After the audio signal is acquired, how to determine the type of the audio signal becomes an important problem to be solved next.
Further, the "determining the type of the audio signal according to the listening time" includes:
and taking the audio signal of a first time interval before the answering time as a ring-back signal and taking the audio signal of a second time interval after the answering time as a voice signal.
As an embodiment, in order to increase the processing speed of the method for improving the call quality, all audio signals before the listening time may be used as a ring-back signal, and all audio signals after the listening time may be used as a voice signal.
Referring to fig. 5, for example, when the listening time is t9, the time interval t1 to t9 before t9 may be used as a first time interval, and all audio signals in the first time interval may be used as a ring-back signal; the time interval t9 to t20 after t9 is taken as a second time interval, and all audio signals in the second time interval are taken as voice signals.
As another embodiment, in order to accurately extract the ringback signal and the voice signal, all audio signals during the transmission of the ringback signal before the listening time may be used as the ringback signal, and all audio signals during the voice communication between the two parties after the listening time may be used as the voice signal.
Referring to fig. 5, the same example of the answering time t9 is described, since it takes a while (t 1-t 6) to establish the voice communication connection between the mobile terminal where the calling subscriber is located and the mobile terminal where the called subscriber is located (step a 1-step A3 in fig. 4), and it also takes a while (t 15-t 20) to hang up the voice communication connection (step A8-step a10 in fig. 4), in fact, except for the time consumed by various request messages, confirmation messages, connection establishment procedures, instruction messages, and the like, the calling subscriber hears the ring back signal in step a4 after the voice communication connection is established, and the voice communication between the calling subscriber and the called subscriber is performed in step a7. Therefore, the time interval t 7-t 9 before the listening time t9 can be used as a first time interval, and all audio signals in the first time interval can be used as ring-back signals; and taking a time interval t 9-t 15 after the listening time t9 as a second time interval, and taking all audio signals in the second time interval as voice signals.
In step S130, if the audio signal is a ring-back signal, a first channel is called to debug the ring-back signal.
After the voice signal and the ring-back signal are distinguished, the voice signal and the ring-back signal are debugged through different channels respectively, so that the problem that the signal quality is damaged due to the same channel is avoided.
Specifically, a preconfigured first channel is called, and the ring-back signal is debugged in the first channel.
For example, as shown in fig. 6, since the low frequency component of the ring-back signal is higher than the low frequency component of the voice signal, the parameters such as the amplitude and/or frequency of the ring-back signal S1 are suppressed, and the gain is reduced to make the debugged ring-back signal S1 reach the lossless playing condition of the handset, when the debugged ring-back signal is input to the handset, the handset can play the ring-back signal losslessly.
For example, if the lossless playback condition of the handset is ± 1dB, and if the frequency range of the ringback signal is ± 3dB, the frequency component of the ringback signal is suppressed to the range of ± 1dB after the ringback signal is debugged in the first channel, and when the ringback signal is played through the handset, the signal quality is not damaged, and noise is not generated.
In step S140, if the audio signal is a voice signal, a second channel is called to debug the voice signal.
Specifically, a pre-configured second channel is called, and the voice signal is debugged in the second channel.
For example, as shown in fig. 6, parameters such as amplitude and/or frequency of the voice signal S2 are debugged in the second channel, so that the debugged voice signal S2 meets the lossless playback condition of the handset, and when the debugged voice signal is input to the handset, the handset can play the voice signal without loss.
For example, if the lossless playback condition of the handset is ± 1dB, and if the frequency range of the voice signal is ± 1dB, the frequency component and amplitude of the voice signal do not need to be compressed in the second channel, and when the voice signal is played through the handset, the signal quality of the voice signal is not damaged, and noise is not generated.
Example 2
Fig. 7 is a flowchart illustrating a method for improving call quality according to a second embodiment of the present invention.
The method for improving the conversation tone quality comprises the following steps:
in step S210, before establishing the voice communication connection, the first channel and the second channel are established in advance.
