WO2010081434A1 - 自适应音量调节的方法、装置及通信终端 - Google Patents

自适应音量调节的方法、装置及通信终端 Download PDF

Info

Publication number
WO2010081434A1
WO2010081434A1 PCT/CN2010/070244 CN2010070244W WO2010081434A1 WO 2010081434 A1 WO2010081434 A1 WO 2010081434A1 CN 2010070244 W CN2010070244 W CN 2010070244W WO 2010081434 A1 WO2010081434 A1 WO 2010081434A1
Authority
WO
WIPO (PCT)
Prior art keywords
input
gain parameter
output
output gain
parameter
Prior art date
Application number
PCT/CN2010/070244
Other languages
English (en)
French (fr)
Inventor
岳中辉
Original Assignee
华为终端有限公司
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 华为终端有限公司 filed Critical 华为终端有限公司
Priority to EP10731068A priority Critical patent/EP2381738A4/en
Publication of WO2010081434A1 publication Critical patent/WO2010081434A1/zh
Priority to US13/186,166 priority patent/US20110274293A1/en

Links

Images

Classifications

    • 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
    • H04M9/00—Arrangements for interconnection not involving centralised switching
    • H04M9/08—Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic
    • H04M9/082—Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic using echo cancellers

Definitions

  • the present invention relates to the field of communications technologies, and in particular, to a method, an apparatus, and a corresponding communication terminal for adaptive volume adjustment.
  • Video conferencing In modern society, video conferencing is increasingly used as a means of multimedia communication in large-scale conferences such as business meetings and forums.
  • Video conferencing system mainly includes video terminal equipment, transmission network equipment, and multipoint control unit (Multipoint) Control Unit, MCU), business management platform and other parts, it can realize the interaction of images, voice and data between two places or multiple places.
  • the voice transmission mainly compresses and encodes the sound signal picked up by the local microphone, and transmits it to the remote conference site through the transmission network.
  • the digital signal received from the remote venue is decoded, restored to a sound signal, and finally played through a local speaker.
  • the input and output gains for adjusting the volume in the existing conference terminal are generally fixed.
  • the voice heard by the user at the other end of the conference is large and small, resulting in a local user not being able to better. Listen to the audio information transmitted by the remote venue.
  • the existence of "acoustic echo" makes people at the conference site often hear the speaker of the video terminal device playing the voice of the speaker, and can not bring a good experience to the user.
  • the inventors have found that at least the following technical problems exist in the prior art: the adjustment of the volume level on the existing video conference terminal is manually adjusted, so that it cannot well meet various requirements in modern conferences. Special scene requirements.
  • the "acoustic echo" phenomenon is not well handled, which interferes with the quality of conference communication.
  • the echo cancellation effect depends on whether the echo reference signal is predicted to be accurate, but the signal collected by the microphone on the video terminal device is composed of the feed echo and the speaker's voice, and the feed echo signal is input and output.
  • the gain has a large influence. Since the accurate input and output gain parameters cannot be obtained effectively, it is difficult to calculate the ratio of the predicted feed echo signal to the echo reference signal by the echo cancellation algorithm, and it is difficult to achieve better results.
  • An object of the present invention is to provide a method, a device, and a communication terminal for adaptive volume adjustment, so that the terminal can obtain a good listening effect without requiring the user to manually adjust the volume.
  • a method for adaptive volume adjustment including:
  • the input and output volume are adjusted according to the input and output gain parameters.
  • the embodiment of the invention further provides a method for adaptive volume adjustment, comprising:
  • the input and output volume are adjusted according to the input and output gain parameters.
  • the embodiment of the invention further provides an apparatus for adaptive volume adjustment, comprising:
  • a receiving unit configured to receive an audio signal from a sound source
  • An acquiring unit configured to acquire sound source information corresponding to the audio signal, and obtain input and output gain parameters corresponding to the sound source information
  • an adjusting unit configured to adjust an input and an output volume according to the input and output gain parameters acquired by the acquiring unit.
  • An embodiment of the present invention further provides a communication terminal, including
  • the device for adaptive volume adjustment further comprising:
  • a receiving unit configured to receive an audio signal from a sound source
  • An acquiring unit configured to acquire sound source information corresponding to the audio signal, and obtain input and output gain parameters corresponding to the sound source information
  • an adjusting unit configured to adjust an input and an output volume according to the input and output gain parameters acquired by the acquiring unit.
  • the embodiment of the invention obtains appropriate input and output gain parameters according to the acquired sound source information, and adjusts the input volume and the output volume through the input and output gain parameters, so that the user obtains a good listening effect.
  • the input and output gain parameters obtained can effectively eliminate the echo in the audio, and better ensure the clarity of the sound.
  • FIG. 1 is a schematic flow chart of a method for adaptive volume adjustment according to Embodiment 1 of the present invention
  • FIG. 2 is a schematic flow chart of a method for adaptive volume adjustment according to Embodiment 2 of the present invention.
  • FIG. 3 is a schematic flow chart of a method for adaptive volume adjustment according to Embodiment 3 of the present invention.
  • FIG. 4 is a schematic flow chart of a method for adaptive volume adjustment according to Embodiment 4 of the present invention.
  • FIG. 5 is a schematic block diagram of an adaptive volume adjustment apparatus according to Embodiment 5 of the present invention.
  • Figure 6 is a schematic block diagram of an embodiment of the acquisition unit 20 of Figure 5;
  • FIG. 7 is a schematic block diagram of another embodiment of the acquisition unit 20 of Figure 5;
  • FIG. 8 is a schematic block diagram of one embodiment of the adjustment unit 30 of FIG. 5.
  • a technology for automatically adjusting the volume is added to the communication terminal, and the volume of the input and output is automatically adjusted according to the information of the received sound source, so that the user in front of the speaker obtains a better listening effect.
  • Embodiment 1 is a schematic flow chart of a method for adaptive volume adjustment according to Embodiment 1 of the present invention.
  • the flow chart includes the following steps:
  • S101 Receive an audio signal from a sound source.
  • the sound source in the embodiment of the present invention is the source of the sound, and generally refers to the person speaking.
  • the audio signal is the sound signal from the sound source, that is, the voice signal of the speaker.
  • the sound source information corresponding to the audio signal is obtained in two ways, one is obtained by using the audio signal, and the sound source information at this time mainly refers to a volume value in the audio signal; the other is receiving The user directly inputs the predetermined sound source information, and the sound source information at this time mainly refers to the distance value between the terminal that receives the audio signal and the sound source.
  • the input and output gain levels are adjusted according to the input and output gain parameters acquired in S103. For example, after receiving the audio signal from the sound source, the input and output gain parameters corresponding to the volume value are acquired according to the volume value in the audio signal, and finally the input and output volume are adjusted according to the input and output gain parameters.
  • the input and output gain parameters corresponding to the volume value are not obtained according to the volume value in the audio signal, and the terminal receiving the audio signal is directly input by the user and is deemed to be appropriate
  • the distance value between the sound sources is obtained according to the distance value, and the input and output gain parameters corresponding to the distance value are obtained, and finally the input and output volume are adjusted according to the input and output gain parameters. In this way, the user no longer needs to manually adjust the volume of the volume, and the terminal device automatically adapts to the received volume, which is very convenient.
  • the method in this embodiment may select a method for acquiring sound source information corresponding to the audio signal according to a specific situation, and may select the best effect according to the performance of the terminal and the comprehensive factors of the collection and playback device. method.
  • the distance between the terminal of the audio signal and the sound source can also be obtained by using the audio signal, and then the input and output gain parameters corresponding to the distance value are obtained.
  • the input and output gain parameters refer to the gain parameters in the terminal device on the same side, that is, the terminal device in the same scene as the speaker can transmit the sound collected by the microphone to the input gain adjustment and then transmit the call to the call.
  • the sound transmitted from the other party to the other party can also be adjusted by the output gain and then played back by the speaker.
  • FIG. 2 is a schematic flow chart of a method for adaptive volume adjustment according to Embodiment 2 of the present invention.
  • the flow chart includes the following steps:
  • the distance between the sound source and the terminal that receives the audio signal is the distance between the speaker and the terminal input by the user, because the distance between the speaker and the terminal directly affects the distance
  • the terminal acquires the effect of the speaker's voice signal and the effect that the user listens after the remote speaker plays the sound. For example, the larger the distance value, the longer the speaker may be from the terminal, and the local terminal is less likely to acquire the collected sound.
  • the input gain parameter of the local terminal needs to be increased, so that the remote terminal is at the far end.
  • the user of the remote end can also be prevented from hearing the sound collected by the local terminal because the voice of the user collected by the local terminal is small.
  • the audio output gain parameter of the remote terminal can also be adjusted, so that the remote user can hear the local collected sound; it can be understood that the local audio input gain parameter and the remote end can also be increased simultaneously.
  • the audio output gain parameter is used to adjust the output volume.
  • the distance value can be judged according to the subjective knowledge of the user, so once the terminal receives the distance value input by the user, in the first mapping table, the corresponding input value of the distance value in the first mapping table should be acquired. , output gain parameters.
  • the first mapping table is an ideal input and output gain parameter table corresponding to different distance values of the speaker and the terminal obtained according to the volume statistics of the normal person, that is, each distance value in the table corresponds to a set of input and output gain parameters.
  • the terminal can receive the ideal input and output volume.
  • the speaker is about 2 meters away from the terminal.
  • the terminal After receiving the input information of 2 meters, the terminal searches for the first mapping table and acquires the input and output gain parameters +3db and +6db corresponding to the first mapping table of 2 meters.
