CN105744084A - Mobile terminal and method for improving conversation tone quality thereof - Google Patents

Mobile terminal and method for improving conversation tone quality thereof Download PDF

Info

Publication number
CN105744084A
CN105744084A CN201610281071.2A CN201610281071A CN105744084A CN 105744084 A CN105744084 A CN 105744084A CN 201610281071 A CN201610281071 A CN 201610281071A CN 105744084 A CN105744084 A CN 105744084A
Authority
CN
China
Prior art keywords
voice
voice signal
parameter
signal
frequency response
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.)
Granted
Application number
CN201610281071.2A
Other languages
Chinese (zh)
Other versions
CN105744084B (en
Inventor
王海盈
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Hisense Mobile Communications Technology Co Ltd
Original Assignee
Hisense Mobile Communications Technology Co Ltd
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Application filed by Hisense Mobile Communications Technology Co Ltd filed Critical Hisense Mobile Communications Technology Co Ltd
Priority to CN201610281071.2A priority Critical patent/CN105744084B/en
Publication of CN105744084A publication Critical patent/CN105744084A/en
Application granted granted Critical
Publication of CN105744084B publication Critical patent/CN105744084B/en
Active legal-status Critical Current
Anticipated expiration legal-status Critical

Links

Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/60Substation equipment, e.g. for use by subscribers including speech amplifiers
    • H04M1/6016Substation equipment, e.g. for use by subscribers including speech amplifiers in the receiver circuit
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/72Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/724User interfaces specially adapted for cordless or mobile telephones
    • H04M1/72448User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions

Abstract

The invention provides a mobile terminal and a method for improving conversation tone quality thereof. The method comprises the steps of after a conversation is started, reading a decoded voice signal in a receiving direction; detecting the signal intensity of the read voice signal and determining corresponding voice adjustment parameters; adjusting the signal intensity of the voice signal and then transmitting the voice signal to a loudspeaker based on the voice adjustment parameters, wherein voice adjustment parameters comprise at least one of the follows: a volume adjustment parameter and a filtering adjustment parameter. By application of the mobile terminal and the method for improving the conversation tone quality thereof, the voice signal in the receiving direction of the mobile terminal can be detected actively, a corresponding mechanism can be started for signal adjustment to improve the conversation tone quality, and the user experience is accordingly improved.

