CN103236263B - Method, system and mobile terminal for improving call quality - Google Patents
Method, system and mobile terminal for improving call quality Download PDFInfo
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Abstract
The invention is suitable for the communication field, and provides a method, a system and a mobile terminal for improving the call quality, wherein the method comprises the following steps: receiving a sound signal in a call process, wherein the sound signal comprises a voice signal and a noise signal; starting a sound equalizer; carrying out noise reduction processing on the voice signal in the call process through a voice equalizer; and outputting the sound signal after the noise reduction processing. The invention reduces noise of the voice signal in the call process through the voice equalizer, so the algorithm for improving the call quality is simple and the production cost is low.
Description
Technical Field
The invention belongs to the field of communication, and particularly relates to a method, a system and a mobile terminal for improving call quality.
Background
The mobile terminal is often used by a user in various environments, and in a quiet environment, the mobile terminal can transmit a clear voice signal to a receiver, but if the mobile terminal is used in various noisy environments, such as on a bus, in an airport, in a subway, on the street, in a stadium, in a restaurant, etc., environmental noise is transmitted to the receiver along with the voice signal, and if the environmental noise is particularly strong, it is possible to mask the voice signal, so that the receiver cannot hear what the sender is saying.
Then, a voice noise reduction technique, also called a voice enhancement technique or a voice noise reduction technique, is introduced into the mobile terminal for removing various noise signals, thereby being capable of providing clear voice to a user of the mobile terminal. The voice noise reduction technology has been well studied in recent thirty years, but due to the complexity of the algorithm and the inexpertness of the application, the actual application is still not very wide, and the voice noise reduction does not achieve a good effect, so that the mobile terminal user cannot obtain the expected experience effect.
At present, there are a plurality of voice noise reduction technologies for performing noise reduction processing on a voice signal, and the implementation manner on a mobile terminal is as follows: a mobile terminal with a voice noise reduction function picks up voice with noise through a microphone, processes signals through a noise reduction technology, obtains cleaner voice signals, transmits the signals to a mobile terminal of a receiving party through a communication network, and then is heard by a mobile terminal user of the receiving party. That is, the noise reduction technique is generally performed at the voice sender, because it is easier to collect and analyze the ambient noise information at the sender, however, the application scheme of the voice noise reduction has some disadvantages in practical use, so that the user does not get a good high-quality voice call experience, and the disadvantages include:
first, the current speech noise reduction method is complex, and for the application of calls requiring real-time data processing, the complex algorithm is not practical, so that the final noise reduction effect is affected while the computational complexity is reduced. The existing noise reduction techniques put into practical use in mobile terminals are not the most advanced methods.
Second, many manufacturers have developed noise reduction chips by using hardware to implement the noise reduction function, but this increases the production cost, so that some mobile terminal manufacturers completely abandon the voice noise reduction function in the low-end mobile terminals, and only add the voice noise reduction function in the high-end mobile terminals.
Disclosure of Invention
The embodiment of the invention aims to provide a method, a system and a mobile terminal for improving the call quality, and aims to solve the problems that in the prior art, the effect of adopting a voice noise reduction method with low operation complexity is poor, and the noise reduction function is realized by using a noise reduction chip, so that the production cost is increased.
The embodiment of the invention is realized in such a way that a method for improving the call quality comprises the following steps:
receiving a sound signal in a call process, wherein the sound signal comprises a voice signal and a noise signal;
starting a sound equalizer;
carrying out noise reduction processing on the voice signal in the call process through a voice equalizer;
and outputting the sound signal after the noise reduction processing.
Another objective of an embodiment of the present invention is to provide a system for improving call quality, where the system includes:
the receiving module is used for receiving a sound signal in a call process, wherein the sound signal comprises a voice signal and a noise signal;
the starting module is used for starting the sound equalizer;
the noise reduction module is used for carrying out noise reduction processing on the sound signal in the conversation process through the sound equalizer;
and the output module is used for outputting the sound signal after the noise reduction processing.
It is a further object of the embodiments of the present invention to provide a mobile terminal including the system for improving the call quality.
In the embodiment of the invention, the noise reduction processing is carried out on the voice signal in the conversation process through the voice equalizer, so the algorithm for improving the conversation quality is simple and the production cost is low.
Drawings
Fig. 1 is a flowchart of a method for improving call quality according to an embodiment of the present invention.
Fig. 2 is a functional block diagram of a system for improving call quality according to a second embodiment of the present invention.
Fig. 3 is a flowchart of a method for improving call quality according to a third embodiment of the present invention.
FIG. 4-1 is a time domain diagram of a clean speech signal produced by a girl.
