WO2013078677A1 - 一种自适应调节音效的方法和设备 - Google Patents

一种自适应调节音效的方法和设备 Download PDF

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Publication number
WO2013078677A1
WO2013078677A1 PCT/CN2011/083373 CN2011083373W WO2013078677A1 WO 2013078677 A1 WO2013078677 A1 WO 2013078677A1 CN 2011083373 W CN2011083373 W CN 2011083373W WO 2013078677 A1 WO2013078677 A1 WO 2013078677A1
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Prior art keywords
value
current
voice
output volume
energy value
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PCT/CN2011/083373
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English (en)
French (fr)
Inventor
谢汉雄
黄妮
杜洪
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海能达通信股份有限公司
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Application filed by 海能达通信股份有限公司 filed Critical 海能达通信股份有限公司
Priority to DE112011105908.6T priority Critical patent/DE112011105908B4/de
Priority to PCT/CN2011/083373 priority patent/WO2013078677A1/zh
Priority to US14/361,367 priority patent/US9183846B2/en
Publication of WO2013078677A1 publication Critical patent/WO2013078677A1/zh

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Classifications

    • HELECTRICITY
    • H03ELECTRONIC CIRCUITRY
    • H03GCONTROL OF AMPLIFICATION
    • H03G3/00Gain control in amplifiers or frequency changers
    • H03G3/20Automatic control
    • H03G3/30Automatic control in amplifiers having semiconductor devices
    • H03G3/32Automatic control in amplifiers having semiconductor devices the control being dependent upon ambient noise level or sound level
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech 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
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208Noise filtering
    • HELECTRICITY
    • H03ELECTRONIC CIRCUITRY
    • H03GCONTROL OF AMPLIFICATION
    • H03G5/00Tone control or bandwidth control in amplifiers
    • H03G5/16Automatic control
    • H03G5/165Equalizers; Volume or gain control in limited frequency bands
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech 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
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208Noise filtering
    • G10L2021/02087Noise filtering the noise being separate speech, e.g. cocktail party
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/60Substation equipment, e.g. for use by subscribers including speech amplifiers
    • H04M1/6016Substation equipment, e.g. for use by subscribers including speech amplifiers in the receiver circuit

Definitions

  • the present invention relates to the field of communication technologies, and in particular, to a method and apparatus for adaptively adjusting sound effects. Background technique
  • wireless devices have been widely used, such as walkie-talkies, mobile phones, televisions, and the like.
  • a wireless device When using a wireless device, it is often necessary to adjust the sound of the device based on changes in the environment. For example, when using a walkie-talkie, the environment in which the user is located is often constantly changing. When the user changes from a quiet place to a noisy place, in order to get better sound and improve the clarity of the voice, the user often needs to manually turn up the volume. This method of requiring the user to manually adjust the volume is cumbersome, so that the user cannot concentrate on his work and reduces the user experience.
  • the method provided by the prior art adjusts the audio response by detecting the ambient noise level to improve the definition of the radio receiver in a high ambient noise region. This method can only adjust the resolution of the radio receiver, and does not automatically adjust the output volume of the radio receiver, so the adjusted sound is not ideal.
  • the prior art does not have a method of adjusting the volume and adjusting the audio response when the environment changes.
  • the method provided by the prior art adjusts the audio response by detecting the ambient noise level, and detecting the ambient noise level does not distinguish the true noise in the background environment from the speech interference well, and it is easy to voice the voice. Judging into noise, forming a false positive. The resulting ambient noise level does not truly reflect changes in ambient noise, and the resulting sound effects are not accurate. Summary of the invention
  • an embodiment of the present invention provides a method and a device for adaptively adjusting a sound effect, which can adjust the volume according to the current ambient noise energy, and can adjust the frequency response, thereby obtaining a better Sound effect adjustment results are simple to implement.
  • an embodiment of the present invention provides a method for adaptively adjusting a sound effect, the method being applied to a device having an audio output device, the method comprising:
  • the treble enhancement processing is performed; when it is determined that the energy value of the current environmental noise is less than the second threshold, the bass enhancement processing is performed.
  • the adjusting the current output volume according to the energy value of the current ambient noise comprises:
  • the method further includes: determining whether the sum value is greater than a third threshold, and if yes, using the third threshold as a current output volume value;
  • the method further includes:
  • the current output volume value is used as a reference volume value; the current ambient noise energy value corresponding to the current output volume value is used as a reference noise energy value.
  • the method before acquiring the energy value of the current ambient noise, the method further includes: Collecting sound data, performing voice activity detection on the sound data, and determining whether the sound data is voice;
  • the performing voice activity detection on the sound data comprises:
  • Obtaining a formant according to the LPC spectrum determining whether the formant meets a preset condition according to a characteristic of the formant; determining that the sound data is a voice when a preset condition is met; and when a preset condition is not met, It is judged that the sound data is not a voice.
  • the preset condition is:
  • the interval is obtained according to the LPC spectrum
  • the amplitude of the formant is in a preset interval, it is determined that the sound data is a voice.
  • the first triggering instruction is:
  • the device detects that there is a call access or the device receives an automatic adjustment command.
  • an embodiment of the present invention further provides an apparatus for adaptively adjusting a sound effect, the device having an audio output device, and the device includes:
  • a noise acquisition module configured to obtain an energy value of current environmental noise
  • a receiving module configured to receive a first trigger instruction
  • a first adjustment module configured to adjust a current output volume according to an energy value of the current ambient noise
  • a second adjustment module configured to perform a treble enhancement process when determining that the energy value of the current ambient noise is greater than a first threshold; and performing a bass enhancement process when determining that the energy value of the current ambient noise is less than a second threshold .
  • the first adjustment module includes a difference acquisition unit, a value acquisition unit, and an output volume acquisition unit, where
  • the difference obtaining unit is configured to obtain a difference between a current ambient noise energy value and a reference noise energy value
  • the sum value obtaining unit is configured to obtain a sum of the difference value and a reference volume value; wherein, the parameter The test volume value is an output volume value set by the recorded user, and the reference noise energy value has a corresponding relationship with the reference volume value;
  • the output volume acquisition unit is configured to adjust the volume of the device to the current output volume value by using the sum value as a current output volume value.
  • the sum value obtaining unit and the output volume acquiring unit further comprise: a first determining unit, configured to determine whether the sum value is greater than a third threshold, and if yes, the third wide The value is used as the current output volume value;
  • the second determining unit is configured to determine whether the sum value is less than a fourth threshold, and if so, the fourth threshold is used as a current output volume value.
  • the device further includes:
  • the voice detection module is configured to collect voice data, perform voice activity detection on the voice data, and determine whether the voice data is voice.
  • the voice detection module includes:
  • An LPC acquisition unit configured to acquire an LPC spectrum of the sound data
  • a third determining unit configured to obtain a formant according to the LPC spectrum, and when the formant meets a preset condition, determine that the sound data is a voice; when the preset condition is not met, determine that the sound data is not a voice .