Description

Mobile terminal and the method promoting mobile terminal call tonequality
Technical field
The present invention relates to communication technical field, specifically, the present invention relates to a kind of mobile terminal and the method and apparatus promoting mobile terminal call tonequality.
Background technology
In the current communications industry, speech business is business currently mainly.When carrying out voice call by mobile terminal, often there is problems in that the sound articulation of partner is inadequate;Communication process is with noise.The voice quality improving constantly terminal use's call is operator and the target of manufacturer's pursuit.
In order to improve call tone quality, in existing mobile communication system, as shown in Figure 1, on call sending direction, can arrange on voice pickup circuit (usually microphone circuit) and simply filter adjustment, Gain tuning scheduling algorithm or circuit, the voice exported by user is adjusted, coordinates the performance of mike that terminal uses with this.It is further possible to cooperation noise reduction algorithm, suppress noise and suppress echo;Afterwards, encoding and decoding speech complete coding, and exported by radio-frequency (RF) receiving and transmission module.
It was found by the inventors of the present invention that it is meet general requirement that the standard of the communications industry often assumes that equipment is sent to the sound quality of partner.Therefore, existing conversing recipient upwards, it is not usually required to and did multiprocessing, it is only necessary to increase some simply filtering or amplifiers, complete the Gain tuning that decoded voice is fixed or fixing filtering adjustment to by encoding and decoding speech.
But, even if actually the properties of terminal is qualified, often because some reasons (such as, the sound of speaking of partner is less) sound quality of causing user to receive is poor, namely sound quality problem occurs;And by the existing Gain tuning being fixed in a receive direction, or fixing filtering adjustment, user can not be met and improve the actual demand of call tone quality.
Summary of the invention
For the defect that above-mentioned prior art exists, the invention provides a kind of mobile terminal and the method and apparatus promoting mobile terminal call tonequality, being used for improving call tone quality, thus improving Consumer's Experience.
The invention provides a kind of method promoting mobile terminal call tonequality, including:
After call starts, read the voice signal that recipient upwards decodes;
The voice signal read is carried out the detection of signal intensity, it is determined that go out corresponding voice and adjust parameter;
Adjust parameter according to described voice, after adjusting the signal intensity of described voice signal, be transferred to speaker;
Wherein, described voice adjust parameter include following at least one: volume adjusting parameters, filtering adjust parameter.
The embodiment of the present invention additionally provides a kind of device promoting call tone quality, including:
Voice signal read module, after starting for call, reads the voice signal that recipient upwards decodes;
Adjust parameter determination module, for the voice signal read being carried out the detection of signal intensity, it is determined that go out corresponding voice and adjust parameter;Wherein, described voice adjust parameter include following at least one: volume adjusting parameters, filtering adjust parameter;
Call tone quality adjusting module, adjusts parameter for the voice determined according to described adjustment parameter determination module, is transferred to speaker after adjusting the signal intensity of described voice signal.
In technical scheme, Regulation mechanism of taking the initiative, before transmitting voice signal recipient upwards decoded is to speaker, actively reads the voice signal that recipient upwards decodes;And based on the signal intensity of the voice signal read, it is determined that go out corresponding voice and adjust parameter.Compare existing by fixing gain or filtering adjust voice signal, the solution of the present invention according to the practical situation of currently received voice signal, can carry out accommodation, meets user and improves the actual demand of call tone quality, improves Consumer's Experience.
Aspect and advantage that the present invention adds will part provide in the following description, and these will become apparent from the description below, or is recognized by the practice of the present invention.
Accompanying drawing explanation
The present invention above-mentioned and/or that add aspect and advantage will be apparent from easy to understand from the following description of the accompanying drawings of embodiments, wherein:
Fig. 1 is the structural representation of existing communication system;
Fig. 2 is the structural representation of the communication system of the embodiment of the present invention;
Fig. 3 is the schematic flow sheet of the method promoting call tone quality of the embodiment of the present invention;
Fig. 4 is the comparison schematic diagram between the actual frequency response curve of the embodiment of the present invention and target frequency response curve;
Fig. 5 a, 5b are the structural representation of the mobile terminal of the embodiment of the present invention;
Fig. 6 is the structural representation adjusting parameter determination module of the embodiment of the present invention.
Detailed description of the invention
Being described below in detail embodiments of the invention, the example of described embodiment is shown in the drawings, and wherein same or similar label represents same or similar element or has the element of same or like function from start to finish.The embodiment described below with reference to accompanying drawing is illustrative of, and is only used for explaining the present invention, and is not construed as limiting the claims.
Those skilled in the art of the present technique are appreciated that unless expressly stated, and singulative used herein " ", " one ", " described " and " being somebody's turn to do " may also comprise plural form.Should be further understood that, the wording " including " used in the description of the present invention refers to there is described feature, integer, step, operation, element and/or assembly, but it is not excluded that existence or adds other features one or more, integer, step, operation, element, assembly and/or their group.It should be understood that when we claim element to be " connected " or during " coupled " to another element, it can be directly connected or coupled to other elements, or can also there is intermediary element.Additionally, " connection " used herein or " coupling " can include wireless connections or wireless couple.Wording "and/or" used herein includes one or more list the whole of item or any cell being associated and combines with whole.
Those skilled in the art of the present technique are appreciated that unless otherwise defined, and all terms used herein (include technical term and scientific terminology), have with the those of ordinary skill in art of the present invention be commonly understood by identical meaning.It should also be understood that, those terms of definition in such as general dictionary, should be understood that there is the meaning consistent with the meaning in the context of prior art, and unless by specific definitions as here, otherwise will not explain by idealization or excessively formal implication.
Those skilled in the art of the present technique are appreciated that, " terminal " used herein above, " terminal unit " had both included the equipment of wireless signal receiver, it only possesses the equipment of wireless signal receiver of non-emissive ability, include again the equipment receiving and launching hardware, it has the reception that on bidirectional communication link, can carry out two-way communication and launches the equipment of hardware.This equipment may include that honeycomb or other communication equipments, and it has single line display or multi-line display or does not have honeycomb or other communication equipments of multi-line display;PCS (PersonalCommunicationsService, PCS Personal Communications System), its can combine voice, data process, fax and/or its communication ability;PDA (PersonalDigitalAssistant, personal digital assistant), it can include radio frequency receiver, pager, the Internet/intranet access, web browser, notepad, calendar and/or GPS (GlobalPositioningSystem, global positioning system) receptor;Conventional laptop and/or palmtop computer or other equipment, it has and/or includes the conventional laptop of radio frequency receiver and/or palmtop computer or other equipment." terminal " used herein above, " terminal unit " can be portable, can transport, be arranged in the vehicles (aviation, sea-freight and/or land), or it is suitable for and/or is configured at local runtime, and/or with distribution form, any other position operating in the earth and/or space is run." terminal " used herein above, " terminal unit " can also is that communication terminal, access terminals, music/video playback terminal, can be such as PDA, MID (MobileInternetDevice, mobile internet device) and/or there is the mobile phone of music/video playing function, it is also possible to it is the equipment such as intelligent television, Set Top Box.
It was found by the inventors of the present invention that in actual user's communication process, even if mobile phone properties is qualified, as similar following reason causes that the sound quality being sent to the other side is poor.Such as, user holds the fault of mobile phone, causes that mobile microphone distance mouth is distant, thus causing that the sound that mobile phone sends is little;Even if it is correct that user holds the posture of mobile phone, but it is because long-term local environment or voice is exactly less than normal when individual physiological factor causes making a phone call;User have purchased Leather cover for handset to prevent mobile phone from breaking, and leather sheath plugs mobile phone Mike hole, causes mobile microphone cavity to change;User is in the Feeble field of base station and makes a phone call to cause choppy voice.And adjusted by existing tonequality method for improving, the simple filtering increased in a receive direction or amplifier, the Gain tuning being fixed or filtering, user can not be met and improve the actual demand of call tone quality.