Fig. 4-2 is a time domain diagram of a white noise signal.
FIG. 5-1 is a frequency domain diagram of a clean speech signal from a girl.
Fig. 5-2 is a frequency domain plot of a white noise signal.
FIG. 6-1 is a time domain diagram of a clean speech signal uttered by a person.
Fig. 6-2 is a time domain plot of a noise signal from an automobile manufacturing plant.
FIG. 7-1 is a frequency domain diagram of a clean speech signal uttered by a person.
Fig. 7-2 is a frequency domain plot of a noise signal from an automobile manufacturing plant.
Fig. 8 is a diagram of an adjustment interface of a digital equalizer provided by the music playing software Winamp.
Fig. 9 is a generalized adjustment interface diagram of the sound equalizer for all users in the third embodiment of the present invention.
Fig. 10 is a diagram of an adjustment interface of a sound equalizer specific to a mobile terminal owner according to a third embodiment of the present invention.
Fig. 11 is a diagram of an adjustment interface of a voice equalizer displayed in a call interface according to a third embodiment of the present invention, which only includes an adjustment interface for adjusting a voice signal transmitted through a communication network.
Fig. 12 is a diagram of an adjustment interface of a voice equalizer displayed in a call interface according to a third embodiment of the present invention, including an adjustment interface for adjusting a voice signal transmitted through a communication network and an adjustment interface for adjusting a voice signal received through the communication network.
FIG. 13 is a schematic diagram of the operation of the sound equalizer for reducing noises in the automobile manufacturer according to the third embodiment of the present invention.
Fig. 14 is a schematic diagram of the operation of the sound equalizer for reducing the noise of the scraping disc in the third embodiment of the present invention.
FIG. 15 is a schematic diagram of the operation of the voice equalizer for suppressing white noise and enhancing male voice according to the third embodiment of the present invention.
Fig. 16 is a functional block diagram of a system for improving call quality according to a fourth embodiment of the present invention.
Detailed Description
In order to make the objects, technical solutions and advantages of the present invention more clearly apparent, the present invention is described in further detail below with reference to the accompanying drawings and embodiments. It should be understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention.
In order to explain the technical means of the present invention, the following description will be given by way of specific examples.
The first embodiment is as follows:
referring to fig. 1, a method for improving call quality according to an embodiment of the present invention includes the following steps:
s101, receiving a sound signal in a call process, wherein the sound signal comprises a voice signal and a noise signal;
s102, starting a sound equalizer;
in the first embodiment of the present invention, the sound equalizer is a tool that can adjust the amplitudes or gains of signals of different frequency components in a sound signal, and may be a single electronic device, or may be a software program, where the sound equalizer is a set of digital filter banks that can adjust the amplitudes or gains of signals of multiple frequency components (for example, three frequency bands of high frequency, intermediate frequency, and low frequency, or multiple frequency components set according to the frequency characteristics of a human voice signal) of a preset sound signal, and can increase or decrease the amplitudes or gains of certain frequency components, so that the sound signal with the amplitude or gain of each frequency component changed has clearer voice.
S103, noise reduction processing is carried out on the sound signals in the call process through a sound equalizer;
and S104, outputting the sound signal after the noise reduction processing.
In the first embodiment of the present invention, since the noise reduction processing is performed on the voice signal in the call process by the voice equalizer, the algorithm for improving the call quality is simple, and the production cost is low.
Example two:
referring to fig. 2, a system for improving communication quality according to a second embodiment of the present invention includes:
the receiving module 11 is configured to receive a sound signal in a call process, where the sound signal includes a voice signal and a noise signal;
a starting module 12 for starting the sound equalizer;
in the second embodiment of the present invention, the sound equalizer is a tool that can adjust the amplitudes or gains of signals of different frequency components in a sound signal, and may be a single electronic device, or may be a software program, where the sound equalizer is a set of digital filter banks, and can adjust the amplitudes or gains of signals of multiple frequency components (for example, three frequency bands of high frequency, intermediate frequency, and low frequency, or multiple frequency components set according to the frequency characteristics of a human voice signal) of a preset sound signal, and can increase or decrease the amplitudes or gains of certain frequency components, so that the sound signal with the amplitude or gain of each frequency component changed has clearer voice.
The noise reduction module 13 is configured to perform noise reduction processing on the voice signal in the call process through the voice equalizer;
and the output module 14 is used for outputting the sound signal after the noise reduction processing.
In the second embodiment of the present invention, since the noise reduction module performs noise reduction processing on the voice signal in the call process through the voice equalizer, the algorithm for improving the call quality is simple, and the production cost is low.