  • the device is a walkie-talkie.
  • the method provided by the embodiment of the present invention can adjust the current output volume according to the current ambient noise energy value, and can be enhanced by treble or bass. Adjust the frequency response.
  • the method provided by the invention can automatically adjust the volume and audio response, and minimize the influence of environmental changes on the user's use for users who need to continuously move in different places, without the user's own adjustment. , to make users more focused on their work.
  • the method provided by the invention adjusts the volume and frequency response according to the current ambient noise energy value, and the adjustment effect is more comprehensive, so that the sound is clearer and more realistic, and a better sound effect is obtained, thereby improving the user experience.
  • FIG. 1 is a flowchart of a first embodiment of a method for adaptively adjusting sound effects according to an embodiment of the present invention
  • FIG. 2 is a flowchart of a second embodiment of a method for adaptively adjusting sound effects according to an embodiment of the present invention
  • FIG. 4 is a schematic diagram of a method for smoothly adjusting a volume according to an embodiment of the present invention.
  • FIG. 5 is a schematic diagram of adjusting a high-pitched enhanced frequency response according to an embodiment of the present invention.
  • FIG. 6 is a schematic diagram of adjustment of a bass enhanced frequency response according to an embodiment of the present invention.
  • FIG. 7 is a schematic diagram of a first embodiment of an apparatus for adaptively adjusting sound effects according to an embodiment of the present invention
  • FIG. 8 is a schematic diagram of a second embodiment of an apparatus for adaptively adjusting sound effects according to an embodiment of the present invention.
  • the embodiment of the invention provides a method and a device for adaptively adjusting the sound effect, which can adjust the volume according to the current ambient noise energy, and can adjust the frequency response, thereby obtaining a better sound effect adjustment result and being simple to implement. .
  • FIG. 1 is a flow chart of a first embodiment of a method for adaptively adjusting sound effects according to an embodiment of the present invention. The method is applied to a device having an audio output device, the method comprising:
  • S102 Receive a first trigger instruction, and adjust a current output volume according to the energy value of the current ambient noise.
  • the first threshold is a preset value used to represent the noise threshold.
  • the second threshold is a pre-set value used to characterize the quiet threshold.
  • the current output volume is adjusted according to the obtained energy value of the current ambient noise; when it is determined that the energy value of the current ambient noise is greater than the first threshold And performing a treble enhancement process; when it is determined that the energy value of the current ambient noise is less than a second threshold, performing a bass enhancement process.
  • the method provided by the embodiment of the invention can adjust the current output volume, adjust the frequency response by high-pitched enhancement or bass enhancement, obtain better sound effect adjustment result, and realize simple implementation.
  • the method provided by the embodiment of the present invention can be applied to a device having an audio output device, including but not limited to a walkie-talkie, a mobile phone, a television, a radio, etc., which is not limited by the present invention.
  • a device having an audio output device including but not limited to a walkie-talkie, a mobile phone, a television, a radio, etc.
  • the specific embodiment of the present invention will be described in detail below by taking a walkie-talkie as an example.
  • FIG. 2 is a flow chart of a second embodiment of a method for adaptively adjusting sound effects according to an embodiment of the present invention.
  • S202 Perform voice activity detection on the voice data, and determine whether the voice data is voice data.
  • the voice activity detection algorithm used is based on the characteristics of the speech formant, and first acquires an LPC (Linear Pred ivet ive Coding) spectrum of the sound data, and then according to the The LPC spectrum acquires a formant, and when the formant satisfies a preset condition, it is determined that the sound data is a voice; when the preset condition is not satisfied, it is determined that the sound data is not a voice.
  • LPC Linear Pred ivet ive Coding
  • step S202 specifically includes:
  • S202A Perform segmentation processing on the acquired sound data.
  • the frame overlap processing is performed on the data of the set frame length to reduce the discontinuity in the inter-frame time domain.
  • Place The frame length set may be a length of a data frame of 20 ms, and the overlap between frames is 10 ms.
  • S202B Perform pre-emphasis processing on the sound data of each segment.
  • the pre-emphasis processing is performed to compensate for the attenuation of the high frequency portion of the speech.
  • the transfer function of the pre-emphasis filter used is as shown in the formula (1):
  • window processing is performed here.
  • the windowing process is performed by the following formula:
  • Sw(n) is the windowed signal and S(n) is the pre-windowed signal
  • W (n) is a window function
  • N is the frame length.
  • the windowing function uses a Hamming algorithm, and the amplitude of the Hamming window is:
  • linear predictive analysis The basic idea of linear predictive analysis is that the present value of a speech sample can be approximated by a weighted linear combination of several speech samples of past values.
  • the system function is:
  • s (n) is the input speech signal and 1 is the serial number of the autocorrelation coefficient.
  • i is the current order
  • e(i) is the prediction variance under the current order
  • k(i) is the reflection system (! ⁇ )
  • the number, ' represents the jth prediction coefficient of the i-order predictor.
  • the frequency range of 100 Hz to 2800 Hz is divided into a plurality of intervals, and specifically, it can be divided into 9 sections, each section having a width of 300 Hz, and each section is searched one by one, if within the section If a formant is found, the value in the interval of the frame is recorded as the amplitude of the formant. M. If the formant is not found, then M is 0. Table 1 shows an example of dividing the frequency range.
  • S202G Determine, according to the obtained formant, whether the sound data is a voice.
  • the current frame is a speech frame.
  • the first formant frequency is less than 1000 Hz, that is, within the first three intervals
  • the formant curve is obtained by linear prediction, and it is determined whether the sound data is voice or not, and other algorithms can be used for language activity detection. Specifically, the following algorithm can be used for voice activity detection:
  • the energy-based algorithm makes a decision as to whether or not the speech is made by comparing the frame energy with a set threshold.
  • the set threshold can be updated by certain rules.
  • Pitch detection algorithm Depending on the principle that the periodicity of the speech signal is stronger than the noise signal, a verbal decision can be made. The advantage of this algorithm is that it can significantly reduce false triggering of shock and white noise.
  • Short-term information entropy algorithm This algorithm is based on the probability summation of the frequency sub-components of the speech signal to make a speech-free decision. The advantage is that the detection is accurate under low signal-to-noise ratio and stationary noise.
  • ⁇ ⁇ ⁇ _ ⁇ + ( ⁇ _ ⁇ ) ⁇ ⁇ ( 8 )
  • is the smoothing factor
  • ⁇ ' is the sound sample data
  • is the number of points included in one frame.
  • N 160 sample point data.
  • the first triggering instruction may be that the device detects that there is a call access or the device receives an automatic adjustment command.
  • the walkie-talkie detects that there is call access as the first trigger instruction.