Therefore, the present inventor considers, can take the initiative Regulation mechanism, before transmitting voice signal recipient upwards decoded is to speaker, the practical situation of the voice signal that active detecting decodes, and start corresponding Regulation mechanism according to practical situation, voice signal is repaired and corrects, improve call tone quality with this, promote Consumer's Experience.
Technical scheme is described in detail below in conjunction with accompanying drawing.
Embodiments provide a kind of communication system, as in figure 2 it is shown, may include that the first filtering adjusting module, the first gain regulation module, noise reduction module, encoding and decoding speech module, radio-frequency (RF) receiving and transmission module, tonequality hoisting module, the second filtering adjusting module, the second gain regulation module.
Wherein, the first filtering adjusting module, the first gain regulation module, noise reduction module, encoding and decoding speech module, radio-frequency (RF) receiving and transmission module, all can adopt the conventional functional module in existing communication system.Such as, the performance of the mike that the first filtering adjusting module, the first gain regulation module use for coordinating terminal, user the voice exported is adjusted;Noise reduction module is used for suppressing noise and suppressing echo;Encoding and decoding speech module, for being encoded the signal received or decode;Radio-frequency (RF) receiving and transmission module is used for receiving or sending voice signal.
And after tonequality hoisting module provided by the invention starts for call, read the voice signal that recipient upwards decodes;The voice signal read is carried out the detection of signal intensity, it is determined that go out corresponding voice and adjust parameter.Wherein, voice adjust parameter include following at least one: volume adjusting parameters, filtering adjust parameter.Afterwards, tonequality hoisting module, according to the volume adjusting parameters determined, generates corresponding volume and adjusts instruction, and be sent to the first gain regulation module;Adjust parameter according to the filtering determined, generate corresponding wave filter and adjust instruction, and be sent to the second filtering adjusting module.
Correspondingly, the voice signal of encoding and decoding speech module output, for adjusting instruction according to the wave filter received, is filtered adjusting, obtains the frequency characteristic close with default target frequency response curve with this by the second filtering adjusting module;The voice signal of encoding and decoding speech module output, for adjusting instruction according to the volume received, is carried out Gain tuning, obtains customer satisfaction system volume with this by the second gain regulation module.
As in figure 2 it is shown, tonequality hoisting module provided by the invention can have two inputs, one of them is the signal that sends of partner voice signal after the decoding of encoding and decoding speech module, and another is the signal quality parameter that radio-frequency (RF) receiving and transmission module sends.
Tonequality hoisting module has three outputs, two of which output filters adjusting module and the second gain regulation module respectively to second, being used for treating the voice signal being transferred to user to be adjusted, the 3rd output, to encoding and decoding speech module, is used for doing related voice prompting.
It is considered that the operation of tonequality hoisting module needs the regular hour, therefore, the tonequality hoisting module that the embodiment of the present invention provides, it is possible to bypass is outside normal voice path, to avoid bringing extra time delay to normal voice call.Or, under the whole communication system less demanding situation to time delay, it is possible to increase a time delay device between adjusting module, the second gain regulation module in encoding and decoding speech module and recipient's the second filtering upwards.The time span of time delay device is suitable with the computing required time of tonequality hoisting module.
About the functional realiey of tonequality hoisting module provided by the invention, it is referred in the method promoting call tone quality that the embodiment of the present invention provides the realization of each step.
The method promoting call tone quality that the embodiment of the present invention provides, as it is shown on figure 3, its idiographic flow may include steps of:
S301: after call starts, reads the voice signal that recipient upwards decodes.
In practical application, the scheme of lifting call tone quality provided by the invention, it is established as triggering signal with call, using end of conversation as termination signal.
When answering or call, after call starts, the signal that radio-frequency (RF) receiving and transmission module is exported is decoded by encoding and decoding speech module, decodes recipient's voice signal upwards.The solution of the present invention will be taken the initiative Regulation mechanism, before the transmitting voice signal that will decode is to speaker, intercept the voice signal decoded, in order to be transmitted further to user after the voice signal intercepted being adjusted by follow-up step.
In the embodiment of the present invention, it is possible to by the signal of output after encoding and decoding speech module decodes, be called recipient's voice signal upwards;And by the signal before encoding and decoding speech module coding, it is called the voice signal on sending direction.
The present inventor considers, determination due to the detection of signal intensity, voice adjustment parameter, and the adjustment of voice signal is required for the regular hour, and the time of some call is shorter, therefore, after being likely to appear in end of conversation, the situation that the adjustment of above-mentioned voice signal does not have started, also just the tonequality of this call cannot be promoted.
Further, the present inventor is it is also contemplated that owing to the detection of signal intensity, voice adjust the determination of parameter, and the adjustment of voice signal is required for the regular hour, and the tonequality being likely to can not get tonequality hoisting module when call just begins setting up promotes.
Therefore, in order to when call begins setting up most, a reasonable sound can be can be obtained by according to the mobile phone feature of partner or people's feature of talking, adjust again without after waiting the detection of tonequality hoisting module, the embodiment of the present invention can pre-build voice adjustment data base.Voice adjusts in data base, it is possible to telephone number for index, and the voice that the different telephone number of record is corresponding adjusts parameter;The voice corresponding with telephone number adjusts parameter, for the voice signal from this telephone number is initially adjusted.
Wherein, corresponding with telephone number voice adjusts parameter, it is possible to talk feature according to the feature of mobile phone in communication process before or people and the voice determined adjusts parameter;Or the voice that can also be based on the acquiescence that big data obtain initially adjusts parameter.
Specifically, after call starts, it is possible to adjust from voice and data base searches the voice adjustment parameter corresponding with the telephone number of partner;If finding out, then can adjust parameter according to the voice found out, the voice signal of encoding and decoding speech module output is initially adjusted;If not finding out, then the voice signal read can be carried out the detection of signal intensity, adjusting at voice and data base determining, corresponding voice adjusts parameter;Adjust parameter according to the voice determined, after adjusting the signal intensity of voice signal, be transferred to speaker.
Such as, the speech feature of partner is exactly that sound is little, and the acoustical signal that partner is sent to user is always less, then, it is possible to adjust one corresponding postiive gain of storage in data base at this voice.After setting up call, intensity according to sound, adjusts data base by retrieving voice, reads this postiive gain, and the Gain tuning instruction corresponding to the postiive gain of reading is passed in the second gain regulation module so that namely initial in call arrange a higher amplification.
In practical application, it is possible to voice is adjusted data base and screens and set.Such as, in order to avoid voice adjustment data base is excessively huge, it is possible to just for the telephone number of contact person in address list, the voice adjustment parameter that storage is corresponding.Or, it is also possible to preset a upper limit threshold, from address list, choose the telephone number of the quantity less than or equal to this upper limit threshold, the voice adjustment parameter that storage is corresponding.Or, it is also possible to from address list, select frequent contact, for the telephone number of the frequent contact selected, the voice adjustment parameter that storage is corresponding.
Popular says, after telephone number being detected, after adjusting parameter according to the voice that telephone number is corresponding, above-mentioned voice can be directly invoked and adjust parameter, carrying out the adjustment of the voice signal inputted, if being not detected by voice corresponding to telephone number to adjust parameter, being then made directly following step.
