CN1964185A - A control method for self-adapting signal amplitude of directional sound source - Google Patents

A control method for self-adapting signal amplitude of directional sound source Download PDF

Info

Publication number
CN1964185A
CN1964185A CN 200610070580 CN200610070580A CN1964185A CN 1964185 A CN1964185 A CN 1964185A CN 200610070580 CN200610070580 CN 200610070580 CN 200610070580 A CN200610070580 A CN 200610070580A CN 1964185 A CN1964185 A CN 1964185A
Authority
CN
China
Prior art keywords
data
amplitude
normalized
peak
data frame
Prior art date
Application number
CN 200610070580
Other languages
Chinese (zh)
Inventor
赵洪亮
朱海生
郑卫华
张丽丽
王亮
Original Assignee
山东科技大学
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 山东科技大学 filed Critical 山东科技大学
Priority to CN 200610070580 priority Critical patent/CN1964185A/en
Publication of CN1964185A publication Critical patent/CN1964185A/en

Links

Abstract

The disclosed adaptive signal amplitude control method for directional sound source comprises: normalizing the input with last frame peak as normalization standard, preprocessing for carrier modulation, hierarchical processing amplitude, and outputting the unipolar sound. This invention avoids defect in real-time led by artificial delay, and is convenient to implement in engineer as simple algorithm.

Description

用于指向性声源的自适应信号幅度控制方法 A signal amplitude of adaptive sound source directivity control method

技术领域 FIELD

本发明涉及一种指向性声源的信号控制方法。 The present invention relates to a method for controlling the signal directivity of the sound source.

背景技术 Background technique

指向性声源的基本原理为将普通音频信号经过预处理对40KHz正弦载波调制,调制信号通过超声换能器阵列形成调制超声波,该调制超声波能够在空气中利用空气的自解调效应再生可听声,同时由于超声波指向性的尾焰效应,形成指向性的可听声波。 The basic principle of the directional sound source audio signals as ordinary pretreated to 40KHz sine carrier modulation, the modulation signal by an ultrasonic transducer array form a modulated ultrasound, the ultrasonic wave can be modulated by the air in the air self-demodulated audible effect regeneration sound, and because the effect of the ultrasonic directivity tail flame forming directivity audible sound waves. 对音频输入信号进行预处理是改善指向性声源音质的有效措施。 Pre-processing the audio input signal is an effective measure to improve the quality of the directivity of the sound source. 载波调制处理是预处理中的基本环节,是后续处理的基础。 Carrier modulation process is a pretreatment of the basic part, is the basis for subsequent processing. Frank J.Pompei、James J.Croft III和Kamakura在他们的专利和文章中采取了动态载波的调制处理方法,即通过响应输入音频信号的大小,动态地调整载波幅度的大小,实现最优调制。 Frank J.Pompei, James J.Croft III Kamakura and dynamic approach taken by the carrier modulation in their patents and articles, i.e. in response to the size of the input audio signal, the amplitude of dynamically adjusting the size of the carrier to achieve optimal modulation. 这种方法的优点是在选择合适的调制指数避免过调制,改善音质的同时,提高了效率,其缺点是系统引入延时,同时算法比较复杂,计算量大,不便于工程上实现。 The advantage of this method is to select an appropriate modulation index to avoid overmodulation, improve the sound quality at the same time, improve efficiency, the drawback is the system delay is introduced, while the algorithm is relatively complex, computationally intensive, not easy to achieve in engineering.

发明内容 SUMMARY

本发明的目的在于,提供一种算法简单,易于工程实现,能够改善音质的用于指向性声源的自适应信号幅度控制方法。 Object of the present invention is to provide a simple and easy to implement engineering, signal amplitude can be improved adaptive control method for a directional sound source of sound.