  • the volume adjustment can also be triggered when an auto adjustment command is received.
  • S204A Obtain a difference between a current ambient noise energy value and a reference noise energy value.
  • the ambient noise energy value En calculated at the previous standby is used as the current noise energy value, and the difference from the reference noise energy value is calculated.
  • the reference volume value is an output volume value set by the recorded user, and the reference noise energy value has a corresponding relationship with the reference volume value. Specifically, when the user manually adjusts the output volume, the output volume set by the user can be used as the reference volume. The ambient noise energy calculated during standby before the user is manually set is used as the reference noise energy. The reference noise energy value is consistent with the current noise energy value calculation method. The reference noise energy value is changed only when the user manually adjusts the volume.
  • S204B Acquire a sum of the difference value and a reference volume value.
  • S204C using the sum value as a current output volume value, and adjusting a volume of the device to the current output volume value.
  • the current output volume value Vn can be calculated according to the following formula:
  • Vn Vr+ (En-Er) ( 9 ) where Vn is the current output volume value, Vr is the reference volume value, and En is the current noise energy value. Er is the reference noise energy value.
  • the increment of the volume change is obtained in increments of ambient noise.
  • the product of the sum value and the adjustment parameter may also be used as the current output volume value according to the specific situation, and the current output volume value may be obtained according to the following formula:
  • Vn [ Vr+ (En-Er) ] * a ( 10 )
  • Vn is the current output volume value
  • Vr is the reference volume value
  • En is the current noise energy value
  • Er is the reference noise energy value
  • a is the adjustment parameter.
  • the adjustment parameter a can be specifically set.
  • different adjustment parameters may be set according to different application scenarios, such as roads, factories, airports, sports venues, and the like, and the adjustment parameters are determined by an empirical value.
  • the adjustment parameters can provide variations in different volume increments.
  • the adjustment parameter can be set to 0.5.
  • the adjustment parameter can be set to 2.
  • the adjustment parameter can be set by the user or set by the system. When the adjustment parameter is 1, the current output volume value is equal to the sum of the reference volume value and the noise increase.
  • the method may further include:
  • the method before the sum value is used as the current output volume value, the method further includes determining whether the sum value is greater than a set third threshold value and determining whether the sum value is less than a set fourth value.
  • the third threshold and the fourth threshold are preset values that characterize the range of volume gain adjustment.
  • 3 is a schematic diagram of a volume adjustment curve according to an embodiment of the present invention. Among them, curve A is the maximum adjustable gain curve, curve C is the minimum adjustable gain curve, and curve B is the normal gain curve. When the range is exceeded, the volume adjustment is no longer performed, so that the user can obtain more Good hearing effect.
  • the method further includes: determining whether the product of the sum value and the adjustment parameter is greater than the third threshold. If yes, the third threshold is used as the current output volume value; determining whether the product of the sum value and the adjustment parameter is less than a fourth threshold, and if so, using the fourth threshold as the current output volume value. In this way, the volume adjustment can be kept in an appropriate range to give the user a better hearing effect.
  • the reference volume may specifically be a volume value when the user manually adjusts the volume.
  • the reference volume value is updated each time the user manually adjusts the volume.
  • the current ambient noise energy value corresponding to the current output volume value is used as the reference noise energy value.
  • FIG. 4 is a schematic diagram of a method for smoothly adjusting volume according to an embodiment of the present invention.
  • line 1 indicates a case where the volume is turned up
  • line 2 indicates a case where the volume is turned down.
  • the frequency response of the normal speech filter in the hearing range is flat.
  • the human ear is relatively sensitive to high-frequency sound. If the ambient noise is high, it is easier to hear the speaker's content when the high frequency is raised. However, when the ambient is quiet, the high-frequency part of the voice is too much, which sounds harsh. .
  • the present invention can also adaptively adjust the frequency response according to the surrounding environment, so that the user can obtain a better sense of hearing.
  • the step S205 may specifically include:
  • the first threshold value may be preset to be a noise threshold.
  • the treble enhancement filter is used to adjust the frequency response. As shown in Figure 5, the treble enhances the frequency response adjustment diagram.
  • the bass enhancement processing is performed.
  • the second threshold may be preset to be a quiet threshold.
  • the bass enhancement filter is used to adjust the frequency response.
  • Figure 6 it is a schematic diagram of the bass boost frequency response adjustment. It can be seen from FIG. 5 and FIG. 6 that the method provided by the embodiment of the present invention improves the frequency response when the frequency response is adjusted, and a better sound effect can be obtained.
  • voice activity detection is performed on the sound data, thereby determining whether the sound data is voice, and when it is determined that the sound data is not voice, it is used as current environmental noise.
  • FIG. 7 is a schematic diagram of a first embodiment of an apparatus for adaptively adjusting sound effects according to an embodiment of the present invention.
  • the device has an audio output device, and the device includes:
  • the noise acquiring module 1 00 is configured to obtain an energy value of the current ambient noise
  • the receiving module 200 is configured to receive a first trigger instruction
  • the first adjustment module 300 is configured to adjust the current output volume according to the energy value of the current ambient noise
  • the second adjustment module 400 is configured to perform a treble enhancement process when determining that the energy value of the current ambient noise is greater than the first threshold; and perform bass enhancement when determining that the energy value of the current ambient noise is less than the second threshold deal with.
  • FIG. 8 is a schematic diagram of a second embodiment of an apparatus for adaptively adjusting sound effects according to an embodiment of the present invention.
  • the first adjustment module includes a difference obtaining unit 31 0, a value acquiring unit 320, and an output volume acquiring module 350, where
  • the difference obtaining unit 31 0 is configured to obtain a current ambient noise energy value and a reference noise energy value. Difference
  • the sum value obtaining unit 320 is configured to obtain a sum of the difference value and the reference volume value, where the reference volume value is an output volume value set by the recorded user, the reference noise energy value and the reference volume Values have a corresponding relationship;
  • the output volume acquisition unit 350 is configured to adjust the volume of the device to the current output volume value by using the sum value as a current output volume value.
  • the sum value acquiring unit 320 and the output volume acquiring unit 350 further include:
  • the first determining unit 330 is configured to determine whether the sum value is greater than a third threshold, and if yes, using the third threshold as a current output volume value;
  • the second determining unit 340 is configured to determine whether the sum value is less than a fourth threshold, and if so, the fourth threshold is used as a current output volume value.
  • the device further includes:
  • the voice detection module 500 is configured to collect voice data, perform voice activity detection on the voice data, and determine whether the voice data is voice.
  • the voice detection module includes:
  • the LPC acquisition unit 51 0 is configured to acquire an LPC spectrum of the sound data.
  • a third determining unit 520 configured to acquire a formant according to the LPC spectrum, and when the formant meets a preset condition, determine that the sound data is a voice; when the preset condition is not met, determine that the sound data is not voice.
  • the device is a walkie-talkie.