In this embodiment, as shown in fig. 5, in step a 1: the method comprises the steps that a first channel and a second channel are established before a mobile terminal where a calling user is located initiates a voice communication connection request to a mobile terminal where a called user is located.
In some other embodiments, in conjunction with fig. 5, at step a 1: after the mobile terminal where the calling party is located initiates a voice communication connection request to the mobile terminal where the called party is located, step a 3: before establishing voice communication connection between a mobile terminal where a calling user is located and a mobile terminal where a called user is located, a first channel and a second channel are established.
The channel is a link established between the mobile terminal of the calling user and the mobile terminal of the called user for voice interaction, such as a Radio Resource Control (RRC) link.
In addition, a plurality of channels can exist between the mobile terminal where the calling user is located and the mobile terminal where the called user is located, and any two channels can be selected from the plurality of channels to be used as the first channel and the second channel respectively.
In step S220, the corresponding channels are configured according to the audio configuration parameters corresponding to the channels.
After the first channel and the second channel are established, the first channel is configured according to the characteristics of the signal entering the first channel, and the second channel is configured according to the characteristics of the signal entering the second channel.
Specifically, the mobile terminal where the calling party is located stores the corresponding relationship between the channel and the audio configuration parameter in advance.
For example, the correspondence between the pre-stored channels and audio configuration parameters is shown in the following table.
In the above table, the corresponding audio configuration parameter in the first channel is a first audio configuration parameter; and the audio configuration parameter corresponding to the second channel is a second audio configuration parameter.
In addition, the audio configuration parameters have a corresponding relationship with the type of the audio signal.
| Audio signal | Audio configuration parameters |
| Ring back signal | First audio configuration parameter |
| Speech signal | Second audio configuration parameter |
In the above table, the ring-back signal corresponds to a first audio configuration parameter; the voice signal corresponds to the second audio configuration parameter.
In this embodiment, as can be seen from the above correspondence, the first channel is configured according to the first audio configuration parameter, and the ring-back signal is debugged according to the configured first channel; and configuring the second channel through the second audio configuration parameters, and debugging the voice signal through the configured second channel.
In some other embodiments, the first channel may correspond to the second audio configuration parameter, and the second channel may correspond to the first audio configuration parameter. The first channel is configured through the second audio configuration parameter, the voice signal is debugged through the configured first channel, the second channel is configured through the first audio configuration parameter, and the ring-back signal is debugged through the configured second channel, which is not limited herein.
It should be noted that, in order to achieve lossless playback of both the ring-back signal and the voice signal and improve the voice communication quality, the first audio configuration parameter is different from the second audio configuration parameter.
For example, after the ring-back signal is debugged through the channel after the first audio configuration parameter is debugged, the frequency component of the ring-back signal can be compressed, so that the lossless playing condition of the receiver is met. After the voice signal is debugged through the channel debugged by the second audio configuration parameter, the frequency component of the voice signal can be debugged, and the lossless playing condition of the receiver is met.
In step S230, after the voice communication connection is established, the audio signal sent from the mobile terminal where the called user is located and the answering time of the called user responding to the voice call are obtained.
This step is the same as step S110, and is not described herein again.
In step S240, the type of the audio signal is determined according to the listening time.
This step is the same as step S120, and is not described herein again.
In step S250, the amplitude of the audio signal entering the channel corresponding to the audio configuration parameter is debugged according to the amplitude information and/or the frequency of the audio signal entering the channel corresponding to the audio configuration parameter is debugged according to the frequency information.
Further, the first audio configuration parameter at least comprises first amplitude information and first frequency information;
invoking the first channel to debug the ringback signal comprises:
and debugging the amplitude of the ring-back signal entering the first channel according to the first amplitude information and/or debugging the frequency of the ring-back signal entering the first channel according to the first frequency information.