本发明为实现其目的所采用的技术方案是:用于指向性声源的自适应信号幅度控制方法,包括下述步骤:1、输入归一化:用于将输入音频信号A/D转换后的数字量转化为+1~-1的范围,便于后续的信号处理;2、载波调制预处理:输入归一化的数据在其数据帧内比较幅度大小,最大的做为其数据帧的峰值,处于当前数据帧的数据在比较大小确定峰值的同时,以上一数据帧的峰值作为归一化的标准,进行再次归一化并作为输出,实现载波调制预处理。 Technical Solution To achieve the object of the present invention is used are: the signal magnitude control method for the adaptive directional sound source, comprising the steps of: 1, the normalized input: an input audio signal for A / D conversion after the digital conversion is in the range of + 1 to -1, to facilitate subsequent signal processing; 2, pre-carrier modulation: input normalized amplitude data which compare a data frame size, the maximum data frame as its peak the data in the current data frame is determined in the comparing the size of the peak at the same time, more than one data frame as peak normalized standard, again normalized and output as to achieve pre-carrier modulation. 由于数据是在再次归一化输出的同时进行大小比较确定峰值,因此在预处理过程中没有引入延时环节,实现了数据的实时处理;3、幅度分级处理:将再次归一化的数据按照幅度A0=0,0<A0<1/2,A0>1/2分为小幅度、中幅度和大幅度采取不同的处理方法,小幅度时控制载波输出为0,大幅度时线性衰减至中幅度,以中幅度的处理方法,控制调制指数在 Since the data is normalized again while a size comparison of the output to determine the peak, so there is no delay is introduced during pretreatment part, to achieve real-time processing of data; 3, an amplitude classification treatment: The normalized data again in accordance with amplitude A0 = 0,0 <A0 <1/2, A0> 1/2 into small amplitude, amplitude and substantially take a different approach, small amplitude control carrier output is 0, when the linear attenuation to substantially amplitude to the amplitude of the processing method, the control modulation index . 这种处理方法可以在保证音质的同时,避免无输入音频信号时,指向性声源却一直输出载波激励换能器阵列的情况,提高了系统效率,减少了换能器阵列的发热,降低了对换能器制作材料及工艺的耐热要求,从而降低了成本;4、单极性音频输出处理:为了后续处理中开平方运算的方便,固定指向性声源系统中超声载波电平,自适应的调整音频输入信号以达到最佳调制和高品质音质系统的输出通过施加一 When this processing method can guarantee quality while avoiding no input audio signal, the sound source directional output of the carrier has been the case of excitation of the transducer array, system efficiency is improved, reducing the heat transducer array, reducing the the heat required for the transducer materials and production processes, thereby reducing the cost; 4, the audio output processing unipolar: easy, fixed directional speaker system, an ultrasonic carrier level for the subsequent processing of the square root operation, since adjusted audio input signal adapted to modulate and optimize the output of high-quality sound systems by applying a 直流分量的方法,将双极性音频输出转化为单极性音频输出。 DC component method, bipolar unipolar audio output into an audio output.

上述载波调制预处理可采用下述方法:1、对A/D转换并输入归一化的数据取绝对值,获取幅度值。 Pretreatment of the carrier modulation methods are employed: 1, the A / D conversion and input data normalized absolute value, obtaining an amplitude value.

2、通过当前数据帧内的数据幅度值比较大小获取当前数据帧的峰值。 2, the peak current frame of data acquired by the amplitude value of the current data frame data magnitude comparison.

3、当前数据帧内的数据实时以上一数据帧的峰值为标准进行归一化输出,即当前数据除以上一数据帧的峰值作为新的输出数据。 3, the real-time data of the current data frame is a data frame or more standard peak normalized output, i.e., the current peak data other than the data frame as a new output data.

4、上述步骤3中输出数据乘以1/2,满足调制指数为1/2。 4, the output data of the above step 3 by 1/2 to meet the modulation index of 1/2.

5、峰值更新,当前数据帧的峰值保存,准备下一帧数据处理,即下一数据帧作为当前数据帧,当前数据帧作为上一数据帧重复1、2、3、4的处理步骤。 5, the peak update the current stored peak data frame, the next frame data preparation process, i.e., the next data frame as a current data frame, the current data frame as a data frame repetition 1,2,3,4 processing steps.