  • the terminal amplifies the input volume by 3 db according to the input and output gain parameters, and amplifies the output volume by 6 db.
  • the distance value between the local terminal and the sound source can be automatically detected by adding a position detecting device to the terminal, and the mapping between the distance value and the local input gain parameter and/or the local output gain parameter is set inside the terminal.
  • the relationship is obtained by querying the predetermined first mapping table by using the distance value of the detected sound source from the terminal to obtain a corresponding local terminal input gain parameter and a remote output gain parameter.
  • the distance value can also be obtained in other ways, for example, by image processing, to obtain the distance of the sound source from the terminal, for example, using the depth camera set by the terminal to obtain the distance between the sound source and the terminal, and the distance value is obtained. Automatically transmitting to the terminal, so that the terminal obtains the input gain parameter of the local terminal according to the predetermined first mapping table with reference to the distance value, and/or the output gain parameter of the remote terminal, by using the parameter to implement the local terminal and/or Or adjust the volume of the remote terminal.
  • the advantage of the embodiment of the present invention is that the input and output gain parameters are obtained without complicated calculation process, and the effect of adjusting the input and output volume according to the parameters is also good, so that it is very easy to be popularized on terminals with limited technology.
  • the input and output gain parameters refer to the gain parameters in the terminal device on the same side, that is, the terminal device in the same scene as the speaker can transmit the sound collected by the microphone to the input gain adjustment and then transmit the call to the call.
  • the sound transmitted from the other party to the other party can also be adjusted by the output gain and then played back by the speaker.
  • FIG. 3 is a schematic flow chart of a method for adaptive volume adjustment according to Embodiment 3 of the present invention.
  • the flow chart includes the following steps:
  • S301 Receive an audio signal from a sound source.
  • the volume value is obtained according to the received sound source audio signal, and it is determined whether the volume value is within a predetermined volume range.
  • the predetermined volume range refers to a range of the optimal volume value set in advance. If the received volume value is within this range, the distance between the speaker and the terminal is considered to be ideal, so the input gain parameter corresponding to the volume value is Very suitable, no need to further adjust the volume of the input. If the received volume value is not within this range, it is considered that the distance between the speaker and the terminal is not ideal, so the input gain parameter corresponding to the volume value is not very suitable, and the input volume needs to be further adjusted.
  • the corresponding input gain parameter is obtained by calculating the difference between the volume value and the predetermined volume range parameter, thereby achieving the purpose of further adjusting the input volume. And searching for the second mapping table according to the unadjusted input gain parameter or the adjusted input gain parameter, and acquiring the corresponding output gain parameter of the input gain parameter in the second mapping table; and finally adjusting the output volume according to the output gain parameter.
  • the volume value of the received audio signal is 60 db
  • the predetermined volume range is 70 db to 75 db
  • the 60 db signal corresponds to the volume of the third file (60 db to 65 db)
  • the difference from the predetermined volume is 10 db, that is, the obtained input.
  • the gain parameter value is +10db
  • the value of the output gain parameter corresponding to the input gain parameter value in the second mapping is +10db, so the volume value in the input audio signal is increased by 10db and played by the speaker.
  • the volume value is increased by 15db.
  • the volume value in the embodiment of the present invention may also be used to analyze the spatial information of the speaker, thereby obtaining the azimuth of the speaker, controlling the camera on the terminal according to the azimuth angle, and aligning the camera with the speaker, and finally according to The focus point of the camera obtains the distance information of the speaker from the terminal. According to the distance value in the distance information, the adjustment of the input and output volume can be completed according to the method of the second embodiment.
  • the advantage of this embodiment compared with the second embodiment is that the whole method does not need to input any parameters by the user, and all of them are automatically completed by the terminal, and the input, output gain parameters can be obtained more accurately through the judgment, calculation and comparison processes, which not only simplifies the user. Operation, and more precise adjustment of input and output volume.
  • the second mapping table in this embodiment is a table for finding an output gain parameter according to the input gain parameter
  • the output gain parameter in the table is an ideal output gain parameter corresponding to the input gain parameter.
  • the input and output gain parameters refer to the gain parameters in the terminal device on the same side, that is, the terminal device in the same scene as the speaker can transmit the sound collected by the microphone to the input gain adjustment and then transmit the call to the call.
  • the sound transmitted from the other party to the other party can also be adjusted by the output gain and then played back by the speaker.
  • FIG. 4 is a schematic flow chart of a method for adaptive volume adjustment according to Embodiment 4 of the present invention.
  • the flow chart includes the following steps:
  • S401 Receive an audio signal from a sound source.
  • S402. Receive an input input gain parameter or an output gain parameter.
  • the user only needs to adjust the input gain parameter or the output gain parameter, and the terminal searches for the third mapping table, and obtains the corresponding output gain parameter of the input gain parameter in the third mapping table, or obtains
  • the output gain parameter is a corresponding input gain parameter in the third mapping table.
  • the third mapping table is obtained according to an ideal output gain parameter or an input gain parameter corresponding to the input gain parameter or the output gain parameter.
  • the advantage of this embodiment compared with the above embodiment is that the user can manually input an input gain parameter or an output gain parameter, and the terminal can automatically match the corresponding output gain parameter or input gain parameter, thereby adjusting the input and output.
  • the volume, especially when the user adjusts the output gain parameter, not only the sound that the user hears is very ideal, but also adjusts the input gain parameter, so that the input volume is well adjusted, which is very simple and practical.
  • the method in this embodiment may be implemented in the same manner as the first embodiment, the second embodiment, or the third embodiment, that is, the terminal device that receives the audio signal from the sound source can implement the method described in the first embodiment.
  • the method described in this embodiment may be implemented; or the method described in the second embodiment or the method in the embodiment may be implemented; or the method described in the third embodiment may be implemented or the embodiment may be implemented.
  • the method described. In this way, the user can selectively use an adaptive volume adjustment method according to the actual situation to achieve better results.
  • the input and output gain parameters refer to the gain parameters in the terminal device on the same side, that is, the terminal device in the same scene as the speaker can transmit the sound collected by the microphone to the input gain adjustment and then transmit the call to the call.
  • the sound transmitted from the other party to the other party can also be adjusted by the output gain and then played back by the speaker. Therefore, when both the scenes of the conversation can implement the method of the embodiment, the two users have the best effect of speaking and listening.
  • the step of filtering the audio signal may be performed:
  • An echo signal in the audio signal is filtered according to the echo processing parameter.
  • the echo processing parameters may be obtained according to the known relationship between the input and output gain parameters and the echo processing parameters in the prior art; if the echo processing parameters are obtained by the look-up table method, the pre-storage parameters may be stored according to the pre-storage
  • the correspondence table of the input and output gain parameters and the echo processing parameters finds suitable echo processing parameters.
  • the echo processing parameters can also be obtained by directly looking up the table. For example, in the second embodiment, the first mapping table is searched according to the obtained distance value, and the input, output gain parameters and echo processing parameters corresponding to the distance value are directly obtained. In the third embodiment, the input gain parameter is searched according to the obtained input gain parameter.
  • the second mapping table directly obtains the output gain parameter and the echo processing parameter corresponding to the input gain parameter; in the fourth embodiment, the third mapping table is searched according to the received input gain parameter or the output gain parameter, and the input is directly obtained.
  • the echo signals in the audio signal can be filtered according to an echo cancellation algorithm or the like.
  • the step of filtering the audio signal step effectively eliminates the echo signal in the audio signal, greatly improving the quality of the audio signal, and making the volume of the input and output very clear.
  • FIG. 5 is a schematic block diagram of an adaptive volume adjustment apparatus according to an embodiment of the present invention. As shown in FIG. 5, the apparatus includes :
  • the receiving unit 10 is mainly configured to receive an audio signal from a sound source
  • the acquiring unit 20 is configured to acquire sound source information corresponding to the audio signal, and obtain input and output gain parameters corresponding to the sound source information;
  • the adjusting unit 30 is mainly configured to adjust the input and output volume according to the input and output gain parameters acquired by the acquiring unit 20.
  • the device acquires the sound source information corresponding to the audio signal by the acquiring unit 20 according to the audio signal received by the receiving unit 10, and obtains the input and output gain parameters corresponding to the sound source information, and finally passes through the adjusting unit 30.
  • the volume of the input and output is adjusted according to the parameters acquired by the acquisition unit 20. In this way, the device adjusts the volume of the input and output according to the actually received audio signal, so that the user communicating through the device can manually obtain the better listening effect without manually adjusting the volume of the input and output according to actual conditions.
  • FIG. 6 is a schematic block diagram of one embodiment of the acquisition unit 20 of FIG. 5. As shown in FIG. 6, the obtaining unit 20 includes:
  • a picking subunit 210 configured to pick up a volume value of the audio signal
  • the parameter determining unit 220 is configured to determine a corresponding input and output gain parameter according to the volume value
  • the parameter determining unit 220 further includes:
  • the determining unit 221 is configured to determine whether the volume value is within a predetermined volume range, and generate a determination result
  • the calculating unit 222 is configured to: when the determination result is no, calculate a ratio of the volume value to a predetermined volume to obtain an input gain parameter;
