WO2015158182A1 - 一种用户终端的音量调节方法、装置及终端 - Google Patents

一种用户终端的音量调节方法、装置及终端 Download PDF

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Publication number
WO2015158182A1
WO2015158182A1 PCT/CN2015/072906 CN2015072906W WO2015158182A1 WO 2015158182 A1 WO2015158182 A1 WO 2015158182A1 CN 2015072906 W CN2015072906 W CN 2015072906W WO 2015158182 A1 WO2015158182 A1 WO 2015158182A1
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WO
WIPO (PCT)
Prior art keywords
volume
sound
user terminal
audio data
sound signal
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PCT/CN2015/072906
Other languages
English (en)
French (fr)
Inventor
刘冬
Original Assignee
华为技术有限公司
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 华为技术有限公司 filed Critical 华为技术有限公司
Priority to EP15780360.2A priority Critical patent/EP3133799B1/en
Priority to EP19173795.6A priority patent/EP3611910B1/en
Priority to JP2016563102A priority patent/JP6381153B2/ja
Priority to KR1020167032006A priority patent/KR101884709B1/ko
Publication of WO2015158182A1 publication Critical patent/WO2015158182A1/zh
Priority to US15/293,372 priority patent/US9866707B2/en
Priority to US15/825,911 priority patent/US10200545B2/en
Priority to US16/230,060 priority patent/US10554826B2/en
Priority to US16/246,729 priority patent/US10516788B2/en
Priority to US16/735,926 priority patent/US11044369B2/en
Priority to US17/325,918 priority patent/US11483434B2/en

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M1/00—Substation equipment, e.g. for use by subscribers
    • H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/724—User interfaces specially adapted for cordless or mobile telephones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M19/00—Current supply arrangements for telephone systems
    • H04M19/02—Current supply arrangements for telephone systems providing ringing current or supervisory tones, e.g. dialling tone or busy tone
    • H04M19/04—Current supply arrangements for telephone systems providing ringing current or supervisory tones, e.g. dialling tone or busy tone the ringing-current being generated at the substations
    • H04M19/042—Current supply arrangements for telephone systems providing ringing current or supervisory tones, e.g. dialling tone or busy tone the ringing-current being generated at the substations with variable loudness of the ringing tone, e.g. variable envelope or amplitude of ring signal
    • H04M19/044—Current supply arrangements for telephone systems providing ringing current or supervisory tones, e.g. dialling tone or busy tone the ringing-current being generated at the substations with variable loudness of the ringing tone, e.g. variable envelope or amplitude of ring signal according to the level of ambient noise
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G10L21/0216—Noise filtering characterised by the method used for estimating noise
    • G10L21/0232—Processing in the frequency domain
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G3/00—Gain control in amplifiers or frequency changers
    • H03G3/20—Automatic control
    • H03G3/30—Automatic control in amplifiers having semiconductor devices
    • H03G3/32—Automatic control in amplifiers having semiconductor devices the control being dependent upon ambient noise level or sound level
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M1/00—Substation equipment, e.g. for use by subscribers
    • H04M1/60—Substation equipment, e.g. for use by subscribers including speech amplifiers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M1/00—Substation equipment, e.g. for use by subscribers
    • H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/724—User interfaces specially adapted for cordless or mobile telephones
    • H04M1/72448—User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions
    • H04M1/72454—User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions according to context-related or environment-related conditions
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M1/00—Substation equipment, e.g. for use by subscribers
    • H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/725—Cordless telephones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M2250/00—Details of telephonic subscriber devices
    • H04M2250/12—Details of telephonic subscriber devices including a sensor for measuring a physical value, e.g. temperature or motion
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W88/00—Devices specially adapted for wireless communication networks, e.g. terminals, base stations or access point devices
    • H04W88/02—Terminal devices

Definitions

  • the present invention relates to the field of communications technologies, and in particular, to a method, device, and terminal for adjusting a volume of a user terminal.
  • the volume of the ringing of most mobile phones and the volume of the call of the handset are usually manually adjusted by the user, or a simple scene that can be self-aware, such as an indoor mode or an outdoor mode, is integrated.
  • the self-perceived scene mode is used to adjust the playing volume of the ringing of the mobile phone
  • the number of decibels of the ambient sound is generally extracted and judged by the sound detecting module, and then the decibel number of the pre-stored sound is Adjust the ringtone and the volume of the handset by the corresponding relationship of the incoming call ring volume.
  • the above-mentioned volume adjustment method only determines the final ringing and earpiece volume according to the decibel number of the ambient sound, but the decibel number of the environmental sound does not accurately reflect the situation in which the user is located, and if there is a higher person in the extracted environmental sound. Sounds can also be mistaken for noise, such as discussing more intense meetings or presentations, press conferences, etc. In these situations, the volume needs to be turned down, but the volume determined by the decibels of the ambient sound is relatively large, resulting in a mobile phone. The volume adjustment is not true and the accuracy is not high.
  • Embodiments of the present invention provide a method, a device, and a terminal for adjusting a volume of a user terminal, so as to enable The volume of the mobile phone is adaptive and more precisely matches the situation in which the user is located, improving the user experience.
  • a method for adjusting a volume of a user terminal includes:
  • composition information includes: a sound type included in the sound signal, and a specific gravity of each type of sound; wherein the sound type Includes: blank sounds, human sounds, and noise;
  • the volume of the user terminal is adjusted according to the determined scene mode.
  • the collecting the sound signal around the user terminal includes:
  • the adjusting a volume of the user terminal according to the determined scenario mode includes:
  • the volume of the user terminal includes: a ringing volume and an earpiece volume; and the volume adjustment coefficient includes: a ringing volume adjustment coefficient And the volume adjustment factor of the handset;
  • Adjusting the volume of the user terminal according to the volume adjustment coefficient including:
  • the method further includes:
  • the handset volume of the user terminal is lowered.
  • the method further includes:
  • the reference volume is averaged and stored as a new reference volume.
  • the analyzing the collected sound signal to obtain the composition information of the sound signal specifically including :
  • the audio data is classified according to the calculated sound frequency according to the blank sound, the human voice, and the noise, and specifically includes:
  • the audio data When determining that the sound frequency of the audio data is within a range of 20 Hz to 20,000 Hz, calculating a fundamental frequency of the audio data, and determining that the basic frequency is within a range of 85 Hz to 255 Hz, the audio data is considered to be a human voice; when the base frequency is judged to be outside the range of 85 Hz to 255 Hz, the audio data is considered to be noise;
  • the audio data is considered to be a blank sound.
  • a volume adjustment apparatus for a user terminal including:
  • An acquisition unit configured to collect a sound signal around the user terminal
  • An analysis unit configured to analyze the collected sound signal to obtain composition information of the sound signal;
  • the composition information includes: a sound type included in the sound signal, and a specific gravity of each type of sound;
  • the types of sounds include: blank sounds, human sounds, and noise;
  • a scene mode determining unit configured to determine a current scene mode of the user terminal according to the composition information of the sound signal
  • a volume adjustment unit configured to adjust a volume of the user terminal according to the determined scene mode.
  • the collecting unit is specifically configured to: when detecting that a call signal arrives, collect a sound signal of a current environment; or periodically collect the current environment. Sound signal.
  • the volume adjustment unit is specifically configured to use the determined scenario mode and the pre-stored scenario mode Corresponding relationship between the volume adjustment coefficients, determining a volume adjustment coefficient, and adjusting a volume of the user terminal according to the volume adjustment coefficient.
  • the volume of the user terminal includes: a ringing volume and an earpiece volume; and the volume adjustment coefficient in the volume adjusting unit specifically includes : ringer volume adjustment factor and handset volume adjustment factor;
  • the volume adjustment unit is configured to adjust a ringing volume of the user terminal according to the ringing volume adjustment coefficient, and adjust an earpiece of the user terminal according to the earpiece volume adjustment coefficient. volume.