Example three:
referring to fig. 3, a method for improving call quality according to a third embodiment of the present invention includes the following steps:
s201, after a call is established, receiving a sound signal input by a sound input device in a communication process, and/or receiving a sound signal transmitted through a communication network in the communication process, wherein the sound signal comprises a voice signal and a noise signal;
in the third embodiment of the present invention, the sound input device may be a microphone or the like.
S202, starting a sound equalizer;
in the third embodiment of the present invention, the sound equalizer is a tool that can adjust the amplitudes or gains of signals of different frequency components in a sound signal, and may be a single electronic device, or may be a software program, where the sound equalizer is a set of digital filter banks that can adjust the amplitudes or gains of signals of multiple frequency components (for example, three frequency bands of high frequency, intermediate frequency, and low frequency, or multiple frequency components set according to the frequency characteristics of a human voice signal) of a preset sound signal, and can increase or decrease the amplitudes or gains of certain frequency components, so that the sound signal with the amplitude or gain of each frequency component changed has clearer voice.
S203, according to the preset parameters of the sound equalizer, the amplitude or the gain of the frequency corresponding to the noise signal in the sound signal in the call process is reduced through the sound equalizer, the preset parameters of the sound equalizer are parameters suitable for all users or parameters with personalized voice characteristics extracted according to the sound characteristics of a sender, and the parameters comprise the corresponding relation between each frequency and the amplitude or the gain corresponding to the frequency;
or,
displaying an adjusting interface of a sound equalizer in a call interface, receiving an adjusting instruction of the sound equalizer, and reducing the amplitude or gain of the frequency corresponding to a noise signal in a sound signal in the call process through the sound equalizer, wherein the adjusting interface is provided with an adjusting tool for adjusting the amplitude or gain corresponding to different frequencies;
in the third embodiment of the present invention, the adjustment interface of the sound equalizer displayed in the call interface may include an adjustment interface that adjusts a sound signal transmitted through the communication network and/or an adjustment interface that adjusts a sound signal received through the communication network. As shown in fig. 11, the adjustment interface of the sound equalizer displayed in the call interface only includes an adjustment interface for adjusting the sound signal transmitted through the communication network, and as shown in fig. 12, the adjustment interface of the sound equalizer displayed in the call interface includes an adjustment interface for adjusting the sound signal transmitted through the communication network and an adjustment interface for adjusting the sound signal received through the communication network, so that the user can touch and adjust the gain of each characteristic frequency with a finger to reduce or increase some frequency components to achieve the effect of suppressing part of the noise signal or improving the voice signal.
In the third embodiment of the present invention, while the amplitude or gain of the frequency corresponding to the noise signal in the voice signal during the call is reduced by the voice equalizer, the amplitude or gain of the frequency corresponding to the voice signal in the voice signal during the call can also be increased by the voice equalizer.
In the third embodiment of the present invention, the adjusting tool may be a window for inputting a numerical value or a button for adjusting a numerical value.
S204, when the sound signal input by the sound input device in the call process is received in the S201, transmitting the sound signal with the amplitude or the gain of the frequency corresponding to the noise signal reduced to a receiving party through a communication network; when the sound signal transmitted through the communication network during the call is received in S201, the sound signal with the amplitude or gain of the frequency corresponding to the noise signal reduced is output through the sound output device.
In the third embodiment of the present invention, the sound output device may be a speaker or the like.
The frequency characteristics of the sound signal are described in detail below.
The sound signal includes a voice signal and a noise signal, the voice signal and the noise signal have different frequency characteristics, and the voice signal between different persons also has different frequency characteristics, and the different noise signals also have respective frequency characteristics different from the voice signal.
In particular, from the analysis of the frequency characteristics of a human speech signal the following conclusions can be drawn: the fundamental tone frequency of human speaking is about 80 Hz-500 Hz, the male voice is lower and about 80 Hz-240 Hz, the child is about 120 Hz-400 Hz, and the female voice is 140 Hz-500 Hz; some changes occur when singing, the lowest male voice can reach 65Hz, the high male voice can reach 700Hz generally, and the highest female voice can reach 1100 Hz. So to speak, the frequency characteristics of the voice signals of children, women and men are significantly different for human voice. But also for different persons, e.g. in the female voice range, the frequency range of female voices with deep voices will be significantly lower than the frequency range of female voices with bright voices. This result makes it possible for each user to enhance the human voice signal according to the frequency characteristics of the voice signal during the call, so that the heard voice signal and the voice signal to be transmitted by the user can be clearer.