  • the searching unit 223 is configured to: when the determination result is yes, find the output gain parameter corresponding to the input gain parameter in the second mapping table according to the determination result of the determining unit, or search according to the calculation result of the calculating unit 222
  • the input gain parameter in the result is the corresponding output gain parameter in the second mapping table.
  • the second mapping table is a table for finding an output gain parameter according to the input gain parameter, and is stored in the searching unit 223, and the output gain parameter in the table is an ideal output gain parameter corresponding to the input gain parameter.
  • FIG. 7 is a schematic block diagram of another embodiment of the acquisition unit 20 of FIG. 5. As shown in FIG. 7, the obtaining unit 20 includes:
  • Distance acquisition subunit 230 And a distance value corresponding to the sound source information corresponding to the audio signal; wherein the distance value is a distance value automatically obtained according to a volume value of the sound source;
  • the query sub-unit 240 is configured to obtain an input and output gain parameter corresponding to the distance value in the first mapping table; the sound source information corresponding to the audio signal is a volume value of the audio signal, and the volume is according to the prior art.
  • the value can also be used to analyze the spatial information of the speaker, thereby obtaining the azimuth of the speaker, controlling the camera on the terminal device according to the azimuth, aligning the camera with the speaker, and finally the terminal device according to the focus point of the camera
  • the distance information of the speaker from the terminal is obtained, thereby obtaining the distance value in the distance information, that is, the distance value corresponding to the volume value.
  • the first mapping table is obtained according to the sound of the normal person speaking when speaking at different distances combined with the ideal distance input and output volume statistics, and the input and output volume are also related to the performance of the related hardware device. Therefore, it is necessary to obtain the first mapping table stored in the query subunit 240 according to the relationship between the performance distance value of the different hardware devices and the input and output gains.
  • the distance obtaining sub-unit 230 is configured to obtain a distance value in the sound source information corresponding to the received audio signal, and may also be a distance value manually input by the user;
  • the query subunit 240 is configured to acquire input and output gain parameters corresponding to the manual input distance value in the first mapping table.
  • the distance obtaining sub-unit 230 notifies the query sub-unit 240 according to the first mapping table by acquiring the distance value input by the user. Input and output gain parameter query.
  • the first mapping table may be pre-set, specifically, according to the sound that is emitted when a normal person speaks at different distances, combined with the ideal distance input and output volume statistics, and input, The output volume is also related to the performance of the associated hardware device. Therefore, it is necessary to obtain the first mapping table stored in the query subunit 240 according to the relationship between the performance distance value of the different hardware devices and the input and output gains.
  • the receiving unit 10 in this embodiment may be further configured to receive an input gain parameter or an output gain parameter; the obtaining unit 20 may be further configured to obtain an input gain parameter corresponding according to the input gain parameter or the output gain parameter received by the receiving unit 10 The output gain parameter or the input gain parameter corresponding to the output gain parameter, so that the adjustment unit 30 is configured to adjust the input and output volume according to the input and output gain parameters acquired by the acquisition unit 20.
  • the obtaining unit 20 obtains an output gain parameter corresponding to the input gain parameter or an input gain parameter corresponding to the output gain parameter according to the input gain parameter or the output gain parameter received by the receiving unit 10.
  • the specific method is as follows: Obtaining an output gain parameter corresponding to the input gain parameter in the third mapping table, or acquiring an input gain parameter corresponding to the output gain parameter in the third mapping table.
  • the third mapping table is obtained according to an ideal output gain parameter or an input gain parameter corresponding to the input gain parameter or the output gain parameter, and may be stored in the acquiring unit 20.
  • the function of the fourth embodiment is added according to the foregoing solution, for example, when the volume is adjusted by inputting the gain parameter or the output gain parameter, by adjusting the peripheral button
  • the input of the input gain parameter or the output gain parameter is used to initiate the steps described in the fourth embodiment.
  • the adjusting unit 30 is mainly used to adjust the input and output volume according to the input and output gain parameters acquired by the acquiring unit. However, in order to input and output a better sound quality, the adjustment unit 30 can also filter the echo signals in the input and output sounds to obtain better input and output sound quality. Therefore, the adjusting unit 30 further includes: a processing unit 301, configured to obtain an echo processing parameter by calculating or looking up a table according to the input and output gain parameters; and an echo signal filtering unit 302, configured to use the echo processing parameter according to the echo processing parameter The echo signal in the audio signal is filtered.
  • a processing unit 301 configured to obtain an echo processing parameter by calculating or looking up a table according to the input and output gain parameters
  • an echo signal filtering unit 302 configured to use the echo processing parameter according to the echo processing parameter The echo signal in the audio signal is filtered.
  • an echo processing parameter for filtering the echo signal in the audio signal can be obtained, so that the echo signal in the audio signal can be further eliminated, and the effective restoration can be effectively performed.
  • the quality of the input and output audio signals makes the played sound clearer, overcoming the problems in the prior art.
  • the input and output gain parameters described in this embodiment all refer to the gain parameters in the terminal device on the same side, that is, the terminal device in the same scene as the speaker can perform the sound collected by the microphone. After the input gain is adjusted, it is transmitted to the other party that is talking to it. The sound transmitted by the other party that is talking to it can also be adjusted by the output gain and then played by the speaker. Therefore, when both the scenes of the conversation can use the device of the embodiment, the two users have the best effect of speaking and listening.
  • the acquiring unit in the adaptive volume adjusting device of the embodiment of the present invention can obtain information, input and output corresponding to the signals and parameters according to the audio signal, the input gain parameter or the output gain parameter received by the receiving unit. Gain parameters to adjust the input and output volume based on the input and output gain parameters.
  • the adaptive volume adjustment device of the embodiment of the invention can be adapted to various occasions, and a suitable solution is selected according to the situation of the occasions. For example, in some small conferences, the device can be used for small conferences, and the space is small. Relatively concentrated characteristics, the scheme of directly obtaining the input distance value and then automatically adjusting the input and output volume is selected; in the personal office or home scene, the device can select the speaker for the scene with less speakers and the position is generally fixed.
  • the input gain parameter or the output gain parameter is used to automatically adjust the input and output volume.
  • the device can select the volume for such a large conference space, and the speaker is not fixed.
  • the device can further eliminate the echo signal in the audio signal, greatly improving the quality of the input and output audio signals, and making the played sound clearer.
  • An embodiment of the present invention further provides a communication terminal, including
  • An apparatus for adaptive volume adjustment comprising: a receiving unit, configured to receive an audio signal from a sound source; and an acquiring unit, configured to acquire sound source information corresponding to the audio signal, and obtain the sound source
  • the input and output gain parameters corresponding to the information; the adjusting unit is mainly used to adjust the input and output volume according to the input and output gain parameters acquired by the acquiring unit.
  • the receiving unit may be further configured to receive an input gain parameter or an output gain parameter; the acquiring unit may be further configured to obtain an output gain corresponding to the input gain parameter according to the input gain parameter or the output gain parameter received by the receiving unit.
  • the input gain parameter corresponding to the parameter or output gain parameter The input gain parameter corresponding to the parameter or output gain parameter.
  • the adjusting unit may further include: a processing unit, configured to obtain an echo processing parameter by calculating or looking up a table according to the input and output gain parameters; and an echo signal filtering unit, configured to filter the audio according to the echo processing parameter The echo signal in the signal.
  • a processing unit configured to obtain an echo processing parameter by calculating or looking up a table according to the input and output gain parameters
  • an echo signal filtering unit configured to filter the audio according to the echo processing parameter The echo signal in the signal.
  • the acquiring unit includes: a first acquiring unit, configured to acquire a volume value of the audio signal; and a second acquiring unit, configured to acquire an input and output gain parameter corresponding to the volume value in a predetermined volume range; or include a third acquiring unit, configured to acquire a distance value corresponding to the sound source information corresponding to the audio signal, and a fourth acquiring unit, configured to acquire a corresponding input and output gain parameter of the distance value in the first mapping table; or include Five acquisition units, And a sixth acquisition unit, configured to acquire a corresponding input and output gain parameter of the distance value in the first mapping table.
  • the input and output gain parameters described in this embodiment all refer to the gain parameters in the terminal device on the same side, that is, the terminal device in the same scene as the speaker can perform the sound collected by the microphone. After the input gain is adjusted, it is transmitted to the other party that is talking to it. The sound transmitted by the other party that is talking to it can also be adjusted by the output gain and then played by the speaker. Therefore, when both the scenes of the conversation can be communicated using the terminal device of the embodiment, the two users have the best effect of speaking and listening.
  • the communication terminal of the embodiment of the present invention can not only adjust the volume of the input and output according to the actual received signal or parameter, so that the user communicating through the device does not need to manually adjust the input and output according to actual conditions.
  • the volume can get better listening results.
  • the echo signal in the audio signal can be further eliminated, the quality of the input and output audio signals is greatly improved, and the played sound is more clear.