  • the first possible implementation of the second aspect, the second possible implementation of the second aspect, or the third possible implementation of the second aspect, in a fourth possible implementation manner Also includes:
  • An acquiring unit configured to acquire a microphone volume of the user terminal in real time after detecting that the call signal is turned on;
  • a comparison unit configured to compare the obtained microphone volume with a pre-stored reference volume
  • the volume adjustment unit is further configured to increase an earpiece volume of the user terminal when the acquired microphone volume is greater than a pre-stored reference volume; and decrease the volume when the acquired microphone volume is less than a pre-stored reference volume The handset volume of the user terminal.
  • the acquiring unit is configured to periodically collect, after detecting that the call signal is turned on, periodically Sound signal
  • the analyzing unit is further configured to analyze, by the collecting module, the sound signal periodically collected after detecting that the call signal is turned on, to obtain composition information of the sound signal; and according to the composition information of the sound signal Determining whether the sound signal contains a human voice and only including one person's voice; if yes, calculating a volume of the sound signal, and averaging the calculated volume of the sound signal and the previously stored reference volume , stored as a new reference volume.
  • the analyzing unit includes:
  • a first processing unit configured to divide the collected sound signal into multiple pieces of audio data
  • a second processing unit configured to calculate a sound frequency of each piece of audio data, and classify each piece of audio data according to the blank sound, the human voice, and the noise according to the calculated sound frequency;
  • a third processing unit for counting the proportion of blank sound, human sound and noise in all audio data
  • a fourth processing unit configured to calculate a Mel frequency cepstrum coefficient for the audio data determined to be a human voice, and perform audio statistics of the same frequency of the Mel frequency cepstral coefficient as a person's voice to determine the human voice The number of people included.
  • the second processing unit is specifically configured to determine whether a sound frequency of each piece of audio data is in a range of 20 Hz to 20,000 Hz;
  • the audio data is considered to be a human voice;
  • the base frequency is judged to be outside the range of 85 Hz to 255 Hz, the audio data is considered to be noise; and when the sound frequency of the audio data is judged to be outside the range of 20 Hz to 20,000 Hz, the audio is considered The data is a blank sound.
  • a terminal including: a speaker and an earpiece; and the volume adjustment device provided by the embodiment of the present invention is further included.
  • the method, device and terminal for adjusting the volume of the user terminal provided by the embodiment of the invention have the following beneficial effects:
  • the corresponding current scene is closer to the real scene, and can be more accurately matched with the situation in which the user is located, greatly reducing The user's experience is improved by the fact that the playback volume adjustment caused by the misjudgment scene does not conform to the scene.
  • FIG. 1 is a schematic flowchart of a volume adjustment method according to an embodiment of the present invention.
  • FIG. 2 is a second schematic flowchart of a volume adjustment method according to an example of the present invention.
  • FIG. 3 is a third schematic flowchart of a volume adjustment method according to an example of the present invention.
  • FIG. 4 is a fourth schematic flowchart of a volume adjustment method according to an example of the present invention.
  • FIG. 5 is a fifth schematic flowchart of a volume adjustment method according to an example of the present invention.
  • FIG. 6 is a schematic structural diagram of a volume adjustment apparatus according to an example of the present invention.
  • FIG. 7 is a schematic structural diagram of a user terminal according to an embodiment of the present invention.
  • the present invention provides a volume adjustment method for a user terminal, in order to solve the problem that the existing user terminal does not automatically match the scene when the volume is automatically adjusted according to the surrounding environment, and the adjusted volume does not match the real scene.
  • the device and the terminal can accurately analyze the ambient sound, and the matched current scene is closer to the real scene, thereby adjusting the appropriate volume, which can greatly reduce the situation that the volume adjustment caused by the misjudgment scene does not conform to the scene.
  • the volume adjustment method of the user terminal provided by the embodiment of the present invention can be mainly applied to communication terminals such as a mobile phone or a walkie-talkie, for example, the mobile phone can temporarily not ring when the call signal is accessed, and the volume provided by the embodiment of the present invention is provided. After the adjustment method determines the volume, control the appropriate ringer volume and control the appropriate earpiece volume after the call is turned on.
  • the volume adjustment method provided by the embodiment of the present invention can also be applied to, for example, a mobile TV terminal installed on a bus or a subway to perform program broadcasting. For example, the mobile TV terminal on the bus can be adjusted according to the volume adjustment method provided by the embodiment of the present invention. The number of the car and the size of the sound in the car are automatically adjusted.
  • a method for adjusting a volume of a user terminal includes the following steps:
  • the time for collecting the sound signal may be controlled according to the processing capability of the device that performs the volume adjustment. For example, when the processing capability of the device is strong, the sound signal of the current environment may be collected when the call signal is detected to arrive. And performing the following process; when the processing capability of the device is weak, the sound signal of the current environment can be periodically collected, and the subsequent process is performed, and the adjusted volume is directly used for ringing when the call signal is accessed;
  • the sound signal of the current environment may be collected by a component such as a microphone of the terminal, or a separate sound sensor may be set in the terminal to collect a sound signal of the current environment, which is not limited herein;
  • composition information includes: a sound type included in the sound signal, and a specific gravity of each type of sound; wherein the sound type includes: blank sound, human Sound and noise;
  • the sound signal can be divided into blank sound, human sound and non-human sound (noise), etc.
  • the blank sound refers to a sound signal that cannot be recognized by the human ear
  • the sound signal with a general sound frequency other than 20 Hz to 20000 Hz can be It is considered to be a blank sound.
  • Noise is a sound signal other than human voice that can be recognized by the human ear.
  • a sound signal with a frequency of 20 Hz to 85 Hz or less and 255 Hz to 20,000 Hz can be regarded as noise through the collected sound.
  • the signal is analyzed to calculate the proportion of the sound signal including the proportion of the blank sound, the proportion of the noise, and the proportion of the human voice for subsequent scene pattern recognition;
  • the user terminal's profile is used to indicate the environment in which the user terminal is located, such as a quiet library, a controversial conference room, a quiet bedroom, or a winding road. In the implementation, it can be used for sound signals.
  • the relationship between the magnitudes of the specific types of sounds establishes corresponding scene pattern correspondences, that is, the sound signals of different specific weights correspond to different scene modes, and each scene mode corresponds to a corresponding volume; further, when the correspondence is established, the corresponding relationship may be added.
  • the volume adjustment coefficient may be determined according to the determined scene mode, and the corresponding relationship between the preset scene mode and the volume adjustment coefficient, and the volume of the user terminal is adjusted according to the volume adjustment coefficient;
  • the volume of the user terminal when applied to a communication terminal such as a mobile phone, may specifically include: ringing (also called ring tone) volume and earpiece volume; volume adjustment coefficient
  • the method may specifically include: a ringing volume adjustment coefficient and an earpiece volume adjustment coefficient;
  • the ringing volume of the user terminal can be adjusted according to the ringing volume adjustment coefficient, and the earpiece volume of the user terminal is adjusted according to the earpiece volume adjustment coefficient.
  • Table 1 shows the ringer volume adjustment factor and the earpiece volume adjustment factor set for different numbers of people in the following scene modes; wherein two numbers separated by a slash ("/") in each scene From left to right, the ringer volume adjustment factor and the handset volume adjustment factor are respectively indicated.
  • the volume adjustment is divided into ten levels, that is, 0.1 to 1.0, and the item with "d" indicates that the coefficient value needs to be determined according to the ambient sound intensity (decibel number), and d represents the environmental volume intensity level, for example, the human ear Acceptable reasonable volume range is 20 to 120 decibels (more than 120 decibels, all calculated by 120), according to each level of 10 decibels, you can divide the ambient volume into 10 levels, that is, the value range of d is 1 , 2...10. There may be some items with d, the calculation result can not accurately correspond to the ten values of 0.1 to 1.0, then select a value that is adjacent and too large; if the calculation result is less than 0.1 or greater than 1.0, the two boundary values are selected.