In addition, since the frequency characteristics of a human voice signal have differences from those of many noise signals, it can be used to reduce noise in a call environment. Two examples are given below. First, fig. 4-1 is a time domain diagram of a clean speech signal from a female student, fig. 4-2 is a time domain diagram of a white noise signal, which is the most common noise in a human living environment (because the mixture of multiple signals will eventually tend to be gaussian distributed according to the gaussian limit theorem), and the white noise is a cluttered signal, which will appear in the frequency domain as a frequency component with a relatively uniform distribution throughout the frequency domain axis. By transforming the clean speech signal into the frequency domain, it can be seen that the dominant frequency region of the signal (as shown in the box in fig. 5-1) is only approximately in the range of 100Hz to 1500Hz, while the frequency components above 1500Hz have small amplitude and become smaller and smaller, and the removal of this portion of the frequency components may have a limited effect on the overall speech signal. Then for both signals, if the frequency components above 1500Hz are removed, the contribution to the speech signal is small, but the large frequency components of the white noise signal are removed (as shown in the block in fig. 5-2), which allows a good removal of a portion of the noise signal by removing the portion that is not significant to the speech signal but is significant to the noise signal directly in the frequency domain. Also as shown in fig. 6-1, a time domain plot of a clean speech signal uttered by a person, and fig. 6-2, a time domain plot of a noise signal from an automobile manufacturing plant. Fig. 7-1 is a frequency domain plot of a clean speech signal uttered by a person, and fig. 7-2 is a frequency domain plot of a noise signal from an automobile manufacturing plant. As can be seen from fig. 7-1, the frequency characteristic dominant frequency region (as shown in the box) of the voice signal is approximately in the range of 100Hz to 1400Hz, and as can be seen from fig. 7-2, the frequency characteristic dominant frequency region (as shown in the box) of the noise signal of the automobile manufacturing plant is approximately in the range of 10Hz to 100Hz, so that the frequency characteristic dominant frequency regions of the two signals are nearly staggered without intersection, in which case, if the components in the low frequency range of 10Hz to 100Hz of the signals are removed or reduced, the noise signal of the automobile manufacturing plant is greatly reduced without having any influence on the voice signal.
From the above detailed description of the frequency characteristics of the sound signal, it can be seen that the influence of the noise signal can be reduced by using the difference between the frequency characteristics of the human voice signal and the frequency characteristics of the noise signal. Therefore, the embodiment of the invention utilizes the sound equalizer to reduce the influence of the noise signal on the sound signal.
Referring to fig. 8, the digital equalizer provided by the music playing software Winamp, as can be seen from fig. 8, the equalizer provides 10 characteristic frequencies from 70Hz to 16kHz for users to adjust. The adjustable frequency range and characteristic frequency provided by different equalizers are different, and they generally have their own strategies for enhancing sound effects, thereby creating different equalizers.
In the third embodiment of the present invention, the sound equalizer may adopt an equalizer of music playing software, such as an equalizer of Winamp, or may develop an equalizer specially designed for voice communication. The adjusting interface of the sound equalizer displayed in the communication interface can also adopt a graphical interface of the equalizer similar to music playing software, so that a user can adjust the sound signal according to the requirement of the user, and the communication quality is improved.
If the equalizer of the music playing software is adopted, the development cost can be reduced, and because the equalizer provides adjustment in different frequency bands of high frequency, intermediate frequency and low frequency, a certain noise reduction effect can be realized, but because the equalizer is provided aiming at the change of the music effect, the adjustment of the adjustable frequency range and the characteristic frequency does not consider the characteristics of voice signals and various noise signals, the effect is limited, and the equalizer specially developed for improving the call quality can be realized.
In the third embodiment of the present invention, the voice equalizer may be a generalized voice equalizer for all users, or may be a voice equalizer designed specifically for the owner of the mobile terminal. These two equalizers are described in detail below:
(1) generalized voice equalizer for all users
Such sound equalizers primarily take into account differences in the sound characteristics of men, women, children, as well as differences in human voice and ambient noise. Generally speaking, to hear the voice uttered by a person without distortion, only the sound with the frequency within 3kHz needs to be considered, and actually, the definition of the speaking voice can be ensured within the range of 2kHz, so the general equalizer for all users is designed within the range of 0-2 kHz, but the user of the mobile terminal often needs to hear some other sounds, such as music sound, environmental sound, etc., so the equalizer should also consider some higher frequency ranges, and here the following frequencies are selected as the characteristic frequencies within the high-frequency non-human voice range of the equalizer: 2kHz,3KHz,5kHz,7kHz,10 kHz. Since the equalizer is designed for a voice signal, only the frequency limit of 10kHz is considered at most. In the range of 0-2 KHz, the sound characteristic difference of different people needs to be considered, and the research fact can be considered here: the treble frequency range of the children sound is 260-880 Hz, the bass frequency range is 196-700 Hz, the treble frequency range of the female sound is 220-1100 Hz, the bass frequency range is 200-700 Hz, the treble frequency range of the male sound is 160-523 Hz, and the bass frequency range is 80-358 Hz. The center frequencies in these frequency ranges or the frequencies in the vicinity thereof can be regarded as characteristic frequencies of sound equalizer adjustment, and their center frequencies are 219Hz,341.5Hz,448Hz,450Hz,570Hz, and 660Hz, respectively. In the speech range, the lowest frequency can be set to 65Hz, which is the lowest sound of man, and 1500Hz can be set between 1100Hz and 2kHz at the highest frequency as a characteristic frequency, so that the sound equalizer includes 13 adjustable characteristic frequencies, 65Hz,220Hz,340Hz,450Hz,570Hz,660Hz,1100Hz,1500Hz,2kHz,3kHz,5kHz,7kHz,10 kHz. The adjustment interface of the sound equalizer involved according to the above analysis is shown in fig. 9.