本发明的有益效果是:该方法在普通音频信号普遍为短时平稳信号的条件下,以上一数据帧的峰值作为当前数据帧归一化的标准,避免了延迟估计方法中普遍采用的人为加入延时以保证输入信号与系统获取的当前数据帧峰值作为载波修正量同步所造成的非实时信号处理的缺点,能够实时处理输入信号,无需额外延时,同时在载波已知的情况下,更容易获得最优的调制指数,抑制再生声波的谐波成分,保证音质,且算法简单,易于工程实现。 Advantageous effect of the invention is: This method is generally normal audio signal short-time stationary condition, above the peak data frame as a current data frame normalized standard avoid human delay estimation method commonly used is added the disadvantage of the non-real-time signal processing the current data frame time delay to ensure the peak input signal and a carrier system to obtain the correction amount as a result of synchronization, real-time processing of the input signal, no additional delay, while in the case where the carrier is known, more easy to obtain optimal modulation index, acoustic reproduction suppressed harmonic components, to ensure quality, and the algorithm is simple and easy to implement.

附图说明 BRIEF DESCRIPTION

图1是本发明所述方法的原理示意图;图2是载波调制预处理原理示意图;图3是带有自适应信号幅度控制系统的开平方预处理的指向性声源应用实例示意图。 1 is a schematic of the principles of the method of the present invention; FIG. 2 is a schematic view of the principle of pre-modulation carrier; FIG. 3 is a schematic diagram of a sound source directivity application example with the square root of the signal magnitude control system adaptive preprocessing.

具体实施方式 Detailed ways

为了易于理解本发明专利,下面将对发明内容进行具体有针对性的说明,发明内容实施的具体化并不代表本发明涉及的范围缩小,对本文中说明的发明原理的修改以及对本文中说明的发明原理的任何其他应用,都被认为在本发明的范围之内,本发明的范围由被准许的权利要求限定,而不受本文对发明内容具体实施方式说明的限制。 For easy understanding of the present patent disclosure, the following invention will be specifically targeted content description SUMMARY Embodiments embodying do not represent the scope of the present invention is reduced, modifications of the principles of the invention herein described and illustrated herein any other application of the principles of the invention, are considered within the scope of the invention, the scope of the invention as set forth in claim licensed defined herein, without limiting the content of the invention by way of illustration specific embodiments.

如图1所示,用于指向性声源的自适应信号幅度控制的实现按照下列步骤进行:1:输入归一化,用于将输入音频信号A/D转换后的数字量转化为+1~-1的范围,便于后续的信号处理。 As shown, the amplitude of the signal for the adaptive directional sound source control follow these steps to achieve 1: 1: input normalized for the amount of the input digital audio signal after A / D conversion is converted to +1 ~ -1 range, to facilitate subsequent signal processing.

2:载波调制预处理,输入归一化的数据在其数据帧内比较幅度大小,最大的做为其数据帧的峰值,处于当前数据帧的数据在比较大小确定峰值的同时,以上一数据帧的峰值作为归一化的标准,进行再次归一化并作为输出,实现载波调制预处理。 2: preprocessing carrier modulation, the input data which the normalized amplitude comparing a data frame size, as its largest peak in the data frame, the data in the current data frame is determined in the comparing the size of the peak at the same time, more than one data frame as peak normalized standard, again normalized and output as to achieve pre-carrier modulation. 由于数据是在再次归一化输出的同时进行大小比较确定峰值,因此在预处理过程中没有引入延时环节,实现了数据的实时处理。 Since the data is in a re-normalization of the size comparison output simultaneously to determine the peak, so there is no delay is introduced during pretreatment part, to achieve real-time processing of data.