  • the device can also be any device having an audio output device, such as a cell phone, television, radio, and the like.
  • the invention may be described in the general context of computer-executable instructions executed by a computer, such as a program module.
  • program modules include routines, programs, objects, components, data structures, and the like that perform particular tasks or implement particular abstract data types.
  • the invention may also be practiced in distributed computing environments where tasks are performed by remote processing devices that are connected through a communication network.
  • program modules can be located in both local and remote computer storage media including storage devices.

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Abstract

一种自适应调节音效的方法和设备,所述方法包括:获取当前环境噪声的能量值;接收第一触发指令,根据所述当前环境噪声的能量值对当前输出音量进行调节;当判断所述当前环境噪声的能量值大于第一阈值时,进行高音增强处理;当判断所述当前环境噪声的能量值小于第二阈值时,进行低音增强处理。所述方法通过采集声音数据,对所述声音数据进行语音活动检测,在接收到第一触发指令时,根据当前环境噪声能量值既可以对当前输出音量进行调节,又可以通过高音增强或低音增强对频响进行调节,获得了更好的音效,而且实现简单。

Description

一种自适应调节音效的方法和设备
技术领域
本发明涉及通信技术领域, 特别是涉及一种自适应调节音效的方法和设 备。 背景技术
随着无线通信技术的发展, 无线设备得到了广泛的应用, 例如对讲机、 手 机、 电视等。 在使用无线设备时, 常常需要根据环境的变化对设备的音效进行 调节。 例如使用对讲机时, 用户所处的环境往往处于不断变化当中。 当用户从 一个安静的地方变换到一个嘈杂的地方时, 为了获取更好的音效,提高语音的 清晰度, 用户往往需要手动调高音量。 这种需要用户手动调节音量的方法比较 麻烦, 使用户不能专注于自己的工作, 降低了用户的使用体验。
现有技术中,存在一种自动调节音频响应以改善清晰度的方法, 其根据环 境噪音电平自动地调节音频响应来改善无线电接收机在高环境噪音区域中的 清晰度。 这种方法通过测定环境噪音电平, 当存在高环境噪音时, 通过访问一 个音频参数数组对音频响应进行调节。 具体调节时,是以低的频率响应为代价 增大较高的音频频率的相对增益以改善清晰度。这种情况下,通过增大高频增 益所造成的较高清晰度是以一定的逼真度为交换的。
在实现本发明的过程中,发明人发现现有技术中至少存在如下问题: 现有 技术提供的方法通过检测环境噪音电平来调节音频响应以改善无线电接收机 在高环境噪音区域中的清晰度,这种方法只能对无线电接收机的清晰度进行调 节, 并不能自动对无线电接收机的输出音量进行调节, 由此调节的音效并不理 想。现有技术并不存在当环境发生变化时, 既能对音量进行调节又能对音频响 应进行调节的方法。
另一方面, 现有技术提供的方法通过检测环境噪音电平来调节音频响应, 而检测环境噪音电平并不能很好地将背景环境中的真正噪声与语音干扰区分 开来, 容易将语音也判断成噪声, 形成误判。 由此获取的环境噪音电平并不能 真实地反映出环境噪音的变化, 由此调节的音效结果并不准确。 发明内容
为解决上述技术问题,本发明实施例提供了一种自适应调节音效的方法和 设备 ,根据当前环境噪声能量既可以对音量进行调节,又可以对频响进行调节, 由此获取了更好的音效调节结果, 实现简单。
一方面, 本发明实施例提供了一种自适应调节音效的方法, 所述方法应用 于具有音频输出装置的设备, 所述方法包括:
获取当前环境噪声的能量值;
当收到第一触发指令时,根据所述当前环境噪声的能量值对当前输出音量 进行调节;
当判断所述当前环境噪声的能量值大于第一阔值时, 进行高音增强处理; 当判断所述当前环境噪声的能量值小于第二阔值时, 进行低音增强处理。
优选的 ,所述根据所述当前环境噪声的能量值对当前输出音量进行调节包 括:
获取当前环境噪声的能量值与参考噪声能量值的差值;
获取所述差值与参考音量值的和值,将所述和值作为当前输出音量值, 将 所述设备的音量调节至所述当前输出音量值; 其中, 所述参考音量值为记录的 用户设置的输出音量值, 所述参考噪声能量值与所述参考音量值具有对应关 系。
优选的, 在将所述和值作为当前输出音量值之前, 所述方法进一步包括: 判断所述和值是否大于第三阔值,如果是,将所述第三阔值作为当前输出 音量值;
和\或
判断所述和值是否小于第四阔值,如果是,将所述第四阔值作为当前输出 音量值。
优选的, 所述方法还包括:
当用户对当前输出音量值进行调节时,将所述当前输出音量值作为参考音 量值;将与所述当前输出音量值对应的当前环境噪声能量值作为参考噪声能量 值。
优选的, 在获取当前环境噪声的能量值之前, 所述方法还包括: 釆集声音数据,对所述声音数据进行语音活动检测, 判断所述声音数据是 否为语音;
当判断所述声音数据不是语音时, 将所述声音数据作为当前环境噪声。 优选的, 所述对所述声音数据进行语音活动检测包括:
获取所述声音数据的 LPC谱;
根据所述 LPC谱获取共振峰,根据所述共振峰的特征判断所述共振峰是否 满足预设条件; 当满足预设条件时, 判断所述声音数据为语音; 当不满足预设 条件时, 判断所述声音数据不是语音。
优选的, 所述预设条件为:
对于连续多个数据帧, 当有至少两个共振峰在同一区间时, 判断所述声音 数据为语音; 所述区间根据所述 LPC谱划分得到;
和 /或
所述共振峰的幅值在预设区间时, 判断所述声音数据为语音。
优选的, 所述第一触发指令为:
所述设备检测到有呼叫接入或者所述设备接收到自动调节命令。
另一方面, 本发明实施例还提供了一种自适应调节音效的设备, 所述设备 具有音频输出装置, 所述设备包括:
噪声获取模块, 用于获取当前环境噪声的能量值;
接收模块, 用于接收第一触发指令;
第一调节模块,用于根据所述当前环境噪声的能量值对当前输出音量进行 调节;
第二调节模块, 用于当判断所述当前环境噪声的能量值大于第一阔值时, 进行高音增强处理; 当判断所述当前环境噪声的能量值小于第二阔值时, 进行 低音增强处理。
优选的, 所述第一调节模块包括差值获取单元、 和值获取单元、 输出音量 获取单元, 其中,
所述差值获取单元用于获取当前环境噪声能量值与参考噪声能量值的差 值;
所述和值获取单元用于获取所述差值与参考音量值的和值; 其中, 所述参 考音量值为记录的用户设置的输出音量值,所述参考噪声能量值与所述参考音 量值具有对应关系;
所述输出音量获取单元用于将所述和值作为当前输出音量值 ,将所述设备 的音量调节至所述当前输出音量值。
优选的, 在所述和值获取单元与所述输出音量获取单元之间进一步包括: 第一判断单元, 用于判断所述和值是否大于第三阔值, 如果是, 将所述第 三阔值作为当前输出音量值;
和\或
第二判断单元用于判断所述和值是否小于第四阔值,如果是,将所述第四 阈值作为当前输出音量值。
优选的, 所述设备还包括:
语音检测模块, 用于釆集声音数据, 对所述声音数据进行语音活动检测, 判断所述声音数据是否为语音。
优选的, 所述语音检测模块包括:
LPC获取单元, 用于获取所述声音数据的 LPC谱;
第三判断单元, 用于根据所述 LPC谱获取共振峰, 当所述共振峰满足预设 条件时, 判断所述声音数据为语音; 当不满足预设条件时, 判断所述声音数据 不是语音。
优选的, 所述设备为对讲机。
本发明实施例能够达到的有益效果为:本发明实施例提供的方法在接收到 第一触发指令时, 根据当前环境噪声能量值既可以对当前输出音量进行调节, 又可以通过高音增强或低音增强对频响进行调节。 当环境发生变化时, 本发明 提供的方法可以自动对音量和音频响应进行调节,对于需要在不同场所不断移 动的用户, 最大限度的减小环境变化对用户使用上的影响, 无需用户自己来调 节, 使用户更能专注于自己的工作。 另一方面, 本发明提供的方法根据当前环 境噪声能量值,对音量和频响进行调节,调节效果更全面,使得声音更加清晰、 逼真, 获取了更好的音效, 提升了用户体验。 附图说明
为了更清楚地说明本发明实施例或现有技术中的技术方案,下面将对实施 例或现有技术描述中所需要使用的附图作简单地介绍,显而易见地, 下面描述 中的附图仅仅是本发明中记载的一些实施例, 对于本领域普通技术人员来讲, 在不付出创造性劳动的前提下, 还可以根据这些附图获得其他的附图。
图 1为本发明实施例提供的自适应调节音效的方法第一实施例流程图; 图 2为本发明实施例提供的自适应调节音效的方法第二实施例流程图; 图 3为本发明实施例音量调整曲线示意图;
图 4为本发明实施例提供的平滑调节音量方法示意图;
图 5为本发明实施例提供的高音增强频响调节示意图;
图 6为本发明实施例提供的低音增强频响调节示意图;
图 7为本发明实施例提供的自适应调节音效的设备第一实施例示意图; 图 8为本发明实施例提供的自适应调节音效的设备第二实施例示意图。
具体实施方式
本发明实施例提供了一种自适应调节音效的方法和设备,根据当前环境噪 声能量既可以对音量进行调节, 又可以对频响进行调节, 由此获取了更好的音 效调节结果, 实现简单。
为了使本技术领域的人员更好地理解本发明中的技术方案,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做出创造性劳动前提下所获 得的所有其他实施例, 都应当属于本发明保护的范围。
参见图 1 ,为本发明实施例提供的自适应调节音效的方法第一实施例流程 图。 所述方法应用于具有音频输出装置的设备, 所述方法包括:
5101 , 获取当前环境噪声的能量值。
5102 ,接收第一触发指令,根据所述当前环境噪声的能量值对当前输出音 量进行调节。
5103 , 当判断所述当前环境噪声的能量值大于第一阔值时, 进行高音增强 处理;当判断所述当前环境噪声的能量值小于第二阈值时,进行低音增强处理。 其中, 第一阔值为预先设定的值, 用于表征噪声门限。 当当前环境噪声的 能量值大于第一阔值时, 表征当前处于较嘈杂的环境, 为了使话音更清晰, 因 此进行高音增强处理。 第二阔值为预先设定的值, 用于表征安静门限。 当判断 所述当前环境噪声的能量值小于第二阔值时,表征当前处于较安静的环境, 为 了是话音获得更柔和的效果, 进行低音增强处理。
在本发明实施例中, 当接收到第一触发指令时,根据获取的所述当前环境 噪声的能量值对当前输出音量进行调节;当判断所述当前环境噪声的能量值大 于第一阔值时, 进行高音增强处理; 当判断所述当前环境噪声的能量值小于第 二阔值时, 进行低音增强处理。本发明实施例提供的方法既可以对当前输出音 量进行调节, 又可以通过高音增强或低音增强对频响进行调节, 获得了更好的 音效调节结果, 而且实现简单。
本发明实施例提供的方法可以应用于具有音频输出装置的设备,包括但不 限于对讲机、 手机、 电视、 收音机等, 本发明对此不做限制。 下面以对讲机为 例, 对本发明具体实施例进行详细介绍。
参见图 2 , 为本发明实施例提供的自适应调节音效的方法第二实施例流程 图。
5201 , 釆集声音数据。
当对讲机处于待机模式时, 打开麦克风, 釆集环境声音数据, 获取声音数 据帧。
5202 ,对所述声音数据进行语音活动检测, 判断所述声音数据是否为语音 数据。
在本发明的一个优选实施例中,所使用的语音活动检测算法是基于语音共 振峰的特性, 先获取所述声音数据的 LPC ( L inear Pred ict ive Coding,线性 预测编码)谱,再根据所述 LPC谱获取共振峰, 当所述共振峰满足预设条件时, 判断所述声音数据为语音;当不满足预设条件时,判断所述声音数据不是语音。
具体的, 步骤 S202具体包括:
S202A, 对获取的声音数据进行分段处理。
对于设定帧长的数据进行帧重叠处理, 以减少帧间时域上的不连续性。 所 述设定的帧长可以为 20ms的数据帧的长度, 帧间重叠为 10ms。
S202B, 对每一段的声音数据进行预加重处理。
这里, 进行预加重处理是为了补偿语音高频部分的衰减。在本发明实施例 中, 使用的预加重滤波器的传递函数如公式(1 ) 所示:
y(n) =x (n)-0.92x(n-l); n=0, 1, 2, .. N-l (1 ) 其中, y(n)代表滤波器输出, x(n)代表滤波器输入 ,N为帧长。