(2) Sound equalizer specially designed for mobile terminal owner
The characteristic frequency of the equalizer considered in the high-frequency part above 2kHz is the same as that of the previous scheme, and only the frequency characteristic of the voice signal of the owner of the mobile terminal is considered in the range of 65Hz to 2kHz, because the frequency characteristic of the voice signal of each person is different, even if the frequency characteristic is the same, the frequency domain characteristics of low-lying sound and high-lying sound are different, the personalized consideration is added into the design of the voice equalizer, so that the voice equalizer can better enhance the sound of a specific person, and can provide better noise reduction effect. This consideration is based on the reality of such a mobile terminal usage: most users' mobile terminals are used by the mobile terminal owner for ninety percent of the time, and have little opportunity to lend themselves for use by others. Then the differences between the speaking characteristics of each mobile terminal owner can be considered, and a set of equalizer based on the frequency characteristics of the voice signals can be specially designed for the mobile terminal owner.
The selection of the frequency characteristics of the voice signal of the mobile terminal owner needs to be obtained by a learning method, and the specific mode is as follows: the mobile terminal or the computer is enabled to record some personal conventional words in a quiet environment, more sentences comprising various different voices can be recorded, then voice signals are analyzed through voice signal analysis software, the approximate low audio frequency range and high audio frequency range of the person are found out, and certainly, the recording process needs to meet some requirements of the voice signal analysis software. After finding the approximate bass and treble frequency ranges of the person, the frequency characteristics of the adjustable sound equalizer can be selected according to the frequency characteristics of the voice signal of the owner of the mobile terminal, and then a sound equalizer specific to the owner of the mobile terminal is established. For example, through the frequency characteristic learning process of the voice signal, the treble frequency range of a man is 420-460 Hz, and the bass frequency range is 150-200 Hz, then the center frequencies of 174Hz and 440Hz can be taken as two characteristic frequencies of the sound equalizer, and the two frequencies are set as adjustable characteristic frequencies considering that the frequency components beyond the lowest frequency of 150Hz and the highest frequency of 460Hz may be affected by the noise signal, and several other characteristic frequencies are set beyond the two frequency points, so as to better provide the noise reduction operation, for example, the characteristic frequencies are set at the places of 80Hz,800Hz and 1100 Hz. Taking the above factors into consideration, the sound equalizer designed specifically for the owner of the mobile terminal is shown in fig. 10, where thirteen adjustable characteristic frequencies are still used, but more or less characteristic frequencies may be used.
The voice equalizer capable of improving the call quality has been described above, and its use in a mobile terminal will be described below. The mobile terminal works in two directions during the conversation, and when a user has a conversation, the user expects the received sound to be clean and clear and also expects the emitted sound to be clean and clear. Generally, the noise reduction technology of the existing mobile terminal is beneficial, that is, before sound is sent, noise reduction processing is performed on a sending party, and then the sound is sent to a mobile terminal of an opposite party, and no mobile terminal in the market has the capability of processing a voice signal with noise sent by the opposite party, so that in this case, the sound equalizer in the embodiment of the invention can be used for performing noise reduction processing on the received sound signal. When the mobile terminal of the sender does not have the voice noise reduction function, the voice equalizer in the embodiment of the invention can also be adopted to carry out primary noise reduction processing on the voice signal input by the mobile terminal of the sender, and then the voice signal is sent out. Actually, the two sound equalizers designed above are used differently, a generalized sound equalizer for all mobile terminal users can process the transmitted and received voice signals, and the sound equalizer designed for the mobile terminal owner can only process the transmitted voice signals, because the adjustable frequency of the sound equalizer only contains the frequency characteristics of the voice signals of the mobile terminal owner, and the voices received by the mobile terminal come from different people, and the frequency characteristics of the voice signals cannot be recorded in the mobile terminals of other people, so that the sound equalizer cannot denoise according to the frequency characteristics of the voice signals of the other people. Therefore, to make the user of the mobile terminal capable of both hearing clean and clear speech and transmitting the speech to the ear of the other party, the two sound equalizers can be called simultaneously during the use of the mobile terminal to realize two noise reduction processes, and the adjustment interface of the sound equalizer displayed in the call interface is as shown in fig. 12. In the conversation process, a mobile terminal user can adjust the gain corresponding to each characteristic frequency in a finger touch mode according to needs, so that received and sent sound signals are clear.