S302: the voice signal read is carried out the detection of signal intensity, it is determined that go out corresponding voice and adjust parameter.
It is considered that the object that promotes of call tone quality is mainly the content of speaking in voice signal, and voice signal often exists some noises or environmental disturbances sound etc..
Therefore, in the embodiment of the present invention, after reading recipient's voice signal upwards, it is possible to first carry out certain pretreatment, then carry out the detection of signal intensity again to determine that voice adjusts parameter.
Wherein, pretreatment can specifically include following at least one: effectively sound extracts, non-voice filters, audio segmentation, FFT (FastFourierTransformation, fast fourier transform) process.
Wherein, effective sound extracts and is mainly used in removing the interval in the middle of partner language;Non-voice filters for removing noise or environmental disturbances sound;Audio segmentation is for according to the segmentation unit preset, becoming junior unit by continuous print acoustic processing, namely obtain the voice signal of each time period;FFT processes for each junior unit is become frequency-region signal by time-domain signal.
Extract about effective sound, non-voice filters, the concrete operations of audio segmentation, FFT process, it is possible to use the technological means that those skilled in the art commonly use, and no longer describes in detail herein.
In the embodiment of the present invention, after the voice signal read is carried out pretreatment, it is possible to the voice signal obtained after pretreatment is carried out the detection of signal intensity the signal intensity according to detection, it is determined that go out corresponding voice and adjust parameter.
Wherein, voice adjust parameter include following at least one: volume adjusting parameters, filtering adjust parameter.
In the application, what voice signal carried out signal intensity detects the process to determine corresponding volume adjusting parameters, can be referred to as volume detection;And the detection that voice signal carries out signal intensity filters, to determine, the process adjusting parameter accordingly, frequency response detection can be referred to as.
About how carrying out volume detection, frequency response detection, will be explained below introducing.
1) volume detection
In the embodiment of the present invention, after the voice signal read is carried out pretreatment, it is possible to the voice signal obtained after pretreatment is carried out volume detection, it is determined that go out corresponding volume adjusting parameters.
Specifically, it is possible to for default each time period, based on voice signal signal intensity on each time point in this time period, it is determined that the signal energy equivalence value of voice signal in this time period;If it is determined that signal energy equivalence value less than default volume threshold, then the volume difference between signal energy equivalence value and volume threshold is defined as the volume adjusting parameters that in this time period, voice signal is corresponding.
For example, it is possible to be integrated adding up by signal intensity corresponding for each time point in this time period, draw signal energy equivalence value corresponding to this time period.Or, it is also possible to calculate the meansigma methods of signal intensity corresponding to each time point in this time period, using the meansigma methods that calculates as signal energy equivalence value corresponding to this time period.
2) frequency response detection
The present inventor is it is considered that the frequency-response characteristic of sound has important effect for voice and semantic understanding.Wherein, the sound of low frequency is very crucial for tone color, reduction degree and the identification that can promote well people's sound of low frequency reduction;And the sound of high frequency has bigger help for the reproducibility of information, the sound of high frequency disappearance sounds that lacking details understands difficulty.
Therefore, in the embodiment of the present invention, it is possible to the frequency response of the signal that partner is sent is modified.More preferably, it is possible to after the voice signal read is carried out pretreatment, the voice signal obtained after pretreatment is carried out frequency response detection, it is determined that go out corresponding filtering and adjust parameter.
In order to avoid frequently adjusting wave filter, in the embodiment of the present invention, for each frequency range, set frequency range thresholding, i.e. first threshold;Frequency response difference is more than just carrying out frequency response correction after first threshold.
Further, it is contemplated that, if the first threshold that only one of which frequency range exceedes, other frequency ranges all not less than, then adjust frequency response meaning be just less than strongly.Therefore, overall frequency response adjustment can also set frequency response thresholding, i.e. Second Threshold.So, before adjustment, it may be determined that frequency response difference, more than the quantity of the frequency range of first threshold, just adjusts frequency response when the quantity determined is more than Second Threshold.
Specifically, it is possible to for default each time period, based on voice signal signal intensity on each time point in this time period, it is determined that go out the actual frequency response curve of voice signal in this time period;The actual frequency response curve calculating voice signal in this time period and the target frequency response curve preset frequency response difference in each frequency range.
Then, it is determined that frequency response difference is more than the quantity of the frequency range of first threshold;If it is determined that quantity more than Second Threshold, then can will be greater than the frequency response difference of first threshold and corresponding band value in described actual frequency response curve and add up, the frequency response curve after being adjusted.By filter parameter corresponding for the frequency response curve after adjusting, it is determined that the filtering corresponding for voice signal in this time period adjusts parameter.
In practical application, the word content expressed when speaking is different, and frequency response is also not quite similar.But the frequency response got up of long-term accumulated is but have higher similarity.The too short time may result in wave filter frequently adjust bring speech quality change excessively frequent;The long time is likely to also have little time to adjust call and is over.Therefore, it can the experience according to those skilled in the art, choose suitable time span as the unit of time gathering actual frequency response, the time span of the time period namely preset.
Such as, as shown in Figure 4, we can be divided into some parts a frequency response as required.Such as, it is possible to the frequency range of 100HZ to 8000Hz is divided into 11 points, the mid frequency of every part respectively f1=200f2=400f3=600f4=800f5=1000f6=2000f7=3000f8=4000f9=5 000f10=6000f11=7000.Frequency response one is target frequency response curve set in advance, and frequency response two is the actual frequency response curve actually receiving voice signal.From Fig. 4 this it appears that actual frequency response curve is significantly lower than target frequency response curve in frequency range from 3000Hz to 7000Hz.
In order to avoid wave filter is frequently adjusted, it is negligible less than the frequency response difference of default first threshold (such as, 6dB), ratio such use-case ignoring less than 6dB, the two-dimensional array of a frequency response difference can be drawn, as follows:
f1 f2 f3 f4 f5 f6 f7 f8 f9 f10 f11
0 0 0 0 0 0 0 8 6 0 0
In practical application, second filtering adjusting module in wave filter, be in order to adapt to the frequency response of locally received equipment (such as, receiver or speaker), cavity response and design.
It is assumed that the two-dimensional array of target frequency response curve is as follows:
f1 f2 f3 f4 f5 f6 f7 f8 f9 f10 f11
a1 a2 a3 a4 a5 a6 a7 a8 a9 a10 a11
So, finally need to be added together frequency response difference and actual frequency response curve, form the two-dimensional array of final frequency response curve (frequency response curve after namely adjusting), as follows:
f1 f2 f3 f4 f5 f6 f7 f8 f9 f10 f11
a1 a2 a3 a4 a5 a6 a7 a8+8 a9+6 a10 a11
The two-dimensional array of frequency response curve after this being adjusted imports in IIR (InfiniteImpulseResponse, infinite impulse response) wave filter, it can be deduced that the filter parameter that frequency response curve after adjustment is corresponding.As filtering, filter parameter corresponding for the frequency response curve after adjusting is adjusted parameter be input in the second filtering adjusting module.
The present inventor considers, when the voice signal on sending direction exists key words such as " louder " " not hearing ", being probably the user of the machine and think that the sound oneself heard is big not, tonequality is clear not, expects that partner can improve volume or speaking clearly.
Therefore, in the embodiment of the present invention, pre-set some key words, when comprising default key word in the voice signal of user's output, it is possible to the voice signal received is adjusted.Specifically, after call starts, it is possible to actively intercept the voice signal on sending direction;For the voice signal on sending direction, identify in the voice signal on sending direction whether there is default key word, if recognizing, then can adjust parameter according to default volume adjusting parameters and/or filtering, adjust recipient's voice signal upwards.
More preferably, voice signal on sending direction can be carried out pretreatment, the voice signal obtained after pretreatment is carried out Semantic detection, identify in the voice signal on sending direction whether there is default key word, if identifying default key word from the voice signal sending direction, then default volume adjusting parameters and/or filtering are adjusted parameter and is defined as voice adjustment parameter.