Wherein, the debugging the amplitude of the ring-back signal entering the first channel according to the first amplitude information comprises:
judging whether the amplitude information of the ring-back signal entering the first channel exceeds the first amplitude information or not, if so, suppressing the amplitude information of the ring-back signal to ensure that the amplitude information of the ring-back signal does not exceed the first amplitude information; if the amplitude information degree of the ring-back signal does not exceed the first amplitude information, the amplitude information of the ring-back signal is not adjusted.
Debugging the frequency of the ring-back signal entering the first channel according to the first frequency information comprises
Judging whether the frequency information of a preset resonance point in the ring-back signal exceeds first frequency information, and if the frequency information of the preset resonance point in the ring-back signal exceeds the first frequency information, reducing the frequency information of the preset resonance point in the ring-back signal to ensure that the frequency information does not exceed the first frequency information; if the frequency information of the preset resonance point in the ring-back signal does not exceed the first frequency information, the frequency information of the ring-back signal is not adjusted.
It is noted that the predetermined resonance point in the ringback signal has been determined after the mobile terminal has been designed.
After the amplitude and/or the frequency of the ring-back signal are debugged, the debugged ring-back signal is transmitted to the receiver through the first channel in a lossless manner.
Further, the second audio configuration parameter at least comprises second amplitude information and second frequency information;
invoking the second channel to debug the speech signal comprises:
and debugging the amplitude of the voice signal entering the second channel according to the second amplitude information and/or debugging the frequency of the voice signal entering the second channel according to the second frequency information.
Wherein, the debugging the amplitude of the voice signal entering the second channel according to the second amplitude information comprises:
judging whether the amplitude information of the voice signal entering the second channel exceeds the second amplitude information or not, if so, suppressing the amplitude information of the voice signal to ensure that the amplitude information of the voice signal does not exceed the second amplitude information; and if the amplitude information degree of the voice signal does not exceed the second amplitude information, not adjusting the amplitude information of the voice signal.
Debugging the frequency of the voice signal entering the second channel according to the second frequency information comprises
Judging whether the frequency information of a preset resonance point in the voice signal exceeds second frequency information or not, and if the frequency information of the preset resonance point in the voice signal exceeds the second frequency information, reducing the frequency information of the preset resonance point in the voice signal to ensure that the frequency information does not exceed the second frequency information; if the frequency information of the preset resonance point in the voice signal does not exceed the second frequency information, the frequency information of the voice signal is not adjusted.
It is noted that the predetermined resonance point in the speech signal has been determined after the mobile terminal has been designed.
After the amplitude and/or the frequency of the voice signal are debugged, the debugged voice signal is transmitted to the receiver through the second channel in a lossless mode.
Example 3
Fig. 8 is a schematic structural diagram illustrating an apparatus for improving quality of call sound according to a third embodiment of the present invention.
The device 400 for improving the quality of the call sound comprises an obtaining module 410, a determining module 420, a first debugging module 430 and a second debugging module 440.
The obtaining module 410 is configured to obtain an audio signal sent by a mobile terminal where a called user is located and answering time of the called user responding to a voice call after establishing a voice communication connection.
A determining module 420, configured to determine a type of the audio signal according to the listening time, where the type of the audio signal includes a ring-back signal and a voice signal.
The first debugging module 430 is configured to invoke a first channel to debug the ringback signal when the audio signal is the ringback signal.
The second debugging module 440 is configured to invoke a second channel to debug the voice signal when the audio signal is a voice signal.
The invention also provides a mobile terminal which can comprise a smart phone, a tablet computer and the like.
The mobile terminal comprises a memory and a processor, wherein the memory mainly comprises a program storage area and a data storage area, wherein the program storage area can store an operating system, an application program required by at least one function and the like; the storage data area may store data created according to the use of the mobile phone, and the like. Further, the memory may include high speed random access memory, and may also include non-volatile memory, such as at least one magnetic disk storage device, flash memory device, or other volatile solid state storage device.