The use of the voice equalizer may not be a simple matter for the user, so it is necessary for the manufacturer of the mobile terminal to embed some description documents or tutorials in the mobile terminal to guide the user to learn to use the function, for example, let the mobile terminal emit various voice signals with noise signals (which are all embedded files for helping the user to learn), and then guide the user to adjust various characteristic frequencies of the voice equalizer to experience the noise reduction effect, so that the user accumulates some usage experiences, mainly knows which frequency band the dominant frequency of each type of noise signal belongs to about the high, medium, and low frequency bands, and of course, the skilled use of the voice equalizer also requires the user to accumulate in many conversation processes. Generally, the effect of using a sound equalizer depends on the difference of dominant frequencies of various noises and human voices. The use of the sound equalizer is described below by way of a few examples:
(1) removing certain sounds with single frequency or narrow bandwidth but disturbing mind and affecting psychological feeling, such as noise of automobile manufacturers, as shown in fig. 7-2, such noise is generally in a low frequency range, and is not very harmful to speech signals, and the purpose of suppressing noise can be achieved by reducing frequency components in the low frequency range, which is approximately in a frequency range of 10-100 Hz, so that the amplitude in this frequency range can be greatly reduced, and a smaller amplitude reduction can be performed in a frequency range in the vicinity thereof, and a specific sound equalizer operation mode is shown in fig. 13.
(2) To remove some single-frequency or narrow-band, but more harsh noises, such as screaming sounds of some children or disc-scraping sounds, which will quickly make the ears feel tired, these sounds are all in the middle or high frequency range, and the amplitude of the components in this frequency range can be reduced, for example, to remove the disc-scraping sounds, the dominant spectrum is generally in the range of 2000Hz to 4000Hz, so that the amplitude of the components in this frequency range can be greatly reduced, and the amplitude of the frequency components in the vicinity thereof can be reduced by a small amount, so that the operation mode of the sound equalizer for this noise is as shown in fig. 14.
(3) For noise signals with a relatively wide bandwidth, it is also possible to remove a certain part of noise components that do not affect the spectrum of the speech signal, or increase the amplitude of the relevant frequency band of the speech signal, for example, for a speech signal affected by white noise, to reduce the effect, the amplitudes of all frequency components outside the frequency range of the speech of a person can be reduced, and the amplitudes of the frequency components of the speech part can be increased, for male voice, the specific operation mode is shown in fig. 15. Unlike the first two examples, where the voice equalizer at the receiving end of the mobile terminal is operated, the voice equalizer at the sending end is operated because the voice of the owner of the mobile terminal is transmitted relatively cleanly, and the influence of the noise environment around the sending end is reduced.
In the third embodiment of the invention, because the noise reduction processing is carried out on the voice signal in the conversation process by the voice equalizer, the algorithm for improving the conversation quality is simple, and the production cost is low; in addition, because the voice equalizer in the third embodiment of the present invention is a software program, and does not occupy a large amount of mobile terminal resources, the implementability of voice real-time processing is ensured, unlike the traditional noise reduction scheme that requires the support of additional hardware.
In addition, because the adjusting interface of the sound equalizer can be displayed in the call interface or the parameters of the sound equalizer preset by the user, the adjusting instruction of the user to the sound equalizer is received, and the amplitude or the gain of the frequency corresponding to the noise signal in the sound signal in the call process is reduced through the sound equalizer, the user can operate the sound equalizer by himself to achieve better call quality.
In addition, because the third embodiment of the invention can not only perform noise reduction processing on the voice signal input by the voice input device in the conversation process, but also perform noise reduction processing on the received voice signal transmitted by the communication network in the conversation process, both parties in the conversation can hear clear voice;
finally, because the sensitivity and preference of the human ear to various sounds is different, different people may have psychological and auditory preferences or dislikes for sounds introduced into their ears, e.g., some people may not like sharp, harsh sounds, while some people may not like low frequency, heavier sounds, and in addition, some people with certain ear disorders, or elderly people, they may be more sensitive to sounds, and certain sounds may have certain effects on their mind and body. Therefore, the embodiment of the invention can enable the user to control some special noises, reduce the influence of the special noises on the hearing and improve the user experience.