Wherein, it is default parameters that the volume adjusting parameters preset and the filtering preset adjust parameter, it is possible to rule of thumb arranged by those skilled in the art.
S303: adjust parameter according to the voice determined, is transferred to speaker after adjusting the signal intensity of voice signal.
In the embodiment of the present invention, after being determined that by step S302 voice adjusts parameter, it is possible to adjust parameter generation according to voice and adjust instruction accordingly;And the adjustment instruction of generation is exported corresponding second gain regulation module or second filtering adjusting module be adjusted.
Specifically, after determining corresponding volume adjusting parameters based on volume detection, corresponding volume can be generated according to the volume adjusting parameters determined and adjust instruction (herein, can be described as the volume generated based on volume detection and adjust instruction), and export to the second gain regulation module.
After determining that corresponding filtering adjusts parameter based on frequency response detection, parameter can be adjusted according to the filtering determined and generate the adjustment instruction of corresponding wave filter (herein, can be described as the wave filter generated based on frequency response detection and adjust instruction), and export to the second filtering adjusting module.
So, the second filtering adjusting module can adjust instruction according to the wave filter received, and is filtered adjusting to recipient's voice signal upwards, so that the frequency characteristic corresponding to voice signal after adjusting and the target frequency response curve proximity preset;Second gain regulation module can adjust instruction according to the volume received, and recipient's voice signal upwards is carried out Gain tuning, obtains customer satisfaction system volume with this.
After the volume adjusting parameters preset determined based on Semantic detection and/or filtering adjust parameter, generate corresponding volume according to default volume adjusting parameters and adjust instruction (herein, can be described as the volume generated based on Semantic detection and adjust instruction), and export to the second gain regulation module;Adjust the parameter corresponding wave filter of generation according to default filtering and adjust instruction (herein, can be described as the wave filter generated based on Semantic detection and adjust instruction), and export to the second filtering adjusting module.
In the embodiment of the present invention, no matter whether the second gain regulation module has been based on the volume adjustment instruction adjustment voice signal that volume detection generates, and second filtering adjusting module whether have been based on frequency response detection generate wave filter adjust instruction adjust voice signal, when there is the key word preset in the voice signal detected on sending direction, capital adjusts instruction according to the volume generated based on Semantic detection, wave filter adjusts instruction, heightens volume and/or heightens the frequency response of HFS.
Therefore, if the second filtering adjusting module had both received the wave filter generated based on frequency response detection adjusts instruction, also receive the wave filter generated based on Semantic detection and adjust instruction, then, the two adjustment order is carried out by the second filtering adjusting module.Similarly, if the second gain regulation module had both received the volume generated based on volume detection and adjusted instruction, also receive the volume generated based on Semantic detection and adjust instruction, then the two adjustment order is carried out by the second gain regulation module.
By the method promoting call tone quality that the embodiment of the present invention provides, can before recipient's voice signal upwards be sent to user, actively go to receive during detection call the tonequality of the voice signal on direction and sending direction, and start corresponding mechanism and signal is repaired and corrects, carry out adapting to adjust according to the practical situation of call with this, improve call tone quality, meet user and improve the actual demand of call tone quality, improve Consumer's Experience.
Further, it is contemplated that, the voice adjustment parameter that voice adjusts in data base can be determined according to mobile phone feature when conversing before or people's feature of talking.
Therefore, in the embodiment of the present invention, adjust data base for the ease of more new speech, it is possible to the voice determined by step S302 adjusts parameter and is temporarily stored in temporary data storage unit.
Specifically, it is possible to the filtering adjustment parameter volume adjusting parameters determined will be detected based on volume, determining based on frequency response detection, volume adjusting parameters and/or filtering and based on Semantic detection generation adjust parameter and are stored in temporary data storage unit.
So, after this end of conversation, it is possible to according to the information of storage in temporary data storage unit, it is determined that go out the voice that in this call, in each time period, voice signal is corresponding and adjust parameter.Afterwards, adjusting parameter according to the voice that voice signal in each time period in this call is corresponding, the average speech calculating this call adjusts parameter;And with the telephone number of partner for index, set up average speech and adjust the corresponding relation between parameter and the telephone number of partner, and update in voice adjustment data base.
Specifically, it is possible to the volume calculating this call on average adjusts parameter, filtering on average adjusts parameter;And with the telephone number of partner for index, volume is on average adjusted parameter, filtering on average adjustment parameter and updates in voice adjustment data base.
More preferably, in the embodiment of the present invention, except can except active mating debit voice signal upwards be adjusted, it is also possible to provide signal quality detection warning function.
It is considered that when local signal is second-rate, it is possible to call can be made to occur discontinuously, the problem such as single-pass, the quality of impact call.
Therefore, in the embodiment of the present invention, after adjusting voice signal, it is also possible to detection local signal quality.Wherein, local signal quality can be exported to tonequality hoisting module by radio-frequency (RF) receiving and transmission module detection as signal quality parameter.If local signal quality is lower than default threshold value, then can add default prompt tone in a receive direction.Such as, " you Location signal difference, window, open field please be find ".Wherein, prompting interval can set fixed value as required, it is also possible to arranges change advisory frequency according to signal quality situation.Such as it is spaced apart 20 seconds once lower than threshold value 3dB prompting, is spaced apart 10 seconds once lower than threshold value 6dB prompting.In practical application, prompt tone playing process interrupts the other side's one's voice in speech, and volume also takes less numerical value and avoids interference call.
Further, the present inventor is it is considered that in order to promote call tone quality, it is also possible to whether the signal that detection the other side sends exists discontinuously.Even if owing to voice signal the other side of normal talking keeps silence, also can there is certain end makes an uproar (i.e. comfort noise, background noise).And when one's own side or the other side's signal difference time, can there is certain null frame when reduction decoding in voice signal.The null frame of the signal sent as the other side exceedes certain threshold value, then the wireless link existing problems of current talking are described.
Therefore, in the embodiment of the present invention, after adjusting voice signal, except detection local signal quality, it is also possible to detection recipient's signal null frame rate upwards.Wherein, recipient's signal null frame rate upwards can also be exported to tonequality hoisting module by radio-frequency (RF) receiving and transmission module detection as signal quality parameter.If the signal null frame rate that recipient is upwards exceedes the null frame threshold value of setting, and local signal quality is more than or equal to default threshold value, then add default prompt tone in a transmit direction.Such as, " the other side's mobile phone You are prompted with, you are likely to be at weak signal area, please near window or open field ".
Based on the method for above-mentioned lifting call tone quality, the embodiment of the present invention additionally provides a kind of mobile terminal, and as shown in Figure 5 a, mobile terminal may include that voice signal read module 501, adjusts parameter determination module 502, call tone quality adjusting module 503.
Wherein, after voice signal read module 501 starts for call, read the voice signal that recipient upwards decodes.
Adjust parameter determination module 502 and carry out the detection of signal intensity for the voice signal that voice signal read module 501 is read, it is determined that go out corresponding voice and adjust parameter.
Wherein, voice adjust parameter can include following at least one: volume adjusting parameters, filtering adjust parameter.
In the embodiment of the present invention, before the voice signal that voice signal read module 501 reads is detected by adjustment parameter determination module 502, it is also possible to voice signal is carried out pretreatment.Wherein, pretreatment can include following at least one: effectively sound extracts, non-voice filters, audio segmentation, FFT process.