3:幅度分级处理,将再次归一化的数据按照幅度A0=0,0<A0<1/2,A0>1/2分为小幅度、中幅度和大幅度采取不同的处理方法,小幅度时控制载波输出为0,大幅度时线性衰减至中幅度,以中幅度的处理方法,控制调制指数在 3: fractionated amplitude, again normalized data in accordance with the amplitude A0 = 0,0 <A0 <1/2, A0> 1/2 into small amplitude, amplitude and substantially take a different approach, small amplitude control carrier output is 0, when the amplitude is attenuated to substantially linearly to the processing method of the amplitude, controls the modulation index . 这种处理方法可以在保证音质的同时,避免无输入音频信号时,指向性声源却一直输出载波激励换能器阵列的情况,提高了系统效率,减少了换能器阵列的发热,降低了对换能器制作材料及工艺的耐热要求,从而降低了成本。 When this processing method can guarantee quality while avoiding no input audio signal, the sound source directional output of the carrier has been the case of excitation of the transducer array, system efficiency is improved, reducing the heat transducer array, reducing the the heat required for the transducer materials and production processes, thereby reducing costs.

4:单极性音频输出处理,为了后续处理中开平方运算的方便,固定指向性声源系统中超声载波电平,自适应的调整音频输入信号以达到最佳调制和高品质音质系统的输出通过施加一 4: unipolar audio output processing, for subsequent processing Square root of convenience, directional speaker system fixed ultrasonic carrier level, adaptive adjustment of the audio input signal in order to achieve optimum modulation and output high-quality sound systems by applying a 直流分量的方法,将双极性音频输出转化为单极性音频输出。 DC component method, bipolar unipolar audio output into an audio output.

每个步骤的具体实现如下:输入归一化:根据A/D转换器的位数(n),2n-1即为归一化的比较标准。 Specific realization of each step is as follows: normalized input: the number of bits A / D converter (n), 2n-1 is the normalized comparison standard. 例如为14位的A/D转换器,2n-1=8192则归一化的标准为8192。 For example, a 14-bit A / D converter, 2n-1 = 8192 reverts to a standard of 8192. 假设当前输入信号为VIN=3V,该A/D转换器的输入范围为V+=5V~V-=-5V,那么该信号归一化后的数据为:(VINV+&times;2n-1)/2n-1=(3V5V&times;214-1)/214-1=0.6]]> Suppose the current input signal VIN = 3V, an input range of the A / D converter is V + = 5V ~ V - = - 5V, then the data after the signal normalization is: (VINV + & times; 2n-1) / 2n-1 = (3V5V & times; 214-1) /214-1=0.6]]>

载波调制预处理:载波调制预处理的过程示意图如附图2所示,输入数据x(n),已归一化为±1的范围,经过以下三方面的处理。 Carrier modulation Pretreatment: Pretreatment of carrier modulation process shown in Figure 2 as a schematic diagram, input data x (n), is normalized to ± 1 range, through the following three processes.

1:x(n)经过取绝对值环节,获得|x(n)|,即x(n)的幅度。 1: x (n) through an absolute value, reaping | x (n) |, i.e., the amplitude of x (n) is. temppeak=previouspeak其中temppeak初始值为0。 temppeak = previouspeak wherein temppeak initial value 0. |x(n)|与temppeak比较大小,若|x(n)|>temppeak,则temppeak=|x(n)|;若|x(n)|≤temppeak,则temppeak不变。 | X (n) | temppeak comparison with the size, if | x (n) |> temppeak, the temppeak = | x (n) |; if | x (n) | ≤temppeak, the temppeak unchanged. n=0,1,2……N-1,N为数据帧的长度。 n = 0,1,2 ...... N-1, N is the length of the data frame. 将temppeak保存至currentpeak,作为当前数据帧的峰值。 Save temppeak to currentpeak, as the peak of the current data frame. 当前数据帧处理完毕,previouspeak=currentpeak,即当前数据帧的峰值赋值给上一数据帧的峰值,接下来的数据帧作为当前数据帧,重复上述的过程,开始新一帧数据的处理。 The current data frame has been processed, previouspeak = currentpeak, i.e., the peak current data frame is assigned to the peak of a data frame, the next data frame as a current data frame, repeating the above procedure, processing of the new frame data. 其中temppeak为临时峰值存储;currentpeak为当前数据帧峰值,对应于图2中的p0;previouspeak为上一数据帧峰值,对应于图2中的p1。 Wherein temppeak temporary storage peak; currentpeak peak of the current data frame, corresponding to FIG. 2 p0; previouspeak a data frame to the peak p1 corresponding to 2 in FIG.