Example four:
referring to fig. 16, a system for improving communication quality according to a fourth embodiment of the present invention includes: a receiving module 21, a starting module 22, a noise reduction module 23 and an output module 24. Wherein,
the receiving module 21 is configured to receive a voice signal input through a voice input device during a call after the call is established, and/or receive a voice signal transmitted through a communication network during the call, where the voice signal includes a voice signal and a noise signal;
in the fourth embodiment of the present invention, the sound input device may be a microphone or the like.
A starting module 22 for starting the sound equalizer;
in the fourth embodiment of the present invention, the sound equalizer is a tool that can adjust the amplitudes or gains of signals of different frequency components in a sound signal, and may be a single electronic device, or may be a software program, where the sound equalizer is a set of digital filter banks, and can adjust the amplitudes or gains of signals of a plurality of preset frequency components (for example, three frequency bands of high frequency, intermediate frequency, and low frequency, or a plurality of frequency components set according to the frequency characteristics of a human voice signal) of the sound signal, and can increase or decrease the amplitudes or gains of some frequency components, so that the sound signal with the changed amplitudes or gains of the frequency components has clearer voice.
The noise reduction module 23 is configured to reduce, by a sound equalizer, an amplitude or a gain of a frequency corresponding to a noise signal in a sound signal during a call according to a preset parameter of the sound equalizer, where the preset parameter of the sound equalizer is a parameter applicable to all users or a parameter having an individualized speech feature extracted according to a sound feature of a sender, and the parameter includes a correspondence between each frequency and the amplitude or the gain corresponding to the frequency;
or,
the noise reduction module 23 specifically includes:
the display module 231 is configured to display an adjustment interface of the sound equalizer in the call interface, where the adjustment interface has an adjustment tool for adjusting amplitudes or gains corresponding to different frequencies;
an instruction receiving module 232, configured to receive an adjustment instruction for the sound equalizer;
the reducing module 233 is configured to reduce, by the voice equalizer, an amplitude or a gain of a frequency corresponding to a noise signal in the voice signal during the call.
In the fourth embodiment of the present invention, the adjustment interface of the sound equalizer displayed in the call interface may include an adjustment interface for adjusting a sound signal transmitted through the communication network and/or an adjustment interface for adjusting a sound signal received through the communication network. As shown in fig. 11, the adjustment interface of the sound equalizer displayed in the call interface only includes an adjustment interface for adjusting the sound signal transmitted through the communication network, and as shown in fig. 12, the adjustment interface of the sound equalizer displayed in the call interface includes an adjustment interface for adjusting the sound signal transmitted through the communication network and an adjustment interface for adjusting the sound signal received through the communication network, so that the user can touch and adjust the gain of each characteristic frequency with a finger to reduce or increase some frequency components to achieve the effect of suppressing part of the noise signal or improving the voice signal.
In the fourth embodiment of the present invention, the system may further include a raising module, configured to raise, by a sound equalizer, an amplitude or a gain of a frequency corresponding to a voice signal in the sound signal during the call.
In the fourth embodiment of the present invention, the adjusting tool may be a window for inputting a numerical value or a button for adjusting a numerical value.
An output module 24, configured to transmit, to a receiving party through a communication network, the sound signal with the amplitude or gain of the frequency corresponding to the noise signal reduced when the receiving module 21 receives the sound signal input through the sound input device during the call; when the receiving module 21 receives the voice signal transmitted through the communication network during the call, the voice signal with the amplitude or gain of the frequency corresponding to the noise signal reduced is output through the voice output device.
In the fourth embodiment of the invention, because the noise reduction processing is carried out on the voice signal in the conversation process by the voice equalizer, the algorithm for improving the conversation quality is simple, and the production cost is low; in addition, because the voice equalizer in the fourth embodiment of the present invention is a software program, and does not occupy a large amount of mobile terminal resources, the implementability of voice real-time processing is ensured, unlike the traditional noise reduction scheme that requires the support of additional hardware.
In addition, because the adjusting interface of the sound equalizer can be displayed in the call interface or the parameters of the sound equalizer preset by the user, the adjusting instruction of the user to the sound equalizer is received, and the amplitude or the gain of the frequency corresponding to the noise signal in the sound signal in the call process is reduced through the sound equalizer, the user can operate the sound equalizer by himself to achieve better call quality.