Correspondingly, adjust parameter determination module 502 and the voice signal after pretreatment is carried out tonequality detection, it is determined that go out corresponding voice and adjust parameter.
Specifically, parameter determination module 502 is adjusted for default each time period, based on voice signal signal intensity on each time point in this time period, it is determined that the signal energy equivalence value of voice signal in this time period;If it is determined that signal energy equivalence value less than default volume threshold, then the volume difference between signal energy equivalence value and volume threshold is defined as the volume adjusting parameters that in this time period, voice signal is corresponding.
Further, each time period that parameter determination module 502 can also be for presetting is adjusted, based on voice signal signal intensity on each time point in this time period, it is determined that go out the actual frequency response curve of voice signal in this time period;Calculate actual frequency response curve and default target frequency response curve frequency response difference in each frequency range;Determine the frequency response difference quantity more than the frequency range of first threshold;If it is determined that quantity more than Second Threshold, then will be greater than the frequency response difference of first threshold and corresponding band value in actual frequency response curve and add up, the frequency response curve after being adjusted;By filter parameter corresponding for the frequency response curve after adjusting, it is determined that the filtering corresponding for voice signal in this time period adjusts parameter.
More preferably, adjust parameter determination module 502 can also identify whether there is default key word in the voice signal on sending direction, if identifying default key word from the voice signal sending direction, then default volume adjusting parameters and/or filtering are adjusted parameter and is defined as voice adjustment parameter.
In the embodiment of the present invention, call tone quality adjusting module 503 is for according to adjusting the voice adjustment parameter that parameter determination module 502 is determined, being transferred to speaker after adjusting the signal intensity of voice signal.
Further, call tone quality adjusting module 503 after call starts, can also adjust from voice and search the voice adjustment parameter corresponding with the telephone number of partner data base;If finding out, then adjust parameter according to the voice found out, voice signal is initially adjusted;If not finding out, then the voice signal read being carried out the detection of signal intensity, adjusting at voice and data base determining, corresponding voice adjusts parameter;Adjust parameter according to the voice determined, after adjusting the signal intensity of voice signal, be transferred to speaker.
More preferably, as shown in Figure 5 b, the device promoting call tone quality that the embodiment of the present invention provides can also include: database update module 504.
Database update module 504 is for adjusting parameter according to the voice of voice signal in each time period in this call, and the average speech calculating this call adjusts parameter;Set up average speech and adjust the corresponding relation between parameter and the telephone number of partner, and update in voice adjustment data base.
More preferably, as shown in Figure 5 b, the device promoting call tone quality that the embodiment of the present invention provides can also include: voice cue module 505.
Voice cue module 505 is for, after call tone quality adjusting module 503 adjusts voice signal, if local signal quality is lower than default threshold value, then adding default prompt tone in a receive direction.
Further, if the signal null frame rate that recipient is upwards exceedes the null frame threshold value of setting, and local signal quality is more than or equal to default threshold value, then voice cue module 505 can also add default prompt tone in a transmit direction.
In the embodiment of the present invention, as shown in Figure 6, adjust parameter determination module 502 and may include that volume detecting unit 601, frequency response detection unit 602 and/or Semantic detection unit 603.
Correspondingly, voice adjusts parameter and may include that volume adjusting parameters, filtering adjust parameter.
Volume detecting unit 601 is for for default each time period, based on voice signal signal intensity on each time point in this time period, it is determined that the signal energy equivalence value of voice signal in this time period;If it is determined that signal energy equivalence value less than default volume threshold, then the volume difference between signal energy equivalence value and volume threshold is defined as the volume adjusting parameters that in this time period, voice signal is corresponding.
Frequency response detection unit 602 is for for default each time period, based on voice signal signal intensity on each time point in this time period, it is determined that go out the actual frequency response curve of voice signal in this time period;Calculate actual frequency response curve and default target frequency response curve frequency response difference in each frequency range;Determine the frequency response difference quantity more than the frequency range of first threshold;If it is determined that quantity more than Second Threshold, then will be greater than the frequency response difference of first threshold and corresponding band value in actual frequency response curve and add up, the frequency response curve after being adjusted;By filter parameter corresponding for the frequency response curve after adjusting, it is determined that the filtering corresponding for voice signal in this time period adjusts parameter.
Whether Semantic detection unit 603 exists default key word for identifying in the voice signal on sending direction, if identifying default key word from the voice signal sending direction, then default volume adjusting parameters and/or filtering are adjusted parameter and is defined as voice adjustment parameter.
In the embodiment of the present invention, the concrete function of each module in above-mentioned mobile terminal and each unit under module realizes promoting implementing of each step in the method for call tone quality, is not described in detail in this.
In technical scheme, Regulation mechanism of taking the initiative, before transmitting voice signal recipient upwards decoded is to speaker, actively reads the voice signal that recipient upwards decodes;And based on the signal intensity of the voice signal read, it is determined that go out corresponding voice and adjust parameter.Compare existing by fixing gain or filtering adjust voice signal, the solution of the present invention according to the practical situation of currently received voice signal, can carry out accommodation, meets user and improves the actual demand of call tone quality, improves Consumer's Experience.
Those skilled in the art of the present technique are appreciated that the present invention includes the one or more equipment relating to perform in operation described herein.These equipment can specialized designs and manufacture for required purpose, or the known device in general purpose computer can also be included.These equipment have storage computer program within it, and these computer programs optionally activate or reconstruct.nullSuch computer program can be stored in equipment (such as,Computer) in computer-readable recording medium or be stored in and be suitable to storage e-command and be coupled to any kind of medium of bus respectively,Described computer-readable medium includes but not limited to that any kind of dish (includes floppy disk、Hard disk、CD、CD-ROM、And magneto-optic disk)、ROM(Read-OnlyMemory,Read only memory)、RAM(RandomAccessMemory,Memorizer immediately)、EPROM(ErasableProgrammableRead-OnlyMemory,Erarable Programmable Read only Memory)、EEPROM(ElectricallyErasableProgrammableRead-OnlyMemory,EEPROM)、Flash memory、Magnetic card or light card.It is, computer-readable recording medium include by equipment (such as, computer) with can read form storage or transmission information any medium.
Those skilled in the art of the present technique are appreciated that, it is possible to the calculation machine programmed instruction combination to the frame in each frame realizing in these structure charts and/or block diagram and/or flow graph and these structure charts and/or block diagram and/or flow graph of using tricks.Those skilled in the art of the present technique are appreciated that, the processor that these computer program instructions can be supplied to general purpose computer, special purpose computer or other programmable data processing methods realizes, and performs the scheme specified in the frame of structure chart disclosed by the invention and/or block diagram and/or flow graph or multiple frame thereby through the processor of computer or other programmable data processing methods.
Those skilled in the art of the present technique are appreciated that the step in the various operations discussed in the present invention, method, flow process, measure, scheme can be replaced, change, combine or delete.Further, have the various operations discussed in the present invention, method, other steps in flow process, measure, scheme can also be replaced, changed, reset, decomposed, combined or deleted.Further, of the prior art have with the present invention disclosed in various operations, method, the step in flow process, measure, scheme can also be replaced, changed, reset, decomposed, combined or deleted.
The above is only the some embodiments of the present invention; it should be pointed out that, for those skilled in the art, under the premise without departing from the principles of the invention; can also making some improvements and modifications, these improvements and modifications also should be regarded as protection scope of the present invention.