  • step S102 is provided in the embodiment of the present invention, and the collected sound signal is analyzed to obtain the composition information in the sound signal, as shown in FIG. 2, which may be specifically implemented as follows:
  • S201 dividing the collected sound signal into multiple pieces of audio data; for example, dividing into n parts: S1, S2, ..., Sn;
  • Audio data is classified according to blank sound, human sound, and noise;
  • S203 Calculate the proportion of blank sound, human sound, and noise in all audio data; specifically, calculate the number of audio data of different classifications separately, and compare the total number of audio data to obtain a specific gravity;
  • the calculation of the MFCC feature may be performed on the audio data discriminated as the human voice, and then the similarity of each of the two MFCC feature matrices may be calculated, and the similar MFCC feature may be considered to be the same person's voice, and vice versa. Sound, so you can count the number of people included in the N samples.
  • the embodiments of the present invention are mainly based on frequency and spectrum analysis to determine the composition information of the collected sound signals, and other similar frequency/spectral analysis methods can achieve the purpose, which are not enumerated here.
  • each piece of audio data is classified according to the calculated sound frequency according to blank sound, human sound and noise, as shown in FIG. 3, which can be realized by the following process:
  • step S301 determining whether the sound frequency of each piece of audio data is within the range of 20Hz-20000Hz, if yes, proceeding to step S302, and if not, executing step S306;
  • S302 Calculating a basic frequency of the audio data; when the sounding body emits a sound due to the vibration, the sound can be generally decomposed into a plurality of simple sine waves, that is, all the natural sounds are basically composed of a plurality of sine waves having different frequencies, The sine wave with the lowest frequency is the fundamental frequency, and the fundamental frequency can be used to distinguish different sounding bodies;
  • the audio data is considered to be a human voice
  • the audio data is considered to be noise
  • the audio data is considered to be a blank sound.
  • step S104 after adjusting the volume of the user terminal according to the determined scenario mode, after the user connects the call signal, during the call, if there is no interference (in a quiet environment)
  • the voice intensity of the speech is generally fixed; once the speaker thinks that the surrounding environment is noisy, the voice intensity of his speech will increase subconsciously; or, once the speaker thinks the current scene is very quiet (such as many people originally)
  • I don’t want the voice of the call to interfere with others; or, when it comes to private information and I don’t want others to hear the content of the call, the voice will be louder than normal. There has been a decline.
  • the volume adjustment method provided by the embodiment of the present invention provides a scheme for fine-tuning the secondary playback volume after detecting that the call signal is turned on, so as to achieve the effect of adjusting the volume of the handset to conform to the current scene.
  • the method further includes the following steps:
  • step S402. Compare the acquired microphone volume with the pre-stored reference volume. When the acquired microphone volume is greater than the pre-stored reference volume, step S403 is performed; when the acquired microphone volume is less than the pre-stored reference volume, the execution is performed. Step S404; when the acquired microphone volume is equal to the pre-stored reference volume, the process is exited;
  • the pre-stored reference volume when comparing the acquired microphone volume with the pre-stored reference volume in S402, the pre-stored reference volume may be set to a single value, or may be set to a range of values, as long as the acquired microphone volume is in the Within the range of values, the volume of the acquired microphone is considered to be equal to the reference volume, and it is not necessary to adjust the playback volume of the handset.
  • the corresponding current scene is closer to the real scene, which can be more accurate.
  • the ground matches the situation of the user, which greatly reduces the situation that the playback volume adjustment caused by the misjudgment scene does not conform to the scene, which improves the user experience.
  • the present invention further provides a volume adjustment device for a user terminal.
  • the method includes:
  • the collecting unit 601 is configured to collect a sound signal around the user terminal
  • the analyzing unit 602 is configured to analyze the collected sound signal to obtain composition information of the sound signal; the composition information includes: a sound type included in the sound signal, and a specific gravity of each type of sound; wherein the sound type includes: a blank sound Human voice and noise;
  • the scene mode determining unit 603 is configured to determine a current scene mode of the user terminal according to the composition information of the sound signal;
  • the volume adjustment unit 604 is configured to adjust the volume of the user terminal according to the determined scene mode.
  • the collecting unit 601 is specifically configured to: when detecting that a call signal arrives, collect a sound signal of the current environment; or periodically collect the sound signal of the current environment. .
  • the volume adjustment unit 604 is specifically configured to: according to the determined scene mode, and the corresponding correspondence between the preset scene mode and the volume adjustment coefficient The relationship determines the volume adjustment factor and adjusts the volume of the user terminal according to the volume adjustment factor.
  • the volume of the user terminal includes a ringing volume and an earpiece volume
  • the volume adjustment coefficient in the volume adjusting unit 604 specifically includes: a ringing volume adjustment coefficient and an earpiece volume adjustment coefficient
  • the volume adjustment unit 604 is specifically configured to adjust the ringing volume of the user terminal according to the ringing volume adjustment coefficient, and adjust the earpiece volume of the user terminal according to the earpiece volume adjustment coefficient.
  • the method further includes:
  • the obtaining unit 605 is configured to acquire the microphone volume of the user terminal in real time after detecting that the call signal is turned on;
  • the comparing unit 606 is configured to compare the acquired microphone volume with a pre-stored reference volume
  • the volume adjustment unit 604 is further configured to: when the acquired microphone volume is greater than the preset reference volume, increase the handset volume of the user terminal; when the acquired microphone volume is less than the pre-stored reference volume, reduce the handset volume of the user terminal. .
  • the collecting unit 601 is specifically configured to periodically collect the sound signal around the user terminal after detecting that the call signal is turned on;
  • the analyzing unit 602 is further configured to analyze the sound signal periodically collected by the collecting module 601 after detecting that the call signal is turned on, to obtain the composition information of the sound signal; and determine whether the sound signal includes the human according to the composition information of the sound signal.
  • the sound contains only one person's voice; if so, the volume of the sound signal is calculated, and the volume of the calculated sound signal is averaged with the previously stored reference volume, and then stored as a new reference volume.
  • the analyzing unit 602 specifically includes:
  • a first processing unit configured to divide the collected sound signal into multiple pieces of audio data
  • a second processing unit configured to calculate a sound frequency for each piece of audio data, and classify according to the calculated audio data according to blank sound, human sound, and noise;
  • a third processing unit for counting the proportion of blank sound, human sound and noise in all audio data
  • a fourth processing unit configured to calculate a Mel frequency cepstrum coefficient for the audio data determined to be a human voice, and perform audio statistics of the same frequency of the Mel frequency cepstral coefficient as a person's voice to determine the human voice The number of people included.
  • the second processing unit is specifically configured to determine whether the sound frequency of each piece of audio data is within a range of 20 Hz to 20,000 Hz; and determine the sound frequency of the audio data at 20 Hz-
  • the fundamental frequency of the audio data is calculated, and when it is judged that the fundamental frequency is within the range of 85 Hz - 255 Hz, the audio data is considered to be a human voice; when it is judged that the fundamental frequency is outside the range of 85 Hz - 255 Hz, it is considered The audio data is noise; when it is judged that the sound frequency of the audio data is outside the range of 20 Hz to 20,000 Hz, the audio data is considered to be a blank sound.
  • the above-mentioned volume adjustment device when determining the current scene mode of the user terminal, can increase the reference factor of the constituent information in the sound signal, and the corresponding current scene is closer to the real scene, which can be more accurately Matching the situation in which the user is located, greatly reducing the situation that the playback volume adjustment caused by the misjudgment scene does not conform to the scene increases the user experience.
  • an embodiment of the present invention further provides a terminal, including: a speaker, an earpiece, and the above-mentioned volume adjustment device provided by the embodiment of the present invention, wherein the volume adjustment device is configured to adjust the volume of the speaker and the earpiece;
  • the terminal may be any product or component having a playing function, such as a mobile phone, a walkie-talkie, a tablet computer, a television, a display, a notebook computer, and the like.
  • a playing function such as a mobile phone, a walkie-talkie, a tablet computer, a television, a display, a notebook computer, and the like.