In addition, since the fourth embodiment of the present invention can perform noise reduction processing on the voice signal input through the voice input device during the call, and can also perform noise reduction processing on the received voice signal transmitted through the communication network during the call, both parties of the call can hear clear voice.
It will be understood by those skilled in the art that all or part of the steps in the method for implementing the above embodiments may be implemented by relevant hardware instructed by a program, and the program may be stored in a computer-readable storage medium, such as ROM/RAM, magnetic disk, optical disk, etc.
The above description is only for the purpose of illustrating the preferred embodiments of the present invention and is not to be construed as limiting the invention, and any modifications, equivalents and improvements made within the spirit and principle of the present invention are intended to be included within the scope of the present invention.
Claims (8)
1. A method for improving call quality, the method comprising:
receiving a sound signal in a call process, wherein the sound signal comprises a voice signal and a noise signal;
starting a sound equalizer;
the method comprises the steps that a voice signal in a call process is subjected to noise reduction processing through a voice equalizer, wherein the noise reduction processing specifically comprises the step of reducing the amplitude or the gain of the frequency corresponding to the noise signal in the voice signal in the call process through the voice equalizer according to the preset parameters of the voice equalizer, the preset parameters of the voice equalizer are parameters suitable for all users or parameters with personalized voice characteristics extracted according to the voice characteristics of a sender, and the parameters comprise the corresponding relation between each frequency and the amplitude or the gain corresponding to the frequency;
the amplitude or the gain of the frequency corresponding to the noise signal in the sound signal in the call process is reduced through the sound equalizer, and meanwhile, the amplitude or the gain of the frequency corresponding to the voice signal in the sound signal in the call process is improved through the sound equalizer; and outputting the sound signal after the noise reduction processing.
2. The method according to claim 1, wherein the receiving the voice signal during the call is specifically:
after the call is established, the voice signal input by the voice input device in the call process is received, and/or the voice signal transmitted by the communication network in the call process is received.
3. The method of claim 1, wherein the performing noise reduction processing on the voice signal in the call process by the voice equalizer specifically further comprises:
displaying an adjusting interface of a sound equalizer in a call interface, receiving an adjusting instruction of the sound equalizer, and reducing the amplitude or gain of the frequency corresponding to the noise signal in the sound signal in the call process through the sound equalizer, wherein the adjusting interface is provided with an adjusting tool for adjusting the amplitude or gain corresponding to different frequencies.
4. The method according to claim 2, wherein the outputting the noise-reduced sound signal is specifically:
when the voice signal input by the voice input device in the call process is received, the voice signal with the amplitude or the gain of the frequency corresponding to the noise signal reduced is transmitted to a receiving party through a communication network;
when the voice signal transmitted by the communication network in the call process is received, the voice signal with the amplitude or the gain of the frequency corresponding to the noise signal reduced is output by the voice output device.
5. A system for improving call quality, the system comprising:
the receiving module is used for receiving a sound signal in a call process, wherein the sound signal comprises a voice signal and a noise signal;
the starting module is used for starting the sound equalizer;
the system comprises a noise reduction module, a voice equalizer and a voice processing module, wherein the noise reduction module is used for carrying out noise reduction processing on a voice signal in a call process through the voice equalizer, the noise reduction module is specifically used for reducing the amplitude or the gain of the frequency corresponding to the noise signal in the voice signal in the call process through the voice equalizer according to the preset parameters of the voice equalizer, the preset parameters of the voice equalizer are parameters suitable for all users or parameters with personalized voice characteristics extracted according to the voice characteristics of a sender, and the parameters comprise the corresponding relation between each frequency and the amplitude or the gain corresponding to the frequency;
the improving module is used for improving the amplitude or the gain of the frequency corresponding to the voice signal in the call process through the voice equalizer;
and the output module is used for outputting the sound signal after the noise reduction processing.
6. The system according to claim 5, wherein the receiving module is specifically configured to receive, after the call is established, the voice signal input through the voice input device during the call, and/or receive the voice signal transmitted through the communication network during the call.
7. The system of claim 5,
the noise reduction module specifically comprises:
the display module is used for displaying an adjusting interface of the sound equalizer in the call interface, and the adjusting interface is provided with an adjusting tool for adjusting the amplitude or the gain corresponding to different frequencies;
the instruction receiving module is used for receiving an adjusting instruction of the sound equalizer;
and the reducing module is used for reducing the amplitude or the gain of the frequency corresponding to the noise signal in the sound signal in the call process through the sound equalizer.
8. A mobile terminal comprising the system for improving call quality of any of claims 5-7.
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