Description

自适应音量调节的方法、装置及通信终端
本申请要求于2009年1月19日提交中国专利局、申请号为200910005236.3、发明名称为“自适应音量调节的方法、装置及通信终端”的中国专利申请的优先权，其全部内容通过引用结合在本申请中。
技术领域
本发明涉及通信技术领域，尤其涉及一种自适应音量调节的方法、装置及相应的通信终端。
发明背景
在现代社会，视讯会议作为一种多媒体通信手段越来越多的被人们在商务会议、论坛分会等大型会议中采用。视频会议系统主要包括视讯终端设备、传输网络设备、多点控制单元（Multipoint Control Unit, MCU）、业务管理平台等几部分，它可以同时实现两地或多个地点之间的图像、语音、数据的交互。其中语音的传输主要是将本地麦克风拾取的声音信号进行压缩编码，经过传输网络，传至远方会场。同时，接收远方会场传来的数字信号，经解码后，还原成声音信号，最后通过本地扬声器播放。在现有会议终端中调节音量大小的输入、输出增益一般都是固定的，当用户在走动中说话时，另一端会场中的用户听到的声音时大时小，导致本地用户无法较好的收听到远端会场所传输的音频信息。特别是“声学回声”的存在，使得会议现场的人们还时常能听见视讯终端设备的扬声器播放本方发言者的声音，无法给用户带来较好的体验。
在实现本发明的过程中，发明人发现现有技术中至少存在以下技术问题：现有视讯会议终端上的音量大小的调节都是通过手动调节，所以并不能很好的满足现代会议中各种特殊场景的要求。特别是“声学回声”现象处理不好，干扰了会议通信的质量。因为回声抵消效果的好坏取决于回声参考信号预测的是否准确，但视讯终端设备上的麦克风采集到的信号是由馈入回声和说话人的声音共同组成，而馈入回声信号受输入、输出增益影响较大，由于不能有效获得精确的输入、输出增益参数，因此通过回声抵消算法计算预测馈入回声信号与回声参考信号的比值难度较大，很难达到较好效果。
发明内容
本发明实施例的目的在于，提供一种自适应音量调节的方法、装置及通信终端，使得终端不需用户手动调节音量便能使用户获得很好的收听效果。
为实现上述发明目的，提供的技术方案如下：提供一种自适应音量调节的方法，包括：
接收来自声源的音频信号；
获取所述音频信号对应的声源信息；
获取与所述声源信息对应的输入、输出增益参数；
根据所述输入、输出增益参数调节输入、输出音量。
本发明实施例还提供一种自适应音量调节的方法，包括：
接收来自声源的音频信号；
接收到输入的输入增益参数或输出增益参数；
获取所述输入增益参数对应的输出增益参数，或者获取所述输出增益参数对应的输入增益参数；
根据所述输入、输出增益参数调节输入、输出音量。
本发明实施例还提供一种自适应音量调节的装置，包括：
接收单元，用于接收来自声源的音频信号；
获取单元，用于获取所述音频信号对应的声源信息，并获得所述声源信息对应的输入、输出增益参数；
调节单元，用于根据所述获取单元获取的输入、输出增益参数调节输入、输出音量。
本发明实施例还提供一种通信终端，包括，
自适应音量调节的装置，所述装置进一步包括：
接收单元，用于接收来自声源的音频信号；
获取单元，用于获取所述音频信号对应的声源信息，并获得所述声源信息对应的输入、输出增益参数；
调节单元，用于根据所述获取单元获取的输入、输出增益参数调节输入、输出音量。
本发明实施例根据获取的声源信息对应得到合适的输入、输出增益参数，通过所述输入、输出增益参数对输入音量和输出音量进行调节，使用户获得很好的收听效果。特别是可通过获得的输入、输出增益参数有效的消除音频中的回声，更好的保证声音的清晰。
附图简要说明
图1是本发明实施例一提供的自适应音量调节的方法的示意流程图；
图2是本发明实施例二提供的自适应音量调节的方法的示意流程图；
图3是本发明实施例三提供的自适应音量调节的方法的示意流程图；
图4是本发明实施例四提供的自适应音量调节的方法的示意流程图；
图5是本发明实施例五提供的自适应音量调节装置的原理框图；
图6是图5中的获取单元20的一种实施例的示意框图；
图7是图5中的获取单元20的另一种实施例的示意框图；
图8是图5中的调节单元30的一种实施例的示意框图。
实施本发明的方式
下面结合附图对本发明作进一步地详细描述。
本发明实施例是在通信终端上增加一种自动调节音量的技术，根据接收到的声源的信息，自动调节输入和输出的音量，使扬声器前的用户获得较佳的收听效果。
实施例一
图1是本发明实施例一的自适应音量调节的方法的示意流程图。该流程图包括以下步骤：
S101、接收来自声源的音频信号；
在实现S101时，本发明实施例中声源即为声音的源头，一般指说话的人。而音频信号则为来自声源的声音信号，即说话者的声音信号。
S102、获取所述音频信号对应的声源信息；
在实现S102时，获取所述音频信号对应的声源信息有两种方式，一种是通过所述音频信号获得，这时的声源信息主要指音频信号中的音量值；另一种就是接收用户直接输入预定的声源信息，这时的声源信息主要指接收所述音频信号的终端与所述声源之间的距离值。
S103、获取与所述声源信息对应的输入、输出增益参数；
在实现S103时，根据获取所述音频信号对应的声源信息的两种方式之一及组合获得的声源信息，择一获取所述声源信息对应的各自的输入、输出增益参数；
S104、根据所述输入、输出增益参数调节输入、输出音量。
在实现S104时，根据S103获取的所述输入、输出增益参数调节输入、输出音量。例如，当接收来自声源的音频信号后，根据音频信号中的音量值，获取与音量值对应的输入、输出增益参数，最后根据所述输入、输出增益参数调节输入、输出音量。或者，当接收来自声源的音频信号后，不根据音频信号中的音量值获取与音量值对应的输入、输出增益参数，而直接由用户输入认为合适的接收所述音频信号的终端与所述声源之间的距离值，根据所述距离值，获取与距离值对应的输入、输出增益参数，最后根据所述输入、输出增益参数调节输入、输出音量。这样用户不再需要不断手动调整音量的大小，完全由终端设备自动适应接收的音量，非常的方便。
需要说明的是，本实施例中的方法可根据具体情况选择一种获取所述音频信号对应的声源信息的方式，即可根据终端的性能和采集、播放设备的综合因素选取效果最好的方法。并且本实施例中通过音频信号也可以获取所述音频信号的终端与所述声源之间的距离值，进而获取与所述距离值对应的输入、输出增益参数。而所述的输入、输出增益参数均指的是同一侧终端设备中的增益参数，即与说话者在同一个场景中的终端设备可将麦克风采集到的声音进行输入增益调节后传输至与其通话的另一方，也可将与其通话的另一方传输来的声音进行输出增益调节后由扬声器播放出来。这样当通话双方的一方场景能实现本实施例的方法时，则可以提高双方用户说话和收听的效果；若对话的双方场景都能实现本实施例的方法时，则两方用户可达到说话和收听的最佳效果。
实施例二
图2是本发明实施例二的自适应音量调节的方法的示意流程图。该流程图包括以下步骤：
S201、接收来自声源的音频信号；
S202、获取接收到的所述声源与所述接收音频信号的终端之间的距离值；
S203、查找第一映射表，并获取所述距离值在所述第一映射表中对应的输入、输出增益参数；
S204、根据所述输入、输出增益参数调节输入、输出音量。
本发明实施例中接收所述声源与所述接收音频信号的终端之间的距离值为用户输入的说话者与所述终端之间的距离值，因为说话者与终端距离的远近直接影响到终端获取说话者声音信号的效果和远端的扬声器播放声音后用户收听的效果。比如距离值越大，说明说话者可能离终端较远，本地终端接获取采集的声音较小，为了保证远端用户的收听效果，就需要调大本地终端的输入增益参数，这样，在远端的终端的输出增益参数保持不变的情况下，也能够使得远端的用户不至于因为本地终端采集到的用户的声音较小，而听不清本地终端所采集到的声音。另一种实现方式，也可以通过调大远端终端的音频输出增益参数，使得远端用户能够听清本地采集到的声音；可以理解，也可以通过同时增大本地音频输入增益参数与远端的音频输出增益参数，来实现输出音量的调节。而距离值是可以根据用户的主观认识判断出来的，所以终端一旦接收到了用户输入的距离值，在第一映射表中，就应当获取所述距离值在所述第一映射表中对应的输入、输出增益参数。所述第一映射表为根据正常人的音量统计获得的说话者与终端的不同距离值对应的理想输入、输出增益参数表，即表中的每个距离值都对应一组输入、输出增益参数，通过所述的输入、输出增益参数调节接收的音频信号和播放的声音信号，终端就能接收到较理想输入、输出音量。例如：说话者距离终端大概2米，终端接收到输入的2米的信息后，查找第一映射表，并获取2米在所述第一映射表中对应的输入、输出增益参数＋3db和＋6db。终端根据所述输入、输出增益参数将输入音量放大3db，将输出音量放大6db。