S202C, 进行加窗处理。
为了减少频语泄漏, 这里进行加窗处理。 在本发明实施例中, 通过以下公 式进行加窗处理:
Sw (n) =S (n) W (n) n=0, 1, 2, .. N-l (2) 其中, Sw(n)为加窗后信号, S(n)为加窗前信号, W(n)为窗函数, N 为帧 长。
在本发明实施例中, 加窗函数釆用汉明窗 (Hamming) 算法, 所述汉明窗 的幅值为:
W(n)=0.54-0.46cos (2*pi*n/ (N-l)) , n=0, 1, 2, .. N-l ( 3)
S202D, 获取 LPC系数。
线性预测分析的基本思想为:一个语音取样的现在值可以用若干个语音取 样过去值的加权线性组合来逼近。 ***函数为:
Figure imgf000008_0001
其中, α' (i=l,2,.. p)为线性预测系数, p为阶数
求 LPC系数 α '主要分为两个部分:
( 1 )根据输入语音信号求自相关系数
(5) 1
R(l) =— S s(n)s(n -/),/ = 0,1,2,. 其中, s (n)为输入语音信号, 1为自相关系数的序号,
( 2 )根据 Levinson-Durbin算法计算 LPC系数
递推过程如下所示:
e(0) = R(0)
for i = l,2,..,p
=剛 - "R( - «)]/φ· - 1)
e{i) = (l-k{i)k{i))e(i-l)
fl,(i) = k(i)
if i> \ then for m =l,2,..,i-l
a, (,''■) -k{i)a ('■— i)
end
end
在上表中, i为当前阶数, e(i)为当前阶数下的预测方差, k(i)为反射系 (!■)
数, '表示 i阶预测器的第 j个预测系数。
通过上述推导, 得出:
α =α^, = 1,2,..,ρ ( 6 )
这样, 线性预测系数 (i=l, 2, .. p)就计算出来了。
S202E, 获取 LPC谱。
根据之前得到的线性预测系数 (i=l, 2, · · p) ,可以求出*** H(z)的频响:
Figure imgf000009_0001
| (e;iU)l即为所要求的 LPC谱, 其中 ω为角频率。
S202F, 查找共振峰。
根据求出的 LPC谱, 将 100Hz到 2800Hz频率范围内分为多个区间, 具体 的,可以划分为 9个区间,每个区间宽度为 300Hz,对每个区间逐一进行查找, 如果在该区间内查找到有共振峰,则该帧该区间内的值记录为此共振峰的幅值 M。 如果找不到共振峰, 则 M记为 0。 表 1示出了一个对频率范围进行划分的 示例。
表 1
Figure imgf000010_0001
S202G, 根据获取的共振峰, 判断所述声音数据是否是语音。
根据仿真的结果, 当满足以下两个条件时, 判定当前帧为语音帧。
( 1 )连续 3帧, 至少有两个共振峰在同一个区间内, 并且共振峰幅值 Μ 大小处于一定范围内, 即 Thresholdl<M< Thresho ld2 , 其中, Thresho ldl与 Threshold2为预先设定的值。
( 2 )第一个共振峰频率小于 1000Hz , 即处于前 3个区间内
在上述实施例中, 是釆用线性预测的方式获取共振峰曲线, 进而判别声音 数据是否为语音, 也可以釆用其他算法进行语言活动检测。 具体的, 可以釆用 以下算法来进行语音活动检测:
( 1 )基于短时能量的算法。 基于能量的算法通过帧能量和一个设定阔值 的比较来做出是否是语音的判决, 设定的阔值可以通过一定的规则进行更新。 这种算法的优点是简单, 实时性好, 特别是基于时域的算法, 在较高性噪比条 件下具有较好的性能。
( 2 )基于过零率的算法。 由于噪声的过零率一般较语音高且是随机的, 尽管清音的过零率也较高,但清音的分布是在一定的范围之内, 可以通过其分 布区别清音与噪声。 因此, 通过过零率的值可以判别出信号是浊音、 清音或噪 声。
( 3 )基音检测算法。 根据语音信号的周期性比噪声信号强的原则可以做 出有话无话判决。这种算法的优点是可以明显的减少对冲击噪声和白噪声的误 触发。 ( 4 )短时信息熵算法。 这种算法是根据语音信号的频率子成分出现概率 求和来进行有话无话判决, 其优点是在信噪比低和平稳噪声下, 检测准确。
S203 , 当判断所述声音数据不是语音时, 将其作为当前环境噪声, 获取所 述当前环境噪声的能量值。
Εη = ρΕη_ι + (ι _ ρ)^ Χι ( 8 )
其中, 为上一个非语音帧所计算得到的能量值, ^是平滑因子, χ'是 声音釆样数据, Ν是一个帧所包含釆样点数。例如如帧长 20ms、釆样率为 8KHz 的声音帧总共有 N=160个釆样点数据。
S204 , 当接收到第一触发指令时,根据所述当前环境噪声的能量值对当前 输出音量进行调节。
具体的,所述第一触发指令可以是所述设备检测到有呼叫接入或者所述设 备接收到自动调节命令。 具体到本发明实施例,将对讲机检测到有呼叫接入作 为第一触发指令。 对于其他设备, 也可以在接收到自动调节命令时, 触发音量 调节。
S204A , 获取当前环境噪声能量值与参考噪声能量值的差值。
当接收到新呼叫时, 将此前待机时计算的环境噪声能量值 En作为当前噪 声能量值, 计算其与参考噪声能量值的差值。 在本发明实施例中, 所述参考音 量值为记录的用户设置的输出音量值,所述参考噪声能量值与所述参考音量值 具有对应关系。 具体的, 当用户手动调节输出音量时, 可以将用户设置的输出 音量作为参考音量。将用户进行手动设置前待机时计算出的环境噪声能量作为 参考噪声能量。参考噪声能量值与当前噪声能量值的计算方法是一致的, 参考 噪声能量值只在用户手动调整音量时才改变。
S204B, 获取所述差值与参考音量值的和值。
S204C , 将所述和值作为当前输出音量值, 将所述设备的音量调节至所述 当前输出音量值。
具体的, 可以按照以下公式计算当前输出音量值 Vn :
Vn=Vr+ (En-Er) ( 9 ) 其中, Vn为当前输出音量值, Vr为参考音量值, En为当前噪声能量值, Er为参考噪声能量值。
在本发明实施例中,是以环境噪声的增量得到音量改变的增量。也可以根 据具体情况,将所述和值与调节参数的乘积作为当前输出音量值, 具体可以按 照以下公式获取当前输出音量值:
Vn= [ Vr+ (En-Er) ] * a ( 10 ) 其中, Vn为当前输出音量值, Vr为参考音量值, En为当前噪声能量值, Er为参考噪声能量值, a为调节参数。