The processor is configured to run the computer program stored in the memory to enable the mobile terminal to execute the method for improving the call sound quality or the functions of each module in the device for improving the call sound quality in the foregoing embodiments.
Alternatively, the processor may include one or more processing units; preferably, the processor may be integrated with an application processor, which primarily handles operating systems, user interfaces, application programs, and the like. The processor may or may not be integrated with the modem processor.
In addition, the mobile terminal may further include: a Radio Frequency (RF) circuit, an input unit, a display unit, a shooting unit, an audio circuit, a wireless fidelity (WiFi) module, and a power supply. The input unit may include a touch panel and may include other input devices, and the display unit may include a display panel.
The radio frequency circuit is used for receiving and sending wireless signals, and particularly comprises a radio frequency receiving circuit and a radio frequency sending circuit, and the radio frequency circuit mainly comprises an antenna, a wireless switch, a receiving filter, a frequency synthesizer, high-frequency amplification, a receiving local oscillator, frequency mixing, intermediate frequency, a transmitting local oscillator, power amplifier control, a power amplifier and the like.
The input unit may be used to receive input numeric or character information and generate key signal inputs related to user settings and function control of the mobile terminal 700. Specifically, the input unit may include a touch panel and other input devices. The touch panel, also called a touch screen, may collect touch operations of a user (for example, operations of the user on or near the touch panel using any suitable object or accessory such as a finger, a stylus, etc.) and drive the corresponding connection device according to a preset program. Alternatively, the touch panel may include two parts, a touch detection device and a touch controller. The touch detection device detects the touch direction of a user, detects a signal brought by touch operation and transmits the signal to the touch controller; the touch controller receives touch information from the touch detection device, converts the touch information into touch point coordinates, sends the touch point coordinates to the processor, and can receive and execute commands sent by the processor. In addition, the touch panel may be implemented in various types such as a resistive type, a capacitive type, an infrared ray, and a surface acoustic wave. The input unit may include other input devices in addition to the touch panel. In particular, other input devices may include, but are not limited to, one or more of a physical keyboard, function keys (such as volume control keys, switch keys, etc.), a trackball, a mouse, a joystick, and the like.
The display unit may be used to display information input by a user or information provided to the user, and various menus and interfaces of the mobile terminal, such as a game interface. The display unit may include a display panel. Alternatively, the Display panel may be configured in the form of a Liquid Crystal Display (LCD), an Organic Light-Emitting Diode (OLED), or the like. Further, the touch panel may cover the display panel, and when the touch panel detects a touch operation thereon or nearby, the touch panel transmits the touch operation to the processor to determine the type of the touch event, and then the processor provides a corresponding visual output on the display panel according to the type of the touch event. Although the touch panel and the display panel are two separate components to implement the input and output functions of the mobile phone, in some embodiments, the touch panel and the display panel may be integrated to implement the input and output functions of the mobile phone.
The shooting unit is used for collecting image information in an imaging range. Specifically, the photographing unit may be a camera, and the camera may include a photosensitive Device, which may include but is not limited to a CCD (Charge Coupled Device) and a CMOS (Complementary Metal-Oxide Semiconductor). The photosensitive device converts light change information into electric charges, the converted electric charges are converted into digital signals through analog-to-digital conversion, and the digital signals are stored by a flash memory or a built-in hard disk card in the shooting unit after being compressed, so that the stored digital signals can be transmitted to a processor, and the processor processes the digital signals according to requirements or instructions (such as displaying images, modifying images and the like).
The audio circuitry may provide an audio interface between a user and the mobile terminal.
WiFi belongs to short-distance wireless transmission technology, and a mobile terminal can help a user to receive and send e-mails, browse webpages, access streaming media and the like through a wireless fidelity module (a WiFi module described below), and provides wireless broadband internet access for the user. It is understood that the WiFi module is not an essential component of the mobile terminal and can be omitted entirely as needed within the scope not changing the essence of the invention.