Claims (10)

1. the method promoting mobile terminal call tonequality, it is characterised in that including:
After call starts, read the voice signal that recipient upwards decodes;
The voice signal read is carried out the detection of signal intensity, it is determined that go out corresponding voice and adjust parameter;
Adjust parameter according to described voice, after adjusting the signal intensity of described voice signal, be transferred to speaker;
Wherein, described voice adjust parameter include following at least one: volume adjusting parameters, filtering adjust parameter.
2. method according to claim 1, it is characterised in that the described voice signal to reading carries out the detection of signal intensity, it is determined that go out corresponding voice and adjust parameter, specifically include:
For default each time period, based on voice signal signal intensity on each time point in this time period, it is determined that the signal energy equivalence value of voice signal in this time period;
If it is determined that signal energy equivalence value less than default volume threshold, then the volume difference between signal energy equivalence value with described volume threshold is defined as the volume adjusting parameters that in this time period, voice signal is corresponding.
3. method according to claim 1, it is characterised in that the described voice signal to reading carries out the detection of signal intensity, it is determined that go out corresponding voice and adjust parameter, specifically include:
For default each time period, based on voice signal signal intensity on each time point in this time period, it is determined that go out the actual frequency response curve of voice signal in this time period;Calculate described actual frequency response curve and default target frequency response curve frequency response difference in each frequency range;
Determine the frequency response difference quantity more than the frequency range of first threshold;If it is determined that quantity more than Second Threshold, then will be greater than the frequency response difference of first threshold and corresponding band value in described actual frequency response curve and add up, the frequency response curve after being adjusted;By filter parameter corresponding for the frequency response curve after adjusting, it is determined that the filtering corresponding for voice signal in this time period adjusts parameter.
4. method according to claim 1, it is characterised in that call also includes after starting:
Obtain the telephone number of partner;
Adjust from voice and data base searches the voice adjustment parameter corresponding with the telephone number of partner;
If finding out, then adjust parameter according to the voice found out, described voice signal is initially adjusted.
5. according to the arbitrary described method of claim 1-4, it is characterised in that after the described voice signal of described adjustment, also include:
Adjusting parameter according to the voice that voice signal in each time period in this call is corresponding, the average speech calculating this call adjusts parameter;
Set up described average speech and adjust the corresponding relation between parameter and the telephone number of partner, and update in voice adjustment data base.
6. a mobile terminal, it is characterised in that including:
Voice signal read module, after starting for call, reads the voice signal that recipient upwards decodes;
Adjust parameter determination module, for the voice signal read being carried out the detection of signal intensity, it is determined that go out corresponding voice and adjust parameter;Wherein, described voice adjust parameter include following at least one: volume adjusting parameters, filtering adjust parameter;
Call tone quality adjusting module, adjusts parameter for the voice determined according to described adjustment parameter determination module, is transferred to speaker after adjusting the signal intensity of described voice signal.
7. mobile terminal according to claim 6, it is characterised in that
Described adjustment parameter determination module is specifically for for default each time period, based on voice signal signal intensity on each time point in this time period, it is determined that the signal energy equivalence value of voice signal in this time period;If it is determined that signal energy equivalence value less than default volume threshold, then the volume difference between signal energy equivalence value with described volume threshold is defined as the volume adjusting parameters that in this time period, voice signal is corresponding.
8. mobile terminal according to claim 7, it is characterised in that
Described adjustment parameter determination module is additionally operable to for default each time period, based on voice signal signal intensity on each time point in this time period, it is determined that go out the actual frequency response curve of voice signal in this time period;Calculate described actual frequency response curve and default target frequency response curve frequency response difference in each frequency range;Determine the frequency response difference quantity more than the frequency range of first threshold;If it is determined that quantity more than Second Threshold, then will be greater than the frequency response difference of first threshold and corresponding band value in described actual frequency response curve and add up, the frequency response curve after being adjusted;By filter parameter corresponding for the frequency response curve after adjusting, it is determined that the filtering corresponding for voice signal in this time period adjusts parameter.
9. mobile terminal according to claim 6, it is characterised in that
Described call tone quality adjusting module is additionally operable to obtain the telephone number of partner, adjusts from voice and searches the voice adjustment parameter corresponding with the telephone number of partner data base;If finding out, then adjust parameter according to the voice found out, described voice signal is initially adjusted.
10. according to the arbitrary described mobile terminal of claim 6-9, it is characterised in that also include:
Database update module, adjusts parameter for the voice corresponding according to voice signal in each time period in this call, and the average speech calculating this call adjusts parameter;Set up described average speech and adjust the corresponding relation between parameter and the telephone number of partner, and update in voice adjustment data base.
CN201610281071.2A 2016-04-29 2016-04-29 Mobile terminal and the method for promoting mobile terminal call sound quality Active CN105744084B (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
CN201610281071.2A CN105744084B (en) 2016-04-29 2016-04-29 Mobile terminal and the method for promoting mobile terminal call sound quality