可以理解，也可以通过在终端增加位置检测装置，来自动的检测所述本地终端与声源的距离值，并在终端内部设定距离值与本地输入增益参数和/或本地输出增益参数的映射关系，通过将检测的声源距离终端的距离值通过查询预先确定的所述第一映射表，得到对应的本地终端输入增益参数，以及远端输出增益参数。
可以知道，距离值也可以用其它的方式获得，例如通过图像处理的方式，来得到声源距离终端的距离，譬如利用终端设置的深度摄像机来获得声源与终端的距离，并将该距离值自动传输给终端，使终端参照该距离值根据预先确定的所述第一映射表得到本地终端的输入增益参数，和/或远处终端的输出增益参数，通过所述参数实现对本地终端和/或远处终端的音量进行调节。
本发明实施例的优点在于无需经过复杂的计算过程获得输入、输出增益参数，而且根据所述的参数调节输入、输出音量的效果也不错，从而非常容易在技术有限的终端上得到普及。
需要再次说明的是，本发明中的第一映射表是根据正常人在不同的距离说话时发出的声音结合所述不同的距离理想中输入、输出音量统计获得的，而且输入、输出音量还和相关硬件设备的性能有关。所以需要根据不同的硬件设备的性能统计所述第一映射表，这样的输入、输出音量的效果才能比较理想。而所述的输入、输出增益参数均指的是同一侧终端设备中的增益参数，即与说话者在同一个场景中的终端设备可将麦克风采集到的声音进行输入增益调节后传输至与其通话的另一方，也可将与其通话的另一方传输来的声音进行输出增益调节后由扬声器播放出来。这样当通话双方的一方场景能实现本实施例的方法时，则可以提高双方用户说话和收听的效果；若对话的双方场景都能实现本实施例的方法时，则两方用户可达到说话和收听的最佳效果。
实施例三
图3是本发明实施例三的自适应音量调节的方法的示意流程图。该流程图包括以下步骤：
S301、接收来自声源的音频信号；
S302、根据所述音频信号获得音量值；
S303、判断所述音量值是否在预定音量范围内；若在所述预定范围内，则转至S304；若不在所述预定范围内，则转至S305；
S304、根据所述音量值获取对应的输入增益参数；
S305、计算所述音量值与预定音量范围参数的差值来获取对应的输入增益参数；
S306、查找第二映射表，并获取所述输入增益参数在第二映射表中对应的输出增益参数；
S307、根据所述输入、输出增益参数调节输入、输出音量。
本发明实施例中根据接收到的声源音频信号获得其音量值，判断音量值是否在预定音量范围内。所述预定音量范围即指预先设置好的最佳音量值的范围，若接收到的音量值在这个范围内，则认为说话者与终端的距离是比较理想的，所以音量值对应的输入增益参数很合适，无需再对输入的音量做进一步的调节。若接收到的音量值不在这个范围内，则认为说话者与终端的距离不是比较理想的，所以音量值对应的输入增益参数不是很合适，需要再对输入的音量做进一步的调节。所以通过计算所述音量值与预定音量范围参数的差值来获取对应的输入增益参数，从而达到进一步调节输入音量的目的。根据没有调节的输入增益参数或者调节的输入增益参数，查找第二映射表，并获取所述输入增益参数在第二映射表中对应的输出增益参数；最后再根据输出增益参数调节输出音量。例如，接收到的音频信号中的音量值为60db，而预定音量范围为70db～75db，60db信号对应第三档（60db～65db）音量，所以与预定音量的差值为10db,即获得的输入增益参数值为+10db，而查找第二映射中所述输入增益参数值为+10db对应的输出增益参数值为+15db，所以将输入的音频信号中的音量值增大10db并且将扬声器播放的音量值增大15db。
本发明实施例中的音量值还可以被用来分析说话者的空间信息，从而获得说话者的方位角，根据所述方位角控制终端上的摄像头，使所述摄像头对准说话者，最终根据摄像头的聚焦点获得发言人距离终端的距离信息。根据所述距离信息中的距离值即可按照实施例二的方法完成对输入、输出音量的调节。
本实施例与实施例二相比的优点在于，整个方法无需用户输入任何参数，全部由终端自动完成，而经过判断、计算和对比过程能更精确的获得输入、输出增益参数，不仅简化了用户操作，而且能更精确的调节输入、输出音量。
需要说明的是，本实施例中的第二映射表是根据输入增益参数查找输出增益参数的表，所述表中的输出增益参数为输入增益参数对应的理想输出增益参数。而所述的输入、输出增益参数均指的是同一侧终端设备中的增益参数，即与说话者在同一个场景中的终端设备可将麦克风采集到的声音进行输入增益调节后传输至与其通话的另一方，也可将与其通话的另一方传输来的声音进行输出增益调节后由扬声器播放出来。这样当通话双方的一方场景能实现本实施例的方法时，则可以提高双方用户说话和收听的效果；若对话的双方场景都能实现本实施例的方法时，则两方用户可达到说话和收听的最佳效果。
实施例四
图4是本发明实施例四的自适应音量调节的方法的示意流程图。该流程图包括以下步骤：
S401、接收来自声源的音频信号；
S402、接收到输入的输入增益参数或输出增益参数；
S403、查找第三映射表，并获取所述输入增益参数在所述第三映射表中对应的输出增益参数，或者获取所述输出增益参数在所述第三映射表中对应的输入增益参数。
S404、根据所述输入、输出增益参数调节输入、输出音量。
本发明实施例中，用户只需要调节输入增益参数或输出增益参数，终端就会查找第三映射表，并获取所述输入增益参数在所述第三映射表中对应的输出增益参数，或者获取所述输出增益参数在所述第三映射表中对应的输入增益参数。从而根据所述输入、输出增益参数调节输入、输出音量。所述第三映射表根据输入增益参数或输出增益参数对应的理想输出增益参数或输入增益参数统计获得。
本实施例与上述实施例相比的优点在于，直接由用户手动输入一个输入增益参数或一个输出增益参数，终端便能自动匹配相对应的输出增益参数或输入增益参数，从而调节输入、输出的音量，特别是当用户调节输出增益参数后，不仅用户听到的声音非常理想，而且同时调节了输入增益参数，使输入的音量得到较好调整，非常简单实用。
需要说明的是，本实施例中的方法也可以与实施例一、实施例二或实施例三并存实现，即接收来自声源的音频信号的终端设备既可以实现实施例一所述的方法也可以实现本实施例所述的方法；或者既可以实现实施例二所述的方法也可以实现本实施例所述的方法；或者既可以实现实施例三所述的方法也可以实现本实施例所述的方法。这样用户就可以根据实际情况的不同有选择的使用一种自适应音量调节的方法来达到较好的效果。而所述的输入、输出增益参数均指的是同一侧终端设备中的增益参数，即与说话者在同一个场景中的终端设备可将麦克风采集到的声音进行输入增益调节后传输至与其通话的另一方，也可将与其通话的另一方传输来的声音进行输出增益调节后由扬声器播放出来。所以当对话的双方场景都能实现本实施例的方法时，两方用户说话和收听的效果最佳。
为了更好的获得输入、输出的音量的质量，必须对接收到的音频信号进行回声处理，所以在上述各个实施例中的获取输入、输出增益参数之后，还可以进行过滤音频信号的步骤：
根据所述输入、输出增益参数，通过计算或查表获得回声处理参数；
根据所述回声处理参数来过滤所述音频信号中的回声信号。
若以计算方式获得回声处理参数，可根据现有技术中输入、输出增益参数与回声处理参数已知的关系计算获得回声处理参数；若以查表的方法获得回声处理参数，则可根据预先存储的所述输入、输出增益参数与回声处理参数的对应表查找合适的回声处理参数。当然，在上述实施例中也可直接查表获得回声处理参数。例如在实施例二中，根据获得的距离值查找第一映射表，直接获得所述距离值对应的输入、输出增益参数和回声处理参数；在实施例三中，根据获得的输入增益参数查找第二映射表，直接获得所述输入增益参数对应的输出增益参数和回声处理参数；在实施例四中，根据接收到的输入增益参数或输出增益参数，查找第三映射表，直接获得所述输入增益参数或输出增益参数对应的输出增益参数或输入增益参数以及回声处理参数。
获得回声处理参数后即可以根据回声抵消算法等方式过滤掉所述音频信号中的回声信号。该过滤音频信号步骤的加入有效的消除了音频信号中的回声信号，大大提高了音频信号的质量，使得输入、输出的音量非常清晰。
实施例五
为了更好的实现上述方法，本发明实施例还提供了一种自适应音量调节装置，图5是本发明实施例的自适应音量调节装置的原理框图，如图5所示，所述装置包括：
接收单元10，主要用于接收来自声源的音频信号；
获取单元20，主要用于获取所述音频信号对应的声源信息，并获得所述声源信息对应的输入、输出增益参数；
调节单元30，主要用于根据所述获取单元20获取的输入、输出增益参数调节输入、输出音量。
本发明实施例的装置根据接收单元10接收的音频信号，通过获取单元20获取所述音频信号对应的声源信息，并获得所述声源信息对应的输入、输出增益参数，最终通过调节单元30根据获取单元20获取的参数调节输入、输出的音量。这样所述装置根据实际接收的音频信号、自适应调节输入、输出的音量，从而使通过该装置进行通信的用户无需根据实际情况手动调节输入、输出的音量就能获得较佳的收听效果。