The power supply can be logically connected with the processor through the power management system, so that the functions of managing charging, discharging, power consumption management and the like are realized through the power management system.
Those skilled in the art will appreciate that the above-described mobile terminal architecture is not intended to be limiting of mobile terminals and may include more or fewer components than those shown, or some components may be combined, or a different arrangement of components.
The present embodiment also provides a computer-readable storage medium for storing the computer program used in the above-mentioned mobile terminal.
In the embodiments provided in the present application, it should be understood that the disclosed apparatus and method can be implemented in other ways. The apparatus embodiments described above are merely illustrative and, for example, the flowchart and block diagrams in the figures illustrate the architecture, functionality, and operation of possible implementations of apparatus, methods and computer program products according to various embodiments of the present invention. In this regard, each block in the flowchart or block diagrams may represent a module, segment, or portion of code, which comprises one or more executable instructions for implementing the specified logical function(s). It should also be noted that, in alternative implementations, the functions noted in the block may occur out of the order noted in the figures. For example, two blocks shown in succession may, in fact, be executed substantially concurrently, or the blocks may sometimes be executed in the reverse order, depending upon the functionality involved. It will also be noted that each block of the block diagrams and/or flowchart illustration, and combinations of blocks in the block diagrams and/or flowchart illustration, can be implemented by special purpose hardware-based systems which perform the specified functions or acts, or combinations of special purpose hardware and computer instructions.
In addition, each functional module or unit in each embodiment of the present invention may be integrated together to form an independent part, or each module may exist separately, or two or more modules may be integrated to form an independent part. The functions, if implemented in the form of software functional modules and sold or used as a stand-alone product, may be stored in a computer readable storage medium. Based on such understanding, the technical solution of the present invention or a part of the technical solution that contributes to the prior art in essence can be embodied in the form of a software product, which is stored in a storage medium and includes instructions for causing a computer device (which may be a smart phone, a personal computer, a server, or a network device, etc.) to execute all or part of the steps of the method according to the embodiments of the present invention. And the aforementioned storage medium includes: a U-disk, a removable hard disk, a Read-Only Memory (ROM), a Random Access Memory (RAM), a magnetic disk or an optical disk, and other various media capable of storing program codes.
The above description is only for the specific embodiments of the present invention, but the scope of the present invention is not limited thereto, and any person skilled in the art can easily conceive of the changes or substitutions within the technical scope of the present invention, and all the changes or substitutions should be covered within the scope of the present invention.
Claims (8)
1. A method for improving the conversation tone quality is characterized in that the corresponding relation between a channel and an audio configuration parameter is prestored in a mobile terminal where a calling user is located, and the method comprises the following steps:
before establishing voice communication connection, a first channel and a second channel are established in advance, and corresponding channels are configured according to the audio configuration parameters corresponding to the channels; the first channel and the second channel are links which are established between a mobile terminal where a calling user is located and a mobile terminal where a called user is located and are used for voice interaction;
after establishing voice communication connection, acquiring an audio signal sent by a mobile terminal where a called user is located and answering time of the called user for responding to voice call;
determining the type of the audio signal according to the listening time, wherein the type of the audio signal comprises a ring-back signal and a voice signal;
if the audio signal is a ring-back signal, calling a first channel to debug the ring-back signal;
if the audio signal is a voice signal, calling a second channel to debug the voice signal;
the audio configuration parameters include amplitude information and frequency information, and the invoking of the first channel or the second channel to debug the corresponding audio signal includes:
and debugging the amplitude of the audio signal entering the channel corresponding to the audio configuration parameter according to the amplitude information and/or debugging the frequency of the audio signal entering the channel corresponding to the audio configuration parameter according to the frequency information.
2. The method for improving the quality of voice in a call according to claim 1, wherein the listening time is obtained by:
receiving an answering signaling sent by the mobile terminal where the called user is located when the mobile terminal answers the voice call;
and analyzing the answering signaling according to the frame structure of the answering signaling to obtain the sending time of the answering signaling, and taking the sending time of the answering signaling as the answering time.