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
CN201610281071.2A CN105744084B (en) 2016-04-29 2016-04-29 Mobile terminal and the method for promoting mobile terminal call sound quality

Publications (2)

Publication Number Publication Date
CN105744084A true CN105744084A (en) 2016-07-06
CN105744084B CN105744084B (en) 2019-05-07

Family

ID=56287774

Family Applications (1)

Application Number Title Priority Date Filing Date
CN201610281071.2A Active CN105744084B (en) 2016-04-29 2016-04-29 Mobile terminal and the method for promoting mobile terminal call sound quality

Country Status (1)

Country Link
CN (1) CN105744084B (en)

Cited By (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN106210260A (en) * 2016-06-17 2016-12-07 乐视控股(北京)有限公司 Communicating data processing method and processing device
CN106559568A (en) * 2016-11-15 2017-04-05 深圳天珑无线科技有限公司 Call terminal and its detection MIC send the words method that hole is blocked
CN107154263A (en) * 2017-05-25 2017-09-12 宇龙计算机通信科技(深圳)有限公司 Sound processing method, device and electronic equipment
CN107331404A (en) * 2017-06-22 2017-11-07 深圳传音通讯有限公司 The sound processing method and device of audio frequency and video
CN109361827A (en) * 2018-10-22 2019-02-19 杭州叙简科技股份有限公司 A kind of secondary suppressing method of the echo of communication terminal
CN113055786A (en) * 2021-03-30 2021-06-29 联想(北京)有限公司 Volume control method and device and electronic equipment
CN115695642A (en) * 2022-09-26 2023-02-03 展讯通信(天津)有限公司 Method and device for adjusting call volume
WO2023082603A1 (en) * 2021-11-09 2023-05-19 深圳传音控股股份有限公司 Reminding method, terminal device, network device and storage medium

Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20090074207A1 (en) * 2007-09-17 2009-03-19 Samsung Electronics Co., Ltd. Mobile communication device capable of setting tone color and method of setting tone color
CN103714824A (en) * 2013-12-12 2014-04-09 小米科技有限责任公司 Audio processing method, audio processing device and terminal equipment
CN103873625A (en) * 2014-03-31 2014-06-18 深圳市中兴移动通信有限公司 Method and device for increasing volume of received voice and mobile terminal
CN104506702A (en) * 2014-11-18 2015-04-08 深圳市金立通信设备有限公司 Volume adjusting method
CN105049632A (en) * 2015-08-17 2015-11-11 联想(北京)有限公司 Call volume adjustment method and electronic equipment

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20090074207A1 (en) * 2007-09-17 2009-03-19 Samsung Electronics Co., Ltd. Mobile communication device capable of setting tone color and method of setting tone color
CN103714824A (en) * 2013-12-12 2014-04-09 小米科技有限责任公司 Audio processing method, audio processing device and terminal equipment
CN103873625A (en) * 2014-03-31 2014-06-18 深圳市中兴移动通信有限公司 Method and device for increasing volume of received voice and mobile terminal
CN104506702A (en) * 2014-11-18 2015-04-08 深圳市金立通信设备有限公司 Volume adjusting method
CN105049632A (en) * 2015-08-17 2015-11-11 联想(北京)有限公司 Call volume adjustment method and electronic equipment

Cited By (9)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN106210260A (en) * 2016-06-17 2016-12-07 乐视控股(北京)有限公司 Communicating data processing method and processing device
CN106559568A (en) * 2016-11-15 2017-04-05 深圳天珑无线科技有限公司 Call terminal and its detection MIC send the words method that hole is blocked
CN107154263A (en) * 2017-05-25 2017-09-12 宇龙计算机通信科技(深圳)有限公司 Sound processing method, device and electronic equipment
CN107331404A (en) * 2017-06-22 2017-11-07 深圳传音通讯有限公司 The sound processing method and device of audio frequency and video
CN109361827A (en) * 2018-10-22 2019-02-19 杭州叙简科技股份有限公司 A kind of secondary suppressing method of the echo of communication terminal
CN109361827B (en) * 2018-10-22 2021-02-09 杭州叙简科技股份有限公司 Echo secondary suppression method for communication terminal
CN113055786A (en) * 2021-03-30 2021-06-29 联想(北京)有限公司 Volume control method and device and electronic equipment
WO2023082603A1 (en) * 2021-11-09 2023-05-19 深圳传音控股股份有限公司 Reminding method, terminal device, network device and storage medium
CN115695642A (en) * 2022-09-26 2023-02-03 展讯通信(天津)有限公司 Method and device for adjusting call volume

Also Published As

Publication number Publication date
CN105744084B (en) 2019-05-07

Similar Documents

Publication Publication Date Title
CN105744084A (en) Mobile terminal and method for improving conversation tone quality thereof
CN107895578B (en) Voice interaction method and device
CN104954555B (en) A kind of volume adjusting method and system
US6138040A (en) Method for suppressing speaker activation in a portable communication device operated in a speakerphone mode
US20120263317A1 (en) Systems, methods, apparatus, and computer readable media for equalization
US8824666B2 (en) Noise cancellation for phone conversation
CN107995360B (en) Call processing method and related product
KR950015199A (en) Speech recognition method and device
CN104067341A (en) Voice activity detection in presence of background noise
US20090018843A1 (en) Speech processor and communication terminal device
US20080312916A1 (en) Receiver Intelligibility Enhancement System
US20140365212A1 (en) Receiver Intelligibility Enhancement System
US8392187B2 (en) Dynamic pruning for automatic speech recognition
KR20240033108A (en) Voice Aware Audio System and Method
US20180174574A1 (en) Methods and systems for reducing false alarms in keyword detection
KR20100068188A (en) Method for signal separation, communication system and voice recognition system using the method
CN113542960B (en) Audio signal processing method, system, device, electronic equipment and storage medium
WO2015152937A1 (en) Modifying sound output in personal communication device
US20140236590A1 (en) Communication apparatus and voice processing method therefor
JPH10322441A (en) Hand-free telephone set
CN115482830A (en) Speech enhancement method and related equipment
US8868417B2 (en) Handset intelligibility enhancement system using adaptive filters and signal buffers
CN117480554A (en) Voice enhancement method and related equipment
US8868418B2 (en) Receiver intelligibility enhancement system
WO2019228329A1 (en) Personal hearing device, external sound processing device, and related computer program product

Legal Events

Date Code Title Description
C06 Publication
PB01 Publication
C10 Entry into substantive examination
SE01 Entry into force of request for substantive examination
GR01 Patent grant
GR01 Patent grant