这里,调节参数 a可以具体设定。一般可以根据不同的应用场景,如马路, 工厂、 机场、 运动场馆等用户使用设备的场所的不同, 设置不同的调节参数, 所述调节参数由一经验值确定。 所述调节参数可以提供不同的音量增量的变 化。 具体的, 可以设置调节参数为 0. 5 , 这时, 当环境噪声能量增加 2db时, 当前输出音量增加 ldb。 也可以设置调节参数为 2 , 这时, 当环境噪声能量增 加 ldb时, 当前输出音量增加 2db。 调节参数可以由用户具体设定, 也可以由 ***设定。 当调节参数为 1时, 则当前输出音量值即等于参考音量值与噪声增 量的和值。
在本发明另一实施例中, 在步骤 S204B之后、 步骤 S204C之前, 进一步还 可以包括:
判断所述和值是否大于第三阔值,如果是,将所述第三阔值作为当前输出 音量值; 判断所述和值是否小于第四阔值, 如果是, 将所述第四阔值作为当前 输出音量值。
在这一实施例中, 当用户处于非常安静的场合时, 如果不加限制地将音量 调小, 可能会由于音量过小导致用户无法获取设备输出的声音; 或者当用户处 于非常嘈杂的场合时,如果不加限制地增大声音, 也可能会由于音量过大而导 致音量不可接受。 因此在本发明实施例中,在将所述和值作为当前输出音量值 之前,进一步包括判断所述和值是否大于设定的第三阔值以及判断所述和值是 否小于设定的第四阔值的步骤。 所述第三阔值以及第四阔值是预先设定的值, 表征了音量增益调节的范围。参见图 3 ,为本发明实施例音量调整曲线示意图。 其中, 曲线 A为最大可调节增益曲线, 曲线 C为最小可调节增益曲线, 曲线 B 为正常增益曲线。 当超出一定范围时, 即不再进行音量调节, 以使用户获取更 好的听觉效果。
需要说明的是, 对于利用公式(10 )获取当前输出音量值的情况, 在步骤 S204B之后、 步骤 S204C之前, 进一步还可以包括: 判断获取所述和值与调节 参数的乘积是否大于第三阔值,如果是,将所述第三阔值作为当前输出音量值; 判断所述和值与调节参数的乘积是否小于第四阔值,如果是,将所述第四阔值 作为当前输出音量值。 这样, 就可以使音量调节保持在合适的范围, 以使用户 获取更好的听觉效果。
在本发明实施例中, 参考音量具体可以为用户手动调节音量时的音量值。 每次用户手动调整音量, 就更新参考音量值。 更新参考音量值时, 将与所述当 前输出音量值对应的当前环境噪声能量值作为参考噪声能量值。
在本发明一个优选实施例中, 进行音量调节时,根据预设的速率将所述设 备的音量平滑调节至所述当前输出音量值。 例如, 打开喇八后, 以上一次的音 量值为起点以 6db/s或者- 6db/s的速率达到这一次需要调节的音量, 使增益 平滑变化, 以免音量突变造成用户使用起来不舒服。 图 4为本发明实施例提供 的平滑调节音量方法示意图。 在图 4中, 线 1表示调大音量的情况, 线 2表示 调小音量的情况。
S205 , 根据所述当前环境噪声的能量值对频响进行调节。
在本发明的第二实施例中,还包括了对频响的调节。正常的语音滤波器在 听觉范围内的频响是平坦的。 而人耳对高频声音相对敏感,如果在环境噪声很 大时, 提升高频, 会更容易听清说话者的内容, 但是周围安静时语音的高频部 分过多, 听起来就会比较刺耳。 为了使用户获得比较好的听觉效果, 本发明还 可以根据周围环境情况, 自适应地对频响进行调节,使用户获得更好的听觉感 具体的, 步骤 S205在实现时具体可以包括:
( 1 ) 当所述当前噪声能量值大于第一阔值时, 进行高音增强处理。
具体的, 可以预先设置第一阔值为噪声门限。 当当前噪声能量值大于第一 阔值时, 判断为很嘈杂的环境, 釆用高音增强滤波器对频响进行调节。 如图 5 所示, 为高音增强频响调节示意图。
( 2 ) 当所述当前噪声能量值小于第二阔值时, 进行低音增强处理。 具体的, 可以预先设置第二阔值为安静门限。 当当前噪声能量值小于第二 阔值时, 判断为很安静的环境, 釆用低音增强滤波器对频响进行调节。 如图 6 所示, 为低音增强频响调节示意图。 从图 5和图 6可以看出, 本发明实施例提 供的方法在对频响进行调节时, 频响曲线是平滑的, 可以获得更好的音效。
在本发明实施例中,通过釆集声音数据,对所述声音数据进行语音活动检 测, 由此判断所述声音数据是否为语音, 当判断所述声音数据不是语音时, 将 其作为当前环境噪声, 获取所述当前环境噪声的能量值, 并根据所述当前环境 噪声能量值对输出音量以及频响进行调节。由于本发明提供的方法在获取环境 噪声能量值时, 进行了语音活动检测,从而准确地将背景环境中的语音和噪声 区分开来, 有效去除了语音的干扰, 获取了真实的环境噪声能量值, 并根据环 境噪声能量值的变化自动对设备的输出音量以及频响进行调节,音效调节结果 准确, 方法实现简单。 对于需要在不同场所不断移动的用户, 最大限度的减小 环境变化对于对讲机使用上的影响, 无需用户自己来调节,使用户更能专注于 自己的工作。 并根据当前环境噪声能量值, 对音量和频响进行调节, 获取了更 好的音效, 使用户获取更清晰、 更逼真的听觉感受。 参见图 7 , 为本发明实施例自适应调节音效的设备第一实施例示意图。 所述设备具有音频输出装置, 所述设备包括:
噪声获取模块 1 00 , 用于获取当前环境噪声的能量值;
接收模块 200 , 用于接收第一触发指令;
第一调节模块 300 , 用于根据所述当前环境噪声的能量值对当前输出音量 进行调节;
第二调节模块 400 , 用于当判断所述当前环境噪声的能量值大于第一阔值 时, 进行高音增强处理; 当判断所述当前环境噪声的能量值小于第二阔值时, 进行低音增强处理。
参见图 8 , 为本发明实施例自适应调节音效的设备第二实施例示意图。 具体的, 所述第一调节模块包括差值获取单元 31 0、 和值获取单元 320、 输出音量获取模块 350 , 其中,
所述差值获取单元 31 0 用于获取当前环境噪声能量值与参考噪声能量值 的差值;
所述和值获取单元 320用于获取所述差值与参考音量值的和值; 其中, 所 述参考音量值为记录的用户设置的输出音量值,所述参考噪声能量值与所述参 考音量值具有对应关系;
所述输出音量获取单元 350用于将所述和值作为当前输出音量值,将所述 设备的音量调节至所述当前输出音量值。
具体的,在所述和值获取单元 320与所述输出音量获取单元 350之间进一 步包括:
第一判断单元 330 , 用于判断所述和值是否大于第三阔值, 如果是, 将所 述第三阔值作为当前输出音量值;
第二判断单元 340用于判断所述和值是否小于第四阔值, 如果是, 将所述 第四阈值作为当前输出音量值。
具体的, 所述设备还包括:
语音检测模块 500 , 用于釆集声音数据, 对所述声音数据进行语音活动检 测, 判断所述声音数据是否为语音。