3. The method for improving the quality of voice in a call according to claim 2, wherein the listening time is obtained by:
and acquiring the signaling receiving time when the answering signaling is received, and taking the signaling receiving time as the answering time.
4. The method for improving the quality of speech sound of claim 1, wherein said determining the type of the audio signal according to the listening time comprises:
and taking the audio signal of a first time interval before the answering time as a ring-back signal and taking the audio signal of a second time interval after the answering time as a voice signal.
5. The method for improving call quality of claim 1, wherein the debugging the frequency of the audio signal entering the channel corresponding to the audio configuration parameter according to the frequency information comprises:
and debugging the frequency of the preset resonance point in the audio signal entering the channel corresponding to the audio configuration parameter according to the frequency information.
6. A device for improving conversation tone quality is characterized in that a mobile terminal where a calling party is located stores a corresponding relation between a channel and an audio configuration parameter in advance, and the device comprises:
the channel establishing module is used for establishing a first channel and a second channel in advance before voice communication connection is established, and configuring the corresponding channels according to the audio configuration parameters corresponding to the channels; the first channel and the second channel are links which are established between a mobile terminal where a calling user is located and a mobile terminal where a called user is located and are used for voice interaction;
the system comprises an acquisition module, a voice communication module and a voice communication module, wherein the acquisition module is used for acquiring an audio signal sent by a mobile terminal where a called user is located and answering time of the called user responding to a voice call after establishing voice communication connection;
the determining module is used for determining the type of the audio signal according to the answering time, wherein the type of the audio signal comprises a ring-back signal and a voice signal;
the first debugging module is used for calling a first channel to debug the ring-back signal when the audio signal is the ring-back signal;
the second debugging module is used for calling a second channel to debug the voice signal when the audio signal is the voice signal;
the audio configuration parameters include amplitude information and frequency information, and the debugging of the corresponding audio signal by the first debugging module calling the first channel or the second debugging module calling the second channel includes:
and debugging the amplitude of the audio signal entering the channel corresponding to the audio configuration parameter according to the amplitude information and/or debugging the frequency of the audio signal entering the channel corresponding to the audio configuration parameter according to the frequency information.
7. A mobile terminal, characterized in that the mobile terminal comprises a memory for storing a computer program and a processor for executing the computer program to make the mobile terminal execute the method for improving the quality of the call sound according to any one of claims 1 to 5.
8. A computer-readable storage medium storing the computer program for use in the mobile terminal of claim 7.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201910444890.8A CN110177176B (en) | 2019-05-27 | 2019-05-27 | Method and device for improving conversation tone quality and mobile terminal |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201910444890.8A CN110177176B (en) | 2019-05-27 | 2019-05-27 | Method and device for improving conversation tone quality and mobile terminal |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN110177176A CN110177176A (en) | 2019-08-27 |
| CN110177176B true CN110177176B (en) | 2021-06-08 |
Family
ID=67695840
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN201910444890.8A Active CN110177176B (en) | 2019-05-27 | 2019-05-27 | Method and device for improving conversation tone quality and mobile terminal |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN110177176B (en) |
Families Citing this family (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN102571758A (en) | 2011-12-16 | 2012-07-11 | 华为技术有限公司 | Method and device for realizing seamless transfer of two-party call transfer conference |