图6是图5中的获取单元20的一个实施例的示意框图。如图6所示，所述获取单元20包括：
拾取子单元210, 用于拾取所述音频信号的音量值；
参数确定单元220, 用于根据所述音量值确定对应的输入、输出增益参数；
如图6所示，所述参数确定单元220还包括：
判断单元221，用于判断所述音量值是否在预定音量范围内，并生成判断结果；
计算单元222，用于当所述判断结果为否时，计算所述音量值与预定音量的比值获得输入增益参数；
查找单元223，用于当所述判断结果为是时，根据判断单元的判断结果查找到结果中所述输入增益参数在第二映射表中对应的输出增益参数，或者根据计算单元222计算结果查找到结果中所述输入增益参数在第二映射表中对应的输出增益参数。此处，第二映射表是根据输入增益参数查找输出增益参数的表，存储在所述查找单元223中，而所述表中的输出增益参数为输入增益参数对应的理想输出增益参数。
图7是图5中的获取单元20的另一种实施例的示意框图。如图7所示，所述获取单元20包括：
距离获取子单元230, 用于获取所述音频信号对应的声源信息对应的距离值；其中，距离值为根据声源的音量值，自动获得的距离值；
查询子单元240，用于获取所述距离值在第一映射表中对应的输入、输出增益参数；所述音频信号对应的声源信息即为音频信号的音量值，根据现有技术所述音量值还可以被用来分析说话者的空间信息，从而获得说话者的方位角，根据所述方位角控制终端设备上的摄像头，使所述摄像头对准说话者，最终终端设备根据摄像头的聚焦点获得发言人距离终端的距离信息，从而获得距离信息中的距离值，即所述音量值对应的距离值。所述第一映射表是根据正常人在不同的距离说话时发出的声音结合所述不同的距离理想中输入、输出音量统计获得的，而且输入、输出音量还和相关硬件设备的性能有关。所以需要根据不同的硬件设备的性能统计距离值与输入、输出增益的关系获得存储在所述查询子单元240中的所述第一映射表。
上述距离获取子单元230，用于获取接收到的所述音频信号对应的声源信息中的距离值还可以是用户手动输入的距离值；
而查询子单元240，则用于获取与所述手动输入距离值在所述第一映射表中对应的输入、输出增益参数。
当距离值通过用户手动输入时，本方案可用于对应实现实施例二的方法，距离获取子单元230通过获取到用户输入的距离值，就会通知查询子单元240根据所述第一映射表进行输入、输出增益参数的查询。需要说明的是，所述第一映射表可以通过预先设定，具体是根据正常人在不同的距离说话时发出的声音结合所述不同的距离理想中输入、输出音量统计获得的，而且输入、输出音量还和相关硬件设备的性能有关。所以需要根据不同的硬件设备的性能统计距离值与输入、输出增益的关系获得存储在所述查询子单元240中的所述第一映射表。
本实施例中的接收单元10还可以用于接收输入增益参数或输出增益参数；获取单元20还可以用于根据所述接收单元10接收到的输入增益参数或输出增益参数来获取输入增益参数对应的输出增益参数或者输出增益参数对应的输入增益参数，从而使得调节单元30，用于根据所述获取单元20获取的输入、输出增益参数调节输入、输出音量。其中所述获取单元20根据所述接收单元10接收到的输入增益参数或输出增益参数来获取输入增益参数对应的输出增益参数或者输出增益参数对应的输入增益参数具体做法为：所述获取单元20获取所述输入增益参数在所述第三映射表中对应的输出增益参数，或者获取所述输出增益参数在所述第三映射表中对应的输入增益参数。所述第三映射表根据输入增益参数或输出增益参数对应的理想输出增益参数或输入增益参数统计获得，可存储在所述获取单元20中。这样在本发明实施例中实施例四所述的方法和实施例一、实施例二、实施例三中任意一个实施例可在同一个自适应音量调节装置中实现。即在可以实现实施例一或实施例二或实施例三的装置中按上述方案增加实施例四的功能，比如当需要通过输入增益参数或输出增益参数来调节音量的时候，通过调节外设按钮来输入输入增益参数或输出增益参数启动执行实施例四所述的步骤。
图9是图5中的调节单元30的一个实施例的示意框图。所述调节单元30主要用于根据所述获取单元获取的输入、输出增益参数调节输入、输出音量。但为了输入、输出较好的声音质量，所述调节单元30还可以对输入、输出的声音中的回声信号进行过滤处理，从而得到较佳的输入、输出的声音质量。所以所述的调节单元30还包括：处理单元301，用于根据所述输入、输出增益参数，通过计算或查表获得回声处理参数；回声信号过滤单元302，用于根据所述回声处理参数来过滤所述音频信号中的回声信号。通过对获取单元20获得的输入、输出增益参数的处理，能得到用来过滤所述音频信号中的回声信号的回声处理参数，这样就可进一步消除所述音频信号中的回声信号，有效的还原了输入、输出的音频信号的质量，使得播放出的声音更加清晰，克服了现有技术中存在的问题。
需要说明的是，本实施例中所述的输入、输出增益参数均指的是同一侧终端设备中的增益参数，即与说话者在同一个场景中的终端设备可将麦克风采集到的声音进行输入增益调节后传输至与其通话的另一方，也可将与其通话的另一方传输来的声音进行输出增益调节后由扬声器播放出来。所以当对话的双方场景都能使用本实施例的装置时，两方用户说话和收听的效果最佳。
由上可以看出，本发明实施例的自适应音量调节装置中的获取单元可以根据接收单元接收到的音频信号、输入增益参数或者输出增益参数获取这些信号和参数各自对应的信息和输入、输出增益参数，从而根据所述输入、输出增益参数调节输入、输出音量。使得本发明实施例的自适应音量调节装置能适应各种场合，并且根据这些场合情况的不同选用适合的方案，比如：在一些小型会议中，本装置就可以针对小型会议中空间不大，人相对集中的特点，选用直接获取输入距离值再自动调节输入、输出音量的方案；在个人办公或家庭场景中，本装置就可以针对所述场景说话者较少，位置一般固定的特点选用先输入输入增益参数或输出增益参数再自动调节输入、输出音量的方案；而在一些较大型的会议中，本装置就可以针对这种会议场景空间较大，说话者不固定等较复杂情况选用获取音量值的方案。本装置还能进一步消除音频信号中的回声信号，大大提高了输入、输出音频信号的质量，使得播放出的声音更加清晰。
实施例六
本发明实施例还提供了一种通信终端，包括，
自适应音量调节的装置，所述装置进一步包括：接收单元，主要用于接收来自声源的音频信号；获取单元，主要用于获取所述音频信号对应的声源信息，并获得所述声源信息对应的输入、输出增益参数；调节单元，主要用于根据所述获取单元获取的输入、输出增益参数调节输入、输出音量。其中所述接收单元还可以用于接收输入增益参数或输出增益参数；所述获取单元还可以用于根据所述接收单元接收到的输入增益参数或输出增益参数来获取输入增益参数对应的输出增益参数或者输出增益参数对应的输入增益参数。所述调节单元还可以包括：处理单元，用于根据所述输入、输出增益参数，通过计算或查表获得回声处理参数；回声信号过滤单元，用于根据所述回声处理参数来过滤所述音频信号中的回声信号。
其中，获取单元包括：第一获取单元，用于获取所述音频信号的音量值；第二获取单元，用于获取所述音量值在预定音量范围内对应的输入、输出增益参数；或者包括第三获取单元，用于获取所述音频信号对应的声源信息对应的距离值；第四获取单元，用于获取所述距离值在第一映射表中对应的输入、输出增益参数；或者包括第五获取单元, 用于获取接收到的所述音频信号对应的声源信息中的距离值; 第六获取单元，用于获取所述距离值在所述第一映射表中对应的输入、输出增益参数。
需要说明的是，本实施例中所述的输入、输出增益参数均指的是同一侧终端设备中的增益参数，即与说话者在同一个场景中的终端设备可将麦克风采集到的声音进行输入增益调节后传输至与其通话的另一方，也可将与其通话的另一方传输来的声音进行输出增益调节后由扬声器播放出来。所以当对话的双方场景都能使用本实施例的终端设备通信时，两方用户说话和收听的效果最佳。
由上可以看出，本发明实施例的通信终端不仅可以根据实际接收的信号或者参数自适应调节输入、输出的音量，从而使通过该装置进行通信的用户无需根据实际情况手动调节输入、输出的音量就能获得较佳的收听效果。而且还能进一步消除音频信号中的回声信号，大大提高了输入、输出音频信号的质量，使得播放出的声音更加清晰。
本领域普通技术人员可以理解实现上述实施例方法中的全部或部分步骤是可以通过程序来指令相关的硬件完成，所述的程序可以存储于一可读取存储介质中，所述的存储介质可以为，如：ROM/RAM、磁碟、光盘等。
当然，以上所述是本发明的优选实施方式，应当指出，对于本技术领域的普通技术人员来说，在不脱离本发明原理的前提下，还可以做出若干改进和润饰，这些改进和润饰也视为本发明的保护范围。