  • An embodiment of the present invention provides another terminal, as shown in FIG. 7, including:
  • the sound sensor 150 is configured to collect sound signals around the user terminal 100; the speaker 130 is configured to emit a ring tone when the user terminal 100 has a call signal access; it can be understood that the speaker 130 can also be used. Playing audio data such as music;
  • the handset 170 is configured to play the voice of the other party when the user talks to the other party through the user terminal 100.
  • the terminal 100 may further include a display unit 140 that can be used to display information input by the user or information provided to the user and various menu interfaces of the terminal 100.
  • the display unit 140 can include a display panel 141.
  • the display panel 141 can be an LCD (Liquid Crystal). Display, liquid crystal display) or OLED (Organic Light-Emitting Diode).
  • the memory 120 stores executable modules or data structures, or a subset thereof, or their extended set:
  • the processor 160 is configured to: analyze the sound signal collected by the sound sensor 150 to obtain composition information of the sound signal;
  • the information includes: a sound type included in the sound signal, and a specific gravity of each type of sound; wherein the sound type includes: a blank sound, a human sound, and a noise sound; determining the user terminal according to the composition information of the sound signal The current scene mode; adjusting the volume of the speaker 130 and/or the earpiece 170 based on the determined scene mode.
  • the sound sensor 150 acquires the volume of the microphone 110 in real time
  • the processor 160 is further configured to:
  • the handset volume of the user terminal When the acquired microphone volume is greater than the pre-stored reference volume, the handset volume of the user terminal is increased; when the acquired microphone volume is less than the pre-stored reference volume, the handset volume of the user terminal is decreased.
  • the sound sensor 150 is a unit for collecting sound signals, and may be integrated into the microphone 110 or a separate component, which is not particularly limited in the present invention.
  • the terminal device 100 can also perform the method and the embodiment of FIG. 1 to FIG. 5 , and details are not described herein again.
  • the user terminal when determining the current scene mode of the user terminal, increases the reference factor for the information in the sound signal, and the corresponding current scene is closer to the real scene, which can be more accurately
  • the situation in which the user is located is matched, which greatly reduces the situation that the playback volume adjustment caused by the misjudgment scene does not conform to the scene, which improves the user experience.
  • the embodiments of the present invention may be implemented by hardware, or may be implemented by means of software plus a necessary general hardware platform.
  • the technical solution of the embodiment of the present invention may be in the form of a software product. It is manifested that the software product can be stored in a non-volatile storage medium (which can be a CD-ROM, a USB flash drive, a mobile hard disk, etc.), and includes a plurality of instructions for making a computer device (which can be a personal computer, a server, Or a network device or the like) performs the methods described in various embodiments of the present invention.
  • a non-volatile storage medium which can be a CD-ROM, a USB flash drive, a mobile hard disk, etc.
  • a computer device which can be a personal computer, a server, Or a network device or the like