Citations (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101867657A (en) * | 2010-06-24 | 2010-10-20 | 中兴通讯股份有限公司 | Method and device for switching audio output equipment on mobile phone |
| CN102083231A (en) * | 2009-11-26 | 2011-06-01 | 中国移动通信集团广东有限公司 | Device and method for eliminating noise during polyphonic ringtone playing |
| CN105007364A (en) * | 2015-05-26 | 2015-10-28 | 腾讯科技（深圳）有限公司 | Voice conversation method and voice conversation apparatus |
| CN106534497A (en) * | 2016-10-31 | 2017-03-22 | 努比亚技术有限公司 | Method of adaptively switching sound effect output channels and terminal |
Family Cites Families (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| WO2006038402A1 (en) * | 2004-10-01 | 2006-04-13 | Matsushita Electric Industrial Co., Ltd. | Acoustic adjustment device and acoustic adjustment method |
| US8121267B2 (en) * | 2005-04-15 | 2012-02-21 | Eclips, Inc. | Forked-call ringback replacement system |
| CN101106603B (en) * | 2007-08-01 | 2013-04-17 | 上海华为技术有限公司 | A method, system and switch for bidirectional communication of CRBT |
| CN105813031A (en) * | 2014-12-30 | 2016-07-27 | 中兴通讯股份有限公司 | CRBT (color ring back tone) play method and device |
-
2019
- 2019-05-27 CN CN201910444890.8A patent/CN110177176B/en active Active
Patent Citations (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN102083231A (en) * | 2009-11-26 | 2011-06-01 | 中国移动通信集团广东有限公司 | Device and method for eliminating noise during polyphonic ringtone playing |
| CN101867657A (en) * | 2010-06-24 | 2010-10-20 | 中兴通讯股份有限公司 | Method and device for switching audio output equipment on mobile phone |
| CN105007364A (en) * | 2015-05-26 | 2015-10-28 | 腾讯科技（深圳）有限公司 | Voice conversation method and voice conversation apparatus |
| CN106534497A (en) * | 2016-10-31 | 2017-03-22 | 努比亚技术有限公司 | Method of adaptively switching sound effect output channels and terminal |
Also Published As
| Publication number | Publication date |
|---|---|
| CN110177176A (en) | 2019-08-27 |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| US8379073B2 (en) | Method for storing data of video telephony call in mobile terminal and system thereof | |
| WO2020151743A1 (en) | Random access method and terminal | |
| US20190380024A1 (en) | Method and system for selecting volte according to sim card by mobile terminal, and mobile terminal | |
| CN104980811A (en) | Remote controller device, call device, call system and call method | |
| CN109496451B (en) | Network parameter configuration method, device and computer readable storage medium | |
| US20230031044A1 (en) | Service processing method, user device, and computer-readable storage medium | |
| CN109582976A (en) | A kind of interpretation method and electronic equipment based on voice communication | |
| KR101034554B1 (en) | Method and apparatus for controlling voice quality of portable device | |
| WO2020103089A1 (en) | Dual-card communication method and related product | |
| US20220141742A1 (en) | Network switching method | |
| CN110177176B (en) | Method and device for improving conversation tone quality and mobile terminal | |
| US11943069B2 (en) | Data transmission method and device, base station, terminal and storage medium | |
| WO2014086292A1 (en) | Method and terminal for depending on called terminal to determine ring-back tone of calling terminal | |
| CN114466324B (en) | Calling method and related device | |
| CN113905128B (en) | Call reminding method and terminal | |
| CN110995939B (en) | Switching method and device of call network and storage medium | |
| US20220400050A1 (en) | Port configuration method and device | |
| US11895659B2 (en) | Control instruction transmission method, base station, terminal and storage medium | |
| US11799705B2 (en) | Message transmission method and device | |
| CN111800836A (en) | Communication method, communication device, electronic equipment and storage medium | |
| KR20070075581A (en) | Method and mobile communication terminal for making image call list by using captured image | |
| CN114697438B (en) | Method, device, equipment and storage medium for carrying out call by utilizing intelligent equipment | |
| CN117135266B (en) | Information processing method, device and computer readable storage medium | |
| CN113055916B (en) | Cell registration method, cell registration device, terminal, and storage medium | |
| WO2022198417A1 (en) | Information sending method and apparatus, and information receiving method and apparatus |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| PB01 | Publication | ||
| PB01 | Publication | ||
| SE01 | Entry into force of request for substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| GR01 | Patent grant | ||
| GR01 | Patent grant |