Claims (17)

  1. 一种自适应音量调节的方法，其特征在于，包括：
    接收来自声源的音频信号；
    获取所述音频信号对应的声源信息；
    获取与所述声源信息对应的输入、输出增益参数；
    根据所述输入、输出增益参数调节输入、输出音量。
  2. 根据权利要求1所述的自适应音量调节的方法，其特征在于：所述声源信息包括接收所述音频信号的终端与所述声源之间的距离值，则所述获取与所述声源信息对应的输入、输出增益参数具体包括：
    查找第一映射表，并获取所述距离值在所述第一映射表中对应的输入、输出增益参数。
  3. 根据权利要求1所述的自适应音量调节的方法，其特征在于：所述声源信息包括接收到的所述音频信号的音量值，则所述获取与所述声源信息对应的输入、输出增益参数具体包括：
    判断所述音量值是否在预定音量范围内；
    若在所述预定范围内，则根据所述音量值获取对应的输入增益参数；
    若不在所述预定范围内，则计算所述音量值与预定音量范围参数的差值来获取对应的输入增益参数。
    查找第二映射表，并获取所述输入增益参数在第二映射表中对应的输出增益参数。
  4. 根据权利要求1-3中任一项所述的自适应音量调节的方法，其特征在于，在所述获取与所述声源信息对应的输入、输出增益参数之后，还包括：
    根据所述输入、输出增益参数，通过计算或查表获得回声处理参数；
    根据所述回声处理参数来过滤所述音频信号中的回声信号。
  5. 一种自适应音量调节的方法，其特征在于，包括：
    接收来自声源的音频信号；
    接收到输入的输入增益参数或输出增益参数；
    获取所述输入增益参数对应的输出增益参数，或者获取所述输出增益参数对应的输入增益参数；
    根据所述输入、输出增益参数调节输入、输出音量。
  6. 根据权利要求5所述的自适应音量调节的方法，其特征在于，所述获取所述输入增益参数对应的输出增益参数，或者获取所述输出增益参数对应的输入增益参数具体包括：
    查找第三映射表，并获取所述输入增益参数在所述第三映射表中对应的输出增益参数，或者获取所述输出增益参数在所述第三映射表中对应的输入增益参数。
  7. 根据权利要求5所述的自适应音量调节的方法，其特征在于，在所述获取所述输入增益参数对应的输出增益参数，或者获取所述输出增益参数对应的输入增益参数之后，还包括：
    根据所述输入、输出增益参数，通过计算或查表获得回声处理参数；
    根据所述回声处理参数过滤所述音频信号中的回声信号。
  8. 一种自适应音量调节的装置，其特征在于，包括：
    接收单元，用于接收来自声源的音频信号；
    获取单元，用于获取所述音频信号对应的声源信息，并获得所述声源信息对应的输入、输出增益参数；
    调节单元，用于根据所述获取单元获取的输入、输出增益参数调节输入、输出音量。
  9. 根据权利要求8所述的自适应音量调节的装置，其特征在于，所述获取单元包括：
    拾取子单元，用于拾取所述音频信号的音量值；
    参数确定单元，用于根据所述音量值确定对应的输入、输出增益参数。
  10. 根据权利要求9所述的自适应音量调节的装置，其特征在于，所述参数确定单元包括：
    判断单元，用于判断所述音量值是否在预定音量范围内，并生成判断结果；当所述判断结果为是时，将所述判断结果传送给查找单元，当所述判断结果为否时，将所述判断结果传送给计算单元；
    计算单元，用于计算所述音量值与预定音量的比值获得输入增益参数；
    查找单元，用于根据判断单元的判断结果或所述计算单元计算结果查找到结果中所述输入增益参数在第二映射表中对应的输出增益参数。
  11. 根据权利要求8所述的自适应音量调节的装置，其特征在于，所述获取单元包括：
    距离获取子单元，用于获取所述音频信号对应的声源信息对应的距离值；
    查询子单元，用于获取所述距离值在所述第一映射表中对应的输入、输出增益参数。
  12. 根据权利要求8所述的自适应音量调节的装置，其特征在于，
    所述接收单元还用于接收输入增益参数或输出增益参数；
    所述获取单元还用于根据所述接收单元接收的输入增益参数或输出增益参数来获取输出增益参数或者输入增益参数。
  13. 根据权利要求12所述的自适应音量调节的装置，其特征在于，所述获取单元根据所述接收单元接收的输入增益参数或输出增益参数来获取输出增益参数或者输入增益参数，具体为：
    获取所述输入增益参数在所述第三映射表中对应的输出增益参数，或者获取所述输出增益参数在所述第三映射表中对应的输入增益参数。
  14. 根据权利要求8-13中任一项所述的自适应音量调节的装置，其特征在于，所述调节单元还包括：
    处理单元，用于根据所述输入、输出增益参数，通过计算或查表获得回声处理参数；
    回声信号过滤单元，用于根据所述回声处理参数来过滤所述音频信号中的回声信号。
  15. 一种通信终端，其特征在于，包括，
    自适应音量调节的装置，所述装置进一步包括：
    接收单元，用于接收来自声源的音频信号；
    获取单元，用于获取所述音频信号对应的声源信息，并获得所述声源信息对应的输入、输出增益参数；
    调节单元，用于根据所述获取单元获取的输入、输出增益参数调节输入、输出音量。
  16. 根据权利要求15所述的通信终端，其特征在于，
    所述接收单元还用于接收输入增益参数或输出增益参数；
    所述获取单元还用于根据所述接收单元接收的输入增益参数或输出增益参数来获取输出增益参数或者输入增益参数。
  17. 根据权利要求15或16所述的通信终端，其特征在于，所述调节单元还包括：
    处理单元，用于根据所述输入、输出增益参数，通过计算或查表获得回声处理参数；
    回声信号过滤单元，用于根据所述回声处理参数来过滤所述音频信号中的回声信号。
PCT/CN2010/070244 2009-01-19 2010-01-18 自适应音量调节的方法、装置及通信终端 WO2010081434A1 (zh)

Priority Applications (2)

Application Number Priority Date Filing Date Title
EP10731068A EP2381738A4 (en) 2009-01-19 2010-01-18 COMMUNICATION TERMINAL AND DEVICE AND METHOD FOR ADAPTIVE VOLUME ADJUSTMENT
US13/186,166 US20110274293A1 (en) 2009-01-19 2011-07-19 Method, device and communication terminal for adjusting volume adaptively

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
CN200910005236.3 2009-01-19
CNA2009100052363A CN101478614A (zh) 2009-01-19 2009-01-19 自适应音量调节的方法、装置及通信终端

Related Child Applications (1)

Application Number Title Priority Date Filing Date
US13/186,166 Continuation US20110274293A1 (en) 2009-01-19 2011-07-19 Method, device and communication terminal for adjusting volume adaptively

Publications (1)

Publication Number Publication Date
WO2010081434A1 true WO2010081434A1 (zh) 2010-07-22

Family

ID=40839242

Family Applications (1)

Application Number Title Priority Date Filing Date
PCT/CN2010/070244 WO2010081434A1 (zh) 2009-01-19 2010-01-18 自适应音量调节的方法、装置及通信终端

Country Status (4)

Country Link
US (1) US20110274293A1 (zh)
EP (1) EP2381738A4 (zh)
CN (1) CN101478614A (zh)
WO (1) WO2010081434A1 (zh)

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN110830901A (zh) * 2019-11-29 2020-02-21 中国科学院声学研究所 一种用于调节扬声器音量的多通道扩声系统及方法
CN116033315A (zh) * 2023-03-30 2023-04-28 南昌航天广信科技有限责任公司 一种广播音量控制方法及系统