  • modules in the apparatus in the embodiments may be distributed in the apparatus of the embodiment according to the description of the embodiments, or the corresponding changes may be located in one or more apparatuses different from the embodiment.
  • the modules of the above embodiments may be combined into one module, or may be further split into multiple sub-modules.

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Abstract

本发明公开了一种音量调节方法、装置及终端，在釆集用户终端周围的声音信号之后，对釆集到的声音信号进行分析，以得到声音信号的构成信息；该构成信息包括：声音信号包含的声音类型、以及各类型声音的比重，其中声音类型包括：空白声音、人类声音以及噪声；根据声音信号的构成信息确定用户终端当前的情景模式；并根据确定出的情景模式调节用户终端的音量。在确定用户终端当前的情景模式时，由于增加了对于声音信号中构成信息的参考因素，对应出的当前场景会更加贴近真实的场景，可以更加精准地与用户所处的情境匹配，大大减少由误判场景带来的播放音量调节不符合场景的情况发生提升了用户体验。

Description

一种用户终端的音量调节方法、装置及终端 技术领域
本发明涉及通讯技术领域，尤其涉及一种用户终端的音量调节方法、装置及终端。
背景技术
随着通信技术的不断发展，便携式用户终端，例如手机、平板电脑，已经成为人们工作、生活中必不可少的随身物品。人们可能会随时随地拨打或者接听电话，在接听电话时，往往需要根据不同的场合设置不同的响铃音量。对于手机用户，通常希望在办公室等安静环境中采用较小音量的响铃，以免产生较大的声音影响到其他办公人员的正常工作；而在商场、车站等公共嘈杂场合则需要采用较大音量的响铃，从而保证可以及时接听接入的呼叫。
目前，多数手机的响铃的音量和听筒的通话音量通常由用户手动调节，或者集成了简单的可以自感知的情景模式如室内模式、户外模式等特定场景。
在利用自感知的情景模式对手机的响铃的播放音量进行调节时，在有呼叫进入时，一般是通过声音检测模块提取并判断环境声音的分贝数，然后与预先存储的声音的分贝数与来电响铃音量的对应关系，调节铃声和听筒的音量。上述这种音量的调节方式仅根据环境声音的分贝数决定最后的响铃和听筒音量，但是环境声音的分贝数并不能准确反映用户所处的情境，提取的环境声音中若有较高的人声也会被误认为是噪声，比如讨论比较激烈的会议或者宣讲会、发布会等，在这些情境下本需要调小音量，但是根据环境声音的分贝数确定出的音量相对较大，导致手机音量的调节不合真实场景，准确性不高。
发明内容
本发明实施例提供了一种用户终端的音量调节方法、装置及终端，以使 手机的音量自适应，并且更加精准地与用户所处的情境匹配，提升用户体验。
第一方面，提供一种用户终端的音量调节方法，包括：
采集所述用户终端周围的声音信号；
对采集到的所述声音信号进行分析，以得到所述声音信号的构成信息；所述构成信息包括：所述声音信号包含的声音类型、以及各种类型声音的比重；其中，所述声音类型包括：空白声音、人类声音以及噪声；
根据所述声音信号的构成信息确定所述用户终端当前的情景模式；
根据确定出的情景模式调节所述用户终端的音量。
结合第一方面，在第一种可能的实现方式中，所述采集所述用户终端周围的声音信号，具体包括：
在检测到有呼叫信号到达时，采集当前所在环境的声音信号；或，
周期性地采集当前所在环境的声音信号。
结合第一方面或第一方面的第一种可能的实现方式，在第二种可能的实现方式中，所述根据确定出的情景模式调节所述用户终端的音量，包括：
根据确定出的情景模式，以及预先存储的情景模式与音量调节系数的对应关系，确定音量调节系数，并根据所述音量调节系数调整所述用户终端的音量。
结合第一方面的第二种可能的实现方式，在第三种可能的实现方式中，所述用户终端的音量包括：响铃音量和听筒音量；所述音量调节系数包括：响铃音量调节系数和听筒音量调节系数；
所述根据所述音量调节系数调节所述用户终端的音量，包括：
根据所述响铃音量调节系数调节所述用户终端的响铃音量，以及根据所述听筒音量调节系数调节所述用户终端的听筒音量。
结合第一方面、第一方面的第一种可能的实现方式、第一方面的第二种可能的实现方式或第一方面的第三种可能的实现方式中，在第四种可能的实现方式中，在检测到呼叫信号接通后，还包括：
实时获取所述用户终端的话筒音量；
当获取到的话筒音量大于预先存储的基准音量时，增大所述用户终端的听筒音量；
当获取到的话筒音量小于预先存储的基准音量时，降低所述用户终端的听筒音量。
结合第一方面的第四种可能的实现方式，在第五种可能的实现方式中，在检测到呼叫信号接通后，还包括：
周期性地采集所述用户终端周围的声音信号，对采集到的所述声音信号进行分析，以得到所述声音信号的构成信息；
根据所述声音信号的构成信息判断所述声音信号是否包含人类声音且仅包含一个人的声音；若是，则计算所述声音信号的音量，将计算出的所述声音信号的音量与预先存储的所述基准音量取平均值后，作为新的基准音量进行存储。
结合第一方面、第一方面的第一种可能的实现方式、第一方面的第二种可能的实现方式、第一方面的第三种可能的实现方式、第一方面的第四种可能的实现方式或第一方面的第五种可能的实现方式，在第六种可能的实现方式中，所述对采集到的所述声音信号进行分析，以得到所述声音信号的构成信息，具体包括：
将采集到的所述声音信号划分为多份的音频数据；
计算各份音频数据的声音频率，并根据计算出的声音频率将各份音频数据按照空白声音、人类声音和噪声进行分类；
统计所有音频数据中空白声音、人类声音和噪声的比重；
针对被判别为人类声音的音频数据，计算人类声音的音频数据的梅尔频率倒谱系数，将梅尔频率倒谱系数相同的音频数据作为一个人的声音进行统计，以判断出人类声音中所包含的人数信息。
结合第一方面的第六种可能的实现方式，在第七种可能的实现方式中，所述根据计算出的声音频率将各份音频数据按照空白声音、人类声音和噪声进行分类，具体包括：
判断各份音频数据的声音频率是否在20Hz-20000Hz范围之内；
在判断所述音频数据的声音频率在20Hz-20000Hz范围之内时，计算所述音频数据的基础频率，并在判断所述基础频率在85Hz-255Hz范围之内时，则认为所述音频数据为人类声音；在判断所述基础频率在85Hz-255Hz范围之外时，则认为所述音频数据为噪声；
在判断所述音频数据的声音频率在20Hz-20000Hz范围之外时，则认为所述音频数据为空白声音。
第二方面，提供一种用户终端的音量调节装置，包括：
采集单元，用于采集所述用户终端周围的声音信号；
分析单元，用于对采集到的所述声音信号进行分析，以得到所述声音信号的构成信息；所述构成信息包括：所述声音信号包含的声音类型、以及各类型声音的比重；其中，所述声音类型包括：空白声音、人类声音以及噪声；
情景模式确定单元，用于根据所述声音信号的构成信息确定所述用户终端当前的情景模式；
音量调节单元，用于根据确定出的情景模式调节用户终端的音量。
结合第二方面，在第一种可能的实现方式中，所述采集单元，具体用于在检测到有呼叫信号到达时，采集当前所在环境的声音信号；或，周期性地采集当前所在环境的声音信号。
结合第二方面或第二方面的第一种可能的实现方式，在第二种可能的实现方式中，所述音量调节单元，具体用于根据确定出的情景模式，以及预先存储的情景模式与音量调节系数的对应关系，确定音量调节系数，并根据所述音量调节系数调整所述用户终端的音量。
结合第二方面的第二种可能的实现方式，在第三种可能的实现方式中，所述用户终端的音量包括：响铃音量和听筒音量；所述音量调节单元中的音量调节系数具体包括：响铃音量调节系数和听筒音量调节系数；
所述音量调节单元，具体用于根据所述响铃音量调节系数调节所述用户终端的响铃音量，以及根据所述听筒音量调节系数调节所述用户终端的听筒 音量。
结合第二方面、第二方面的第一种可能的实现方式、第二方面的第二种可能的实现方式或第二方面的第三种可能的实现方式，在第四种可能的实现方式中，还包括：
获取单元，用于在检测到呼叫信号接通后，实时获取所述用户终端的话筒音量；
比较单元，用于比较获取到的话筒音量与预先存储的基准音量的大小；
所述音量调节单元，还用于当获取到的话筒音量大于预先存储的基准音量时，增大所述用户终端的听筒音量；当获取到的话筒音量小于预先存储的基准音量时，降低所述用户终端的听筒音量。
结合第二方面的第四种可能的实现方式，在第五种可能的实现方式中，所述采集单元，具体用于在检测到呼叫信号接通后，周期性地采集所述用户终端周围的声音信号；