具体的, 所述语音检测模块包括:
LPC获取单元 51 0 , 用于获取所述声音数据的 LPC谱。
第三判断单元 520 , 用于根据所述 LPC谱获取共振峰, 当所述共振峰满足 预设条件时, 判断所述声音数据为语音; 当不满足预设条件时, 判断所述声音 数据不是语音。
优选的, 所述设备为对讲机。 所述设备还可以是任何具有音频输出装置的 设备, 例如手机、 电视、 收音机等。
需要说明的是,在本文中, 诸如第一和第二等之类的关系术语仅仅用来将 一个实体或者操作与另一个实体或操作区分开来,而不一定要求或者暗示这些 实体或操作之间存在任何这种实际的关系或者顺序。 而且, 术语"包括"、 "包 含,,或者其任何其他变体意在涵盖非排他性的包含, 从而使得包括一系列要素 的过程、 方法、 物品或者设备不仅包括那些要素, 而且还包括没有明确列出的 其他要素, 或者是还包括为这种过程、 方法、 物品或者设备所固有的要素。 在 没有更多限制的情况下, 由语句 "包括一个 ... ...,,限定的要素, 并不排除在包括 所述要素的过程、 方法、 物品或者设备中还存在另外的相同要素。 本发明可以在由计算机执行的计算机可执行指令的一般上下文中描述,例 如程序模块。一般地,程序模块包括执行特定任务或实现特定抽象数据类型的 例程、 程序、 对象、 组件、 数据结构等等。 也可以在分布式计算环境中实践本 发明,在这些分布式计算环境中, 由通过通信网络而被连接的远程处理设备来 执行任务。在分布式计算环境中,程序模块可以位于包括存储设备在内的本地 和远程计算机存储介质中。
以上所述仅是本发明的具体实施方式,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, 这些改进和润饰也应视为本发明的保护范围。

Claims

权 利 要 求
1、 一种自适应调节音效的方法, 其特征在于, 所述方法应用于具有音频 输出装置的设备, 所述方法包括:
获取当前环境噪声的能量值;
接收第一触发指令,根据所述当前环境噪声的能量值对当前输出音量进行 调节;
当判断所述当前环境噪声的能量值大于第一阔值时, 进行高音增强处理; 当判断所述当前环境噪声的能量值小于第二阔值时, 进行低音增强处理。
2、 根据权利要求 1所述的方法, 其特征在于, 所述根据所述当前环境噪 声的能量值对当前输出音量进行调节包括:
获取当前环境噪声的能量值与参考噪声能量值的差值;
获取所述差值与参考音量值的和值, 将所述和值作为当前输出音量值,将 所述设备的音量调节至所述当前输出音量值; 其中, 所述参考音量值为记录的 用户设置的输出音量值, 所述参考噪声能量值与所述参考音量值具有对应关 系。
3、 根据权利要求 2所述的方法, 其特征在于, 在将所述和值作为当前输 出音量值之前, 所述方法进一步包括:
判断所述和值是否大于第三阔值,如果是,将所述第三阔值作为当前输出 音量值;
和\或
判断所述和值是否小于第四阔值,如果是,将所述第四阔值作为当前输出 音量值。
4、 根据权利要求 2所述的方法, 其特征在于, 所述方法还包括: 当用户对当前输出音量值进行调节时,将所述当前输出音量值作为参考音 量值;将与所述当前输出音量值对应的当前环境噪声能量值作为参考噪声能量 值。
5、 根据权利要求 1所述的方法, 其特征在于, 在获取当前环境噪声的能 量值之前, 所述方法还包括:
釆集声音数据,对所述声音数据进行语音活动检测, 判断所述声音数据是 否为语音;
当判断所述声音数据不是语音时, 将所述声音数据作为当前环境噪声。
6、 根据权利要求 5所述的方法, 其特征在于, 所述对所述声音数据进行 语音活动检测包括:
获取所述声音数据的 LPC谱;
根据所述 LPC谱获取共振峰,根据所述共振峰的特征判断所述共振峰是否 满足预设条件; 当满足预设条件时, 判断所述声音数据为语音; 当不满足预设 条件时, 判断所述声音数据不是语音。
7、 根据权利要求 6所述的方法, 其特征在于, 所述预设条件为: 对于连续多个数据帧, 当有至少两个共振峰在同一区间时, 判断所述声音 数据为语音; 所述区间根据所述 LPC谱划分得到;
和 /或
所述共振峰的幅值在预设区间时, 判断所述声音数据为语音。
8、 根据权利要求 1所述的方法, 其特征在于, 所述第一触发指令为: 所述设备检测到有呼叫接入或者所述设备接收到自动调节命令。
9、 一种自适应调节音效的设备, 其特征在于, 所述设备具有音频输出装 置, 所述设备包括:
噪声获取模块, 用于获取当前环境噪声的能量值;
接收模块, 用于接收第一触发指令;
第一调节模块,用于根据所述当前环境噪声的能量值对当前输出音量进行 调节;
第二调节模块, 用于当判断所述当前环境噪声的能量值大于第一阔值时, 进行高音增强处理; 当判断所述当前环境噪声的能量值小于第二阔值时, 进行 低音增强处理。
10、 根据权利要求 9所述的设备, 其特征在于, 所述第一调节模块包括差 值获取单元、 和值获取单元、 输出音量获取单元, 其中,
所述差值获取单元用于获取当前环境噪声能量值与参考噪声能量值的差 值;
所述和值获取单元用于获取所述差值与参考音量值的和值; 其中, 所述参 考音量值为记录的用户设置的输出音量值,所述参考噪声能量值与所述参考音 量值具有对应关系;
所述输出音量获取单元用于将所述和值作为当前输出音量值 ,将所述设备 的音量调节至所述当前输出音量值。
11、 根据权利要求 1 0所述的设备, 其特征在于, 在所述和值获取单元与 所述输出音量获取单元之间进一步包括:
第一判断单元, 用于判断所述和值是否大于第三阔值, 如果是, 将所述第 三阔值作为当前输出音量值;
和\或
第二判断单元用于判断所述和值是否小于第四阔值,如果是,将所述第四 阔值作为当前输出音量值。
12、 根据权利要求 9所述的设备, 其特征在于, 所述设备还包括: 语音检测模块, 用于釆集声音数据, 对所述声音数据进行语音活动检测, 判断所述声音数据是否为语音。
1 3、 根据权利要求 12所述的设备, 其特征在于, 所述语音检测模块包括:
LPC获取单元, 用于获取所述声音数据的 LPC谱;
第三判断单元, 用于根据所述 LPC谱获取共振峰, 当所述共振峰满足预设 条件时, 判断所述声音数据为语音; 当不满足预设条件时, 判断所述声音数据 不是语音。
14、 根据权利要求 9所述的设备, 其特征在于, 所述设备为对讲机。
PCT/CN2011/083373 2011-12-02 2011-12-02 一种自适应调节音效的方法和设备 WO2013078677A1 (zh)

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