Families Citing this family (32)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN101478614A (zh) * 2009-01-19 2009-07-08 深圳华为通信技术有限公司 自适应音量调节的方法、装置及通信终端
CN101909116A (zh) * 2010-07-31 2010-12-08 宇龙计算机通信科技(深圳)有限公司 一种音量调节方法及系统
CN103139351B (zh) * 2011-11-24 2016-10-05 联想(北京)有限公司 音量控制方法、装置及通信终端
CN102693722A (zh) * 2012-05-23 2012-09-26 Tcl集团股份有限公司 一种语音识别的方法、装置及数字电视
CN102981422B (zh) * 2012-11-23 2015-12-23 广州华多网络科技有限公司 一种音量调节方法及系统
WO2014143060A1 (en) * 2013-03-15 2014-09-18 Intel Corporation Mechanism for facilitating dynamic adjustment of audio input/output (i/o) setting devices at conferencing computing devices
CN103533271B (zh) * 2013-10-18 2018-03-30 深圳Tcl新技术有限公司 电子设备开机音量的调整方法及电子设备
US9391575B1 (en) * 2013-12-13 2016-07-12 Amazon Technologies, Inc. Adaptive loudness control
CN105323352A (zh) * 2014-06-30 2016-02-10 中兴通讯股份有限公司 音频的调节方法及装置
CN104754099A (zh) * 2015-03-12 2015-07-01 深圳市金立通信设备有限公司 一种通话音量的调节方法
CN104754120B (zh) * 2015-03-12 2019-04-23 深圳市金立通信设备有限公司 一种终端
CN104703047B (zh) * 2015-03-23 2018-03-23 北京京东方多媒体科技有限公司 一种调节显示参数的方法、遥控器及显示装置
TWI604715B (zh) * 2015-05-28 2017-11-01 仁寶電腦工業股份有限公司 電話會議音量調整方法及系統
US9729118B2 (en) * 2015-07-24 2017-08-08 Sonos, Inc. Loudness matching
CN105375897A (zh) * 2015-11-30 2016-03-02 北京光年无限科技有限公司 一种面向智能机器人的环境信息处理方法和装置
CN105679346A (zh) * 2015-12-31 2016-06-15 深圳还是威健康科技有限公司 控制收音机的方法和装置
CN107197403B (zh) * 2016-03-15 2021-03-16 中兴通讯股份有限公司 一种终端音频参数管理方法、装置及系统
CN105872180A (zh) * 2016-03-28 2016-08-17 乐视控股（北京）有限公司 一种调节终端设备音量的方法和装置
CN107920263B (zh) * 2016-10-11 2021-10-29 杭州萤石网络股份有限公司 音量调节方法及装置
EP3513379A4 (en) * 2016-12-05 2020-05-06 Hewlett-Packard Development Company, L.P. AUDIO-VISUAL TRANSMISSION ADJUSTMENTS VIA OMNIDIRECTIONAL CAMERAS
CN106686249B (zh) * 2017-01-17 2020-04-24 维沃移动通信有限公司 一种语音通话方法及移动终端
US10074371B1 (en) * 2017-03-14 2018-09-11 Amazon Technologies, Inc. Voice control of remote device by disabling wakeword detection
CN107277209B (zh) * 2017-07-27 2020-05-29 维沃移动通信有限公司 一种通话调整方法及移动终端
CN107404682B (zh) 2017-08-10 2019-11-05 京东方科技集团股份有限公司 一种智能耳机
CN107566618B (zh) * 2017-08-18 2021-01-22 Oppo广东移动通信有限公司 音量调节方法、装置、终端设备及存储介质
CN110913062B (zh) * 2018-09-18 2022-08-19 西安中兴新软件有限责任公司 一种音频控制方法、装置、终端及可读存储介质
US10355658B1 (en) * 2018-09-21 2019-07-16 Amazon Technologies, Inc Automatic volume control and leveler
CN109495649B (zh) * 2018-12-14 2021-10-01 深圳市沃特沃德信息有限公司 音量调节方法、系统及存储介质
CN110347365B (zh) * 2019-07-12 2023-04-07 广东美的厨房电器制造有限公司 自动调节播报音量的方法和装置及声音播报设备
CN110719545B (zh) * 2019-09-12 2022-11-08 连尚（新昌）网络科技有限公司 音频播放设备及用于播放音频的方法
CN113571086B (zh) * 2020-04-28 2022-07-08 阿里巴巴集团控股有限公司 声音信号处理方法、装置、电子设备及可读存储介质
CN112863534B (zh) * 2020-12-31 2022-05-10 思必驰科技股份有限公司 噪声音频消除方法、语音识别方法

Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1367978A (zh) * 1999-06-01 2002-09-04 艾利森电话股份有限公司 对多种免提通信附件的自适应技术
CN101192813A (zh) * 2007-12-12 2008-06-04 四川长虹电器股份有限公司 自适应音量控制方法
CN101267189A (zh) * 2008-04-16 2008-09-17 深圳华为通信技术有限公司 音量自动调节装置、方法以及移动终端
CN101478614A (zh) * 2009-01-19 2009-07-08 深圳华为通信技术有限公司 自适应音量调节的方法、装置及通信终端

Family Cites Families (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6744882B1 (en) * 1996-07-23 2004-06-01 Qualcomm Inc. Method and apparatus for automatically adjusting speaker and microphone gains within a mobile telephone
AU3812497A (en) * 1997-07-24 1999-02-16 Qualcomm Incorporated Method and apparatus for automatically adjusting speaker and microphone gains within a mobile telephone
US7242784B2 (en) * 2001-09-04 2007-07-10 Motorola Inc. Dynamic gain control of audio in a communication device
EP1521241A1 (en) * 2003-10-01 2005-04-06 Siemens Aktiengesellschaft Transmission of speech coding parameters with echo cancellation
US8116485B2 (en) * 2005-05-16 2012-02-14 Qnx Software Systems Co Adaptive gain control system
US20070172083A1 (en) * 2006-01-25 2007-07-26 Cheng-Te Tseng Method and apparatus for controlling a gain of a voice signal
JP4727542B2 (ja) * 2006-09-26 2011-07-20 富士通株式会社 電子機器、そのエコーキャンセル方法、そのエコーキャンセルプログラム、記録媒体及び回路基板
KR101459319B1 (ko) * 2008-01-29 2014-11-07 삼성전자주식회사 오디오 볼륨 자동 조절 방법 및 장치

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1367978A (zh) * 1999-06-01 2002-09-04 艾利森电话股份有限公司 对多种免提通信附件的自适应技术
CN101192813A (zh) * 2007-12-12 2008-06-04 四川长虹电器股份有限公司 自适应音量控制方法
CN101267189A (zh) * 2008-04-16 2008-09-17 深圳华为通信技术有限公司 音量自动调节装置、方法以及移动终端
CN101478614A (zh) * 2009-01-19 2009-07-08 深圳华为通信技术有限公司 自适应音量调节的方法、装置及通信终端

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN110830901A (zh) * 2019-11-29 2020-02-21 中国科学院声学研究所 一种用于调节扬声器音量的多通道扩声系统及方法
CN116033315A (zh) * 2023-03-30 2023-04-28 南昌航天广信科技有限责任公司 一种广播音量控制方法及系统

Also Published As

Publication number Publication date
CN101478614A (zh) 2009-07-08
EP2381738A1 (en) 2011-10-26
US20110274293A1 (en) 2011-11-10
EP2381738A4 (en) 2012-08-01

Similar Documents

Publication Publication Date Title
WO2010081434A1 (zh) 自适应音量调节的方法、装置及通信终端
US8379076B2 (en) System and method for displaying a multipoint videoconference
WO2014044064A1 (zh) 一种自动控制手机双麦克风消噪的方法及手机
WO2012102464A1 (ko) 이어마이크로폰 및 이어마이크로폰용 전압 제어 장치
EP3228096B1 (en) Audio terminal
JP2003304590A (ja) リモコン装置と音量調整方法および音量自動調整システム
WO2021141197A1 (ko) 영상 감시 앰프 내장형 ip 스피커 시스템
WO2020155089A1 (zh) 蓝牙耳机的控制方法、蓝牙耳机及计算机可读存储介质
WO2014021670A1 (en) Mobile apparatus and control method thereof
WO2018147573A1 (ko) 인이어 마이크와 아웃이어 마이크 수음특성을 이용한 소음 제거 이어셋 및 소음 제거 방법
GB2351872A (en) Descriminating between voice and ringing tone data and sending each to a dedicated audio output
CN203747882U (zh) 一种高效检测ip电话音频性能的测试设备
WO2023005125A1 (zh) 蓝牙耳机的模式控制方法、设备及计算机可读存储介质
US8526589B2 (en) Multi-channel telephony
EP2216975A1 (en) Telecommunication device
JPS61172475A (ja) 会議用の完全デユープレツクス電話
CN210053486U (zh) 一种远程会议声音接收装置
CN210578519U (zh) 一种可一键对讲的执法仪设备
TWI548278B (zh) 音視訊同步控制設備及方法
WO2020101358A2 (ko) 이어셋을 이용한 서비스 제공방법
US20050037783A1 (en) Cordless telephone system
JP2001036881A (ja) 音声伝送システム及び音声再生装置
TW201530537A (zh) 電話語音輸出之方法及耳機
CN215300880U (zh) 一种助听器耳机
KR20000044065A (ko) 화상회의 시스템의 카메라 제어장치

Legal Events

Date Code Title Description
121 Ep: the epo has been informed by wipo that ep was designated in this application

Ref document number: 10731068

Country of ref document: EP

Kind code of ref document: A1

NENP Non-entry into the national phase

Ref country code: DE

WWE Wipo information: entry into national phase

Ref document number: 2010731068

Country of ref document: EP