所述分析单元，还用于对所述采集模块在检测到呼叫信号接通后周期性地采集到的声音信号进行分析，以得到所述声音信号的构成信息；根据所述声音信号的构成信息判断所述声音信号是否包含人类声音且仅包含一个人的声音；若是，则计算所述声音信号的音量，将计算出的所述声音信号的音量与预先存储的所述基准音量取平均值后，作为新的基准音量进行存储。
结合第二方面、第二方面的第一种可能的实现方式、第二方面的第二种可能的实现方式、第二方面的第三种可能的实现方式、第二方面的第四种可能的实现方式或第二方面的第五种可能的实现方式，在第六种可能的实现方式中，所述分析单元，具体包括：
第一处理单元，用于将采集到的所述声音信号划分为多份的音频数据；
第二处理单元，用于计算各份音频数据的声音频率，并根据计算出的声音频率将各份音频数据按照空白声音、人类声音和噪声进行分类；
第三处理单元，用于统计所有音频数据中空白声音、人类声音和噪声的比重；
第四处理单元，用于针对被判别为人类声音的音频数据，计算梅尔频率倒谱系数，将梅尔频率倒谱系数相同的音频数据作为一个人的声音进行统计，以判断人类声音中所包含的人数信息。
结合第二方面的第六种可能的实现方式，在第七种可能的实现方式中，所述第二处理单元，具体用于判断各份音频数据的声音频率是否在20Hz-20000Hz范围之内；在判断所述音频数据的声音频率在20Hz-20000Hz范围之内时，计算所述音频数据的基础频率，并在判断所述基础频率在85Hz-255Hz范围之内时，则认为所述音频数据为人类声音；在判断所述基础频率在85Hz-255Hz范围之外时，则认为所述音频数据为噪声；在判断所述音频数据的声音频率在20Hz-20000Hz范围之外时，则认为所述音频数据为空白声音。
第三方面，提供一种终端，包括：扬声器和听筒；还包括本发明实施例提供的上述音量调节装置。
本发明实施例提供的用户终端的音量调节方法、装置及终端，具有以下有益效果：
在确定用户终端当前的情景模式时，由于增加了对于声音信号中构成信息的参考因素，对应出的当前场景会更加贴近真实的场景，可以更加精准地与用户所处的情境匹配，大大减少由误判场景带来的播放音量调节不符合场景的情况发生提升了用户体验。
附图说明
图1为本发明实施例提供的音量调节方法的流程示意图之一；
图2为本发明实例例提供的音量调节方法的流程示意图之二；
图3为本发明实例例提供的音量调节方法的流程示意图之三；
图4为本发明实例例提供的音量调节方法的流程示意图之四；
图5为本发明实例例提供的音量调节方法的流程示意图之五；
图6为本发明实例例提供的音量调节装置的结构示意图；
图7为本发明实施例提供的用户终端的结构示意图。
具体实施方式
为了解决现有的用户终端根据周围环境自动调节音量时对场景的识别精确程度不高导致调节出的音量与真实场景不匹配的问题，本发明实施例提供了一种用户终端的音量调节方法、装置及终端，能够对环境声音进行精确分析，匹配出的当前场景更贴近真实场景，进而调节出合适的音量，可以大大减少由误判场景带来的音量调节不符合场景的情况发生。下面将结合本发明实施例中的附图，对本发明实施例中的技术方案进行清楚、完整地描述，显然，所描述的实施例仅仅是本发明一部分实施例，而不是全部的实施例。基于本发明中的实施例，本领域普通技术人员在没有创造性劳动前提下所提出的其他实施例都属于本发明所保护的范围。
本发明实施例提供的用户终端的音量调节方法可以主要应用于例如手机或对讲机等通信终端进行通信，例如手机在有呼叫信号接入时，可以暂不响铃，通过本发明实施例提供的音量调节方法确定出音量后，控制合适的响铃音量，并在呼叫接通后，控制合适的听筒音量。本发明实施例提供的音量调节方法还可以应用于例如公交车或地铁上安装的移动电视终端进行节目播放，例如在公交车上的移动电视终端可以通过本发明实施例提供的音量调节方法根据乘客的数目以及车厢内声音的大小进行节目音量的自动调节。
参见图1所示，本发明实施例提供的一种用户终端的音量调节方法，该方法包括以下步骤：
S101、采集用户终端周围的声音信号；
在具体实施时，可以根据执行音量调节的设备的处理能力，控制采集声音信号的时间，例如在设备处理能力较强时，可以在检测到有呼叫信号到达时，才采集当前所在环境的声音信号，并进行后续流程；在设备处理能力较弱时，可以周期性地采集当前所在环境的声音信号，并进行后续流程，在呼叫信号接入时直接采用调节好的音量进行响铃播放；
在具体实施时，可以通过终端的麦克等部件采集当前环境的声音信号，也可以在终端中设置单独的声音传感器采集当前环境的声音信号，在此不做限定；
S102、对采集到的声音信号进行分析，以得到声音信号的构成信息；该构成信息包括：声音信号包含的声音类型、以及各种类型声音的比重；其中，该声音类型包括：空白声音、人类声音以及噪声；
在具体实施时，声音信号可以分为空白声音、人类声音以及非人类声音(噪声)等，其中，空白声音是指人耳无法识别的声音信号，一般声音频率在20Hz到20000Hz以外的声音信号可以被认为是空白声音，噪声是指可以被人耳识别的除人类声音的声音信号，一般声音频率在20Hz到85Hz以内以及255Hz到20000Hz以内的声音信号可以被认为是噪声，通过对采集到的声音信号进行分析，计算出声音信号中包含空白声音的比重、噪声的比重和人类声音的比重，以便进行后续情景模式识别；
S103、根据声音信号的构成信息确定用户终端当前的情景模式；
用户终端的情景模式用于表示该用户终端所处的环境情况，比如安静的图书馆、充满争论的会议室、安静的卧室，或者喧嚣的马路上，在具体实施时，可以针对声音信号中三种类型声音的比重的大小关系建立对应的情景模式对应关系，即不同比重的声音信号对应于不同的情景模式，每种情景模式对应有相应的音量；进一步地，在建立对应关系时还可以增加了人类声音的数量的参考因素，这样对应出的情景模式会更加贴近真实的环境，可以大大减少由误判环境带来的音量调节不符的情况发生；
S104、根据确定出的情景模式调节用户终端的音量。
在具体实施时，可以根据确定出的情景模式，以及预先存储的情景模式与音量调节系数的对应关系，确定音量调节系数，并根据音量调节系数调整用户终端的音量；
在具体实施时，当应用于例如手机的通信终端时，用户终端的音量可以具体包括：响铃(也叫电话铃音，ring tone)音量和听筒音量；音量调节系数 可以具体包括：响铃音量调节系数和听筒音量调节系数两种；
对应的，可以根据响铃音量调节系数调节用户终端的响铃音量，根据听筒音量调节系数用户终端的听筒音量。
表1示出了在下述几种情景模式中，针对人数不同而设定的响铃音量调节系数和听筒音量调节系数；其中，每个场景中用斜线(“/”)分开的两个数字从左到右分别表示响铃音量调节系数和听筒音量调节系数。
假设音量调节分成十个等级，即0.1到1.0，带有“d”的项表示该系数值还需要依据环境声音强度(分贝数)来确定，d表示环境音量强度等级，举例具体为：人耳可以接受的合理音量范围是20到120分贝(超过120分贝的，都以120来计算)，按照每10分贝提升一个等级，可以把环境音量也分成10个等级，即d的取值范围是1，2……10。可能有些带有d的项，计算结果无法精确对应到0.1到1.0这十个数值，则选取临近并且偏大的一个数值；如果计算结果小于0.1或者大于1.0，则选取这两个边界值。
表1
Figure PCTCN2015072906-appb-000001
在具体实施时，本发明实施例提供的上述步骤S102、对采集到的声音信号进行分析，以得到声音信号中的构成信息，如图2所示，具体可以通过如下方式实现：
S201、将采集到的声音信号划分为多份的音频数据；例如分成n份：S1、S2……Sn；
S202、计算各份音频数据的声音频率，并根据计算出的声音频率将各份 音频数据按照空白声音、人类声音和噪声进行分类；
S203、统计所有音频数据中空白声音、人类声音和噪声的比重；具体地，分别计算出不同分类的音频数据份数，与总的音频数据份数进行比较，得到比重；
S204、针对被判别为人类声音的音频数据，计算梅尔频率倒谱系数(MFCC，Mel-Frequency Cepstral Coefficients)，然后将梅尔频率倒谱系数相同的音频数据作为一个人的声音进行统计，以判断出人类声音中所包含的人数信息；
具体地，可以针对被判别为人类声音的音频数据进行MFCC特征的计算，之后计算每两个MFCC特征矩阵的相似度，结果相似的MFCC特征可以认为是同一个人的声音，反之则是不同人的声音，这样可以统计出N份采样中包含的人数信息。
可以理解的是，本发明实施例主要基于频率和频谱分析，以确定采集到的声音信号的构成信息，其它类似的频率/频谱分析方法均能实现该目的，在此不再一一列举。
具体地，在上述步骤S202根据计算出的声音频率将各份音频数据按照空白声音、人类声音和噪声进行分类，如图3所示，可以通过下述流程实现：
S301、判断各份音频数据的声音频率是否在20Hz-20000Hz范围之内，若是，则执行步骤S302，若否，则执行步骤S306；
S302、计算音频数据的基础频率；当发声体由于震动而发出声音时，声音一般可以分解为许多单纯的正弦波，也就是说所有的自然声音基本都是由许多频率不同的正弦波组成的，其中频率最低的正弦波即为基础频率(fundamental frequency)，采用基础频率可以用来区分不同的发声体；
S303、判断基础频率是否在85Hz-255Hz范围之内，若是，则执行步骤S304；若否，则执行步骤S305；
S304、认为音频数据为人类声音；
S305、认为音频数据为噪声；
S306、认为音频数据为空白声音。
进一步地，在执行本发明实施例提供的上述步骤S104根据确定出的情景模式调节用户终端的音量后，用户接通呼叫信号后，在通话进行中，若在没有干扰的情况下(安静环境下)通话时，讲话的声音强度一般是固定的；一旦讲话者认为周围的环境比较吵，他讲话的声音强度就会下意识地提高；或，一旦讲话者认为当前的情景非常安静(比如原本很多人在讲话，有一个人接听电话后，其他人都不说话了)，不希望通话声音干扰到别人；或，涉及到隐私信息而不希望别人听到通话的内容，讲话的声音强度会比正常情况下有所下降。针对上述这些情况，本发明实施例提供的音量调节方法在检测到呼叫信号接通后，还提供了二次播放音量微调的方案，以达到调节听筒音量使其符合当前情景的效果。
基于此，在本发明实施例提供的上述音量调节方法中，如图4所示，还包括如下步骤：
S401、在检测到呼叫信号接通后，实时获取用户终端的话筒音量；
S402、比较获取到的话筒音量与预先存储的基准音量的大小，当获取到的话筒音量大于预先存储的基准音量时，执行步骤S403；当获取到的话筒音量小于预先存储的基准音量时，执行步骤S404；当获取到的话筒音量等于预先存储的基准音量时，退出流程；
S403、增大用户终端的听筒音量；
S404、降低用户终端的听筒音量。
在具体实施时，S402中比较获取到的话筒音量与预先存储的基准音量的大小时，可以将预先存储的基准音量设置为单一数值，也可以设置为一数值范围，只要获取的话筒音量在该数值范围内，都可认为获取到的话筒音量与基准音量相等，不必调整听筒的播放音量。
上述步骤S401～S404的执行是基于预先存储的基础音量实现的，该基础音量一般为固定值，是在本次通话之前的通话过程中确定并存储的，当然，基础音量也可以进行更新，如图5所示，可以通过如下步骤实现：
S501、在检测到呼叫信号接通后，周期性地采集用户终端周围的声音信号；
S502、对采集到的声音信号进行分析，以得到声音信号的构成信息；在具体实施时，具体执行步骤可以参见步骤S201～S204；
S503、根据声音信号的构成信息判断声音信号是否包含人类声音且仅包含一个人的声音；若是，则执行步骤S504；若否，则执行二次播放音量微调的方案，即执行步骤S401～S404；
S504、计算声音信号的音量，将计算出的声音信号的音量与预先存储的基准音量取平均值后，作为新的基准音量进行存储。
本发明实施例提供的上述音量调节方法中，在确定用户终端当前的情景模式时，由于增加了对于声音信号中构成信息的参考因素，对应出的当前场景会更加贴近真实的场景，可以更加精准地与用户所处的情境匹配，大大减少由误判场景带来的播放音量调节不符合场景的情况发生提升了用户体验。
基于同一发明构思，本发明还提供了一种用户终端的音量调节装置，参见图6所示，包括：
采集单元601，用于采集用户终端周围的声音信号；
分析单元602，用于对采集到的声音信号进行分析，得到声音信号的构成信息；该构成信息包括：声音信号包含的声音类型、以及各类型声音的比重；其中，该声音类型包括：空白声音、人类声音以及噪声；
情景模式确定单元603，用于根据声音信号的构成信息确定用户终端当前的情景模式；
音量调节单元604，用于根据确定出的情景模式调节用户终端的音量。
具体地，在本发明实施例提供的上述装置中，采集单元601，具体用于在检测到有呼叫信号到达时，采集当前所在环境的声音信号；或，周期性地采集当前所在环境的声音信号。
具体地，在本发明实施例提供的上述装置中，音量调节单元604具体用于根据确定出的情景模式，以及预先存储的情景模式与音量调节系数的对应 关系，确定音量调节系数，并根据音量调节系数调整所述用户终端的音量。
具体地，在本发明实施例提供的上述装置中，用户终端的音量包括响铃音量和听筒音量；音量调节单元604中的音量调节系数具体包括：响铃音量调节系数和听筒音量调节系数；
音量调节单元604，具体用于根据响铃音量调节系数调节用户终端的响铃音量，以及根据听筒音量调节系数调节用户终端的听筒音量。
具体地，在本发明实施例提供的上述装置中，如图6所示，还包括：
获取单元605，用于在检测到呼叫信号接通后，实时获取用户终端的话筒音量；
比较单元606，用于比较获取到的话筒音量与预先存储的基准音量的大小；
音量调节单元604，还用于当获取到的话筒音量大于预先存储的基准音量时，增大用户终端的听筒音量；当获取到的话筒音量小于预先存储的基准音量时，降低用户终端的听筒音量。
具体地，在本发明实施例提供的上述装置中，采集单元601，具体用于在检测到呼叫信号接通后，周期性地采集用户终端周围的声音信号；
分析单元602，还用于对采集模块601在检测到呼叫信号接通后周期性地采集到的声音信号进行分析，以得到声音信号的构成信息；根据声音信号的构成信息判断声音信号是否包含人类声音且仅包含一个人的声音；若是，则计算声音信号的音量，将计算出的声音信号的音量与预先存储的基准音量取平均值后，作为新的基准音量进行存储。
具体地，在本发明实施例提供的上述装置中，分析单元602，具体包括：
第一处理单元，用于将采集到的声音信号划分为多份的音频数据；
第二处理单元，用于计算对各份音频数据的声音频率，并根据计算出的音频数据按照空白声音、人类声音和噪声进行分类；
第三处理单元，用于统计所有音频数据中空白声音、人类声音和噪声的比重；
第四处理单元，用于针对被判别为人类声音的音频数据，计算梅尔频率倒谱系数，将梅尔频率倒谱系数相同的音频数据作为一个人的声音进行统计，以判断人类声音中所包含的人数信息。
具体地，在本发明实施例提供的上述装置中，第二处理单元，具体用于对判断各份音频数据的声音频率是否在20Hz-20000Hz范围之内；在判断音频数据的声音频率在20Hz-20000Hz范围之内时，计算音频数据的基础频率，并在判断基础频率在85Hz-255Hz范围之内时，则认为音频数据为人类声音；在判断基础频率在85Hz-255Hz范围之外时，则认为音频数据为噪声；在判断音频数据的声音频率在20Hz-20000Hz范围之外时，则认为音频数据为空白声音。
本发明实施例提供的上述音量调节装置，在确定用户终端当前的情景模式时，由于增加了对于声音信号中构成信息的参考因素，对应出的当前场景会更加贴近真实的场景，可以更加精准地与用户所处的情境匹配，大大减少由误判场景带来的播放音量调节不符合场景的情况发生提升了用户体验。
基于同一发明构思，本发明实施例还提供了一种终端，包括：扬声器、听筒、以及本发明实施例提供的上述音量调节装置，其中，音量调节装置用于对扬声器和听筒的音量进行调节；具体地，该终端可以为：手机、对讲机、平板电脑、电视机、显示器、笔记本电脑等任何具有播放功能的产品或部件。该终端的实施可以参见上述播放音量的控制装置的实施例，重复之处不再赘述。本发明实施例提供另一种终端，如图7所示，包括：
声音传感器150，用于采集用户终端100周围的声音信号；扬声器130，用于在用户终端100有呼叫信号接入时，发出来电铃音(ring tone)；可以理解的是，扬声器130还可以用于播放音乐等音频数据；
听筒170，用于在用户通过用户终端100与对方通话时，播放对方的话音；
该终端100还可以包括显示单元140，该显示单元140可用于显示由用户输入的信息或提供给用户的信息以及终端100的各种菜单界面。该显示单元140可包括显示面板141，可选的，显示面板141可以为LCD(Liquid Crystal  Display，液晶显示器)或OLED(Organic Light-Emitting Diode,有机发光二极管)等。
在一些实施方式中，存储器120存储了可执行模块或者数据结构，或者他们的子集，或者他们的扩展集:
在本发明实施例中，通过调用存储器120存储的程序或指令，处理器160用于：对声音传感器150采集到的所述声音信号进行分析，以得到所述声音信号的构成信息；所述构成信息包括：所述声音信号包含的声音类型、以及各种类型声音的比重；其中，所述声音类型包括：空白声音、人类声音以及噪声声音；根据所述声音信号的构成信息确定所述用户终端当前的情景模式；根据确定出的情景模式调节扬声器130和/或听筒170的音量。
可选地，作为一个实施例，在检测到呼叫信号接通后，声音传感器150实时获取话筒110的音量；
所述处理器160还用于：
当获取到的话筒音量大于预先存储的基准音量时，增大所述用户终端的听筒音量；当获取到的话筒音量小于预先存储的基准音量时，降低所述用户终端的听筒音量。
需要说明的是，声音传感器150为用于采集声音信号的单元，它具体可以集成到话筒110中，也可以是单独部件，本发明不做特别限定。
另外，终端设备100还可执行图1至图5的方法及实施例，本发明实施例在此不再赘述。
本发明实施例提供的上述用户终端，在确定用户终端当前的情景模式时，由于增加了对于声音信号中构成信息的参考因素，对应出的当前场景会更加贴近真实的场景，可以更加精准地与用户所处的情境匹配，大大减少由误判场景带来的播放音量调节不符合场景的情况发生提升了用户体验。
通过以上的实施方式的描述，本领域的技术人员可以清楚地了解到本发明实施例可以通过硬件实现，也可以借助软件加必要的通用硬件平台的方式来实现。基于这样的理解，本发明实施例的技术方案可以以软件产品的形式 体现出来，该软件产品可以存储在一个非易失性存储介质(可以是CD-ROM，U盘，移动硬盘等)中，包括若干指令用以使得一台计算机设备(可以是个人计算机，服务器，或者网络设备等)执行本发明各个实施例所述的方法。
本领域技术人员可以理解附图只是一个优选实施例的示意图，附图中的模块或流程并不一定是实施本发明所必须的。
本领域技术人员可以理解实施例中的装置中的模块可以按照实施例描述进行分布于实施例的装置中，也可以进行相应变化位于不同于本实施例的一个或多个装置中。上述实施例的模块可以合并为一个模块，也可以进一步拆分成多个子模块。
上述本发明实施例序号仅仅为了描述，不代表实施例的优劣。
显然，本领域的技术人员可以对本发明进行各种改动和变型而不脱离本发明的精神和范围。这样，倘若本发明的这些修改和变型属于本发明权利要求及其等同技术的范围之内，则本发明也意图包含这些改动和变型在内。

Claims (17)

  1. 一种用户终端的音量调节方法，其特征在于，包括：
    采集所述用户终端周围的声音信号；
    对采集到的所述声音信号进行分析，以得到所述声音信号的构成信息；所述构成信息包括：所述声音信号包含的声音类型、以及各种类型声音的比重；其中，所述声音类型包括：空白声音、人类声音以及噪声；
    根据所述声音信号的构成信息确定所述用户终端当前的情景模式；
    根据确定出的情景模式调节所述用户终端的音量。
  2. 如权利要求1所述的方法，其特征在于，所述采集所述用户终端周围的声音信号，具体包括：
    在检测到有呼叫信号到达时，采集当前所在环境的声音信号；或，
    周期性地采集当前所在环境的声音信号。
  3. 根据权利要求1或2所述的方法，其特征在于，所述根据确定出的情景模式调节所述用户终端的音量，包括：
    根据确定出的情景模式，以及预先存储的情景模式与音量调节系数的对应关系，确定音量调节系数，并根据所述音量调节系数调整所述用户终端的音量。
  4. 如权利要求3所述的方法，其特征在于，所述用户终端的音量包括：响铃音量和听筒音量；所述音量调节系数包括：响铃音量调节系数和听筒音量调节系数；
    所述根据所述音量调节系数调节所述用户终端的音量，包括：
    根据所述响铃音量调节系数调节所述用户终端的响铃音量，以及根据所述听筒音量调节系数调节所述用户终端的听筒音量。
  5. 如权利要求1-4任一项所述的方法，其特征在于，在检测到呼叫信号接通后，还包括：
    实时获取所述用户终端的话筒音量；
    当获取到的话筒音量大于预先存储的基准音量时，增大所述用户终端的听筒音量；
    当获取到的话筒音量小于预先存储的基准音量时，降低所述用户终端的听筒音量。
  6. 如权利要求5所述的方法，其特征在于，在检测到呼叫信号接通后，还包括：
    周期性地采集所述用户终端周围的声音信号，对采集到的所述声音信号进行分析，以得到所述声音信号的构成信息；
    根据所述声音信号的构成信息判断所述声音信号是否包含人类声音且仅包含一个人的声音；若是，则计算所述声音信号的音量，将计算出的所述声音信号的音量与预先存储的所述基准音量取平均值后，作为新的基准音量进行存储。
  7. 如权利要求1-6任一项所述的方法，其特征在于，所述对采集到的所述声音信号进行分析，以得到所述声音信号的构成信息，具体包括：
    将采集到的所述声音信号划分为多份的音频数据；
    计算各份音频数据的声音频率，并根据计算出的声音频率将各份音频数据按照空白声音、人类声音和噪声进行分类；
    统计所有音频数据中空白声音、人类声音和噪声的比重；
    针对被判别为人类声音的音频数据，计算人类声音的音频数据的梅尔频率倒谱系数，将梅尔频率倒谱系数相同的音频数据作为一个人的声音进行统计，以判断出人类声音中所包含的人数信息。
  8. 如权利要求7所述的方法，其特征在于，所述根据计算出的声音频率将各份音频数据按照空白声音、人类声音和噪声进行分类，具体包括：
    判断各份音频数据的声音频率是否在20Hz-20000Hz范围之内；
    在判断所述音频数据的声音频率在20Hz-20000Hz范围之内时，计算所述音频数据的基础频率，并在判断所述基础频率在85Hz-255Hz范围之内时，则认为所述音频数据为人类声音；在判断所述基础频率在85Hz-255Hz范围之外 时，则认为所述音频数据为噪声；
    在判断所述音频数据的声音频率在20Hz-20000Hz范围之外时，则认为所述音频数据为空白声音。
  9. 一种用户终端的音量调节装置，其特征在于，包括：
    采集单元，用于采集所述用户终端周围的声音信号；
    分析单元，用于对采集到的所述声音信号进行分析，以得到所述声音信号的构成信息；所述构成信息包括：所述声音信号包含的声音类型、以及各类型声音的比重；其中，所述声音类型包括：空白声音、人类声音以及噪声；
    情景模式确定单元，用于根据所述声音信号的构成信息确定所述用户终端当前的情景模式；
    音量调节单元，用于根据确定出的情景模式调节用户终端的音量。
  10. 如权利要求9所述的装置，其特征在于，所述采集单元，具体用于在检测到有呼叫信号到达时，采集当前所在环境的声音信号；或，周期性地采集当前所在环境的声音信号。
  11. 如权利要求9或10所述的装置，其特征在于，所述音量调节单元，具体用于根据确定出的情景模式，以及预先存储的情景模式与音量调节系数的对应关系，确定音量调节系数，并根据所述音量调节系数调整所述用户终端的音量。
  12. 如权利要求11所述的装置，其特征在于，所述用户终端的音量包括：响铃音量和听筒音量；所述音量调节单元中的音量调节系数具体包括：响铃音量调节系数和听筒音量调节系数；
    所述音量调节单元，具体用于根据所述响铃音量调节系数调节所述用户终端的响铃音量，以及根据所述听筒音量调节系数调节所述用户终端的听筒音量。
  13. 如权利要求9-12任一项所述的装置，其特征在于，还包括：
    获取单元，用于在检测到呼叫信号接通后，实时获取所述用户终端的话筒音量；
    比较单元，用于比较获取到的话筒音量与预先存储的基准音量的大小；
    所述音量调节单元，还用于当获取到的话筒音量大于预先存储的基准音量时，增大所述用户终端的听筒音量；当获取到的话筒音量小于预先存储的基准音量时，降低所述用户终端的听筒音量。
  14. 如权利要求13所述的装置，其特征在于，所述采集单元，具体用于在检测到呼叫信号接通后，周期性地采集所述用户终端周围的声音信号；
    所述分析单元，还用于对所述采集模块在检测到呼叫信号接通后周期性地采集到的声音信号进行分析，以得到所述声音信号的构成信息；根据所述声音信号的构成信息判断所述声音信号是否包含人类声音且仅包含一个人的声音；若是，则计算所述声音信号的音量，将计算出的所述声音信号的音量与预先存储的所述基准音量取平均值后，作为新的基准音量进行存储。
  15. 如权利要求9-14任一项所述的装置，其特征在于，所述分析单元，具体包括：
    第一处理单元，用于将采集到的所述声音信号划分为多份的音频数据；
    第二处理单元，用于计算各份音频数据的声音频率，并根据计算出的声音频率将各份音频数据按照空白声音、人类声音和噪声进行分类；
    第三处理单元，用于统计所有音频数据中空白声音、人类声音和噪声的比重；
    第四处理单元，用于针对被判别为人类声音的音频数据，计算梅尔频率倒谱系数，将梅尔频率倒谱系数相同的音频数据作为一个人的声音进行统计，以判断人类声音中所包含的人数信息。
  16. 如权利要求15所述的装置，其特征在于，所述第二处理单元，具体用于判断各份音频数据的声音频率是否在20Hz-20000Hz范围之内；在判断所述音频数据的声音频率在20Hz-20000Hz范围之内时，计算所述音频数据的基础频率，并在判断所述基础频率在85Hz-255Hz范围之内时，则认为所述音频数据为人类声音；在判断所述基础频率在85Hz-255Hz范围之外时，则认为所述音频数据为噪声；在判断所述音频数据的声音频率在20Hz-20000Hz范围之 外时，则认为所述音频数据为空白声音。
  17. 一种终端，包括：扬声器和听筒；其特征在于，还包括如权利要求9-16任一项所述的音量调节装置。
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