2:在进行1中处理的同时,x(n)进行过调制归一化的处理过程。 2: 1 while the process is performed, x (n) for normalization of the overmodulation process. 特别强调的是1、2的处理是同时进行的,这是区别于其它类似方法的显著特点。 Particular emphasis is processing 1 is performed simultaneously, which is different from other similar methods notable features. 即x(n)在进行峰值获取的同时,进行了过调制归一化的处理,处理是实时的。 Meanwhile i.e. x (n) acquired during the peak, were treated normalized modulation, the process in real time.

过调制归一化的基本原理为:以输入音频信号的N个采样数据为一帧数据,即x(n),n=0,1,2……N-1。 The basic principle of the overmodulation normalized as: input audio signal to the N sample data is a data, i.e., x (n), n = 0,1,2 ...... N-1. 则有x(n+N),n=0,1,2……N-1为紧接x(n)的数据帧。 There is x (n + N), n = 0,1,2 ...... N-1 immediately following x (n) data frame. 以x(n),n=0,1,2……N-1,作为当前数据帧,则x(n+N),n=0,1,2……N-1为上一数据帧。 To x (n), n = 0,1,2 ...... N-1, as the current data frame, then x (n + N), n = 0,1,2 ...... N-1 is the previous data frame.

currentpeak=|x(n)|maxn=0,1,2……N-1previouspeak=|x(n+N)|maxn=0,1,2……N-1x(n)previouspeak=x0(n)]]>n=0,1,2……N-1x0(n)即为过调制归一化后的输出。 currentpeak = | x (n) | maxn = 0,1,2 ...... N-1previouspeak = | x (n + N) | maxn = 0,1,2 ...... N-1x (n) previouspeak = x0 (n) ]]> n = 0,1,2 ...... N-1x0 (n) is the over-modulation of a normalized outputs. 常见音频信号为短时平稳信号,以该信号上一数据帧的峰值为标准对当前数据帧进行归一化,即以能够获得较好调制参数的前一段信号幅度的最大值为单位1,当前段信号以其为标准归一化到±1的范围,显而易见当前段信号也能获得较好的调制。 Common short audio signal is a stationary signal, a signal to the data frame standard peak current frame of data is normalized, i.e. the maximum value of the previous paragraph is possible to obtain a better signal amplitude modulation parameter units 1, the current its standard normalized signal segment a range of ± 1 to the apparent current signal segment better modulation can be obtained.

3:为了保证调制指数m&le;12,]]>out0(n)=12&times;x0(n).]]>幅度分级处理:若|x0(n)|max=0,则采取小幅度处理,控制单位载波幅度C0=0;若0<|x0(n)|max<1,则采取中幅度处理,控制单位载波幅度C0=1;若|x0(n)|max≥1,则采取大幅度处理,按照调制指数为 3: To ensure the modulation index m & le; 12,]]> out0 (n) = 12 & times; x0 (n)]]> amplitude classification treatment: if | x0 (n) | max = 0, then take a small amplitude processing, the control unit carrier amplitude C0 = 0; if 0 <| x0 (n) | max <1, the amplitude is taken in the processing, the control unit carrier amplitude C0 = 1; if | x0 (n) | max≥1, the process to take a substantial according to the modulation index 线性衰减至0<|x0(n)|max<1,控制单位载波幅度C0=1。 Linear attenuation to 0 <| x0 (n) | max <1, the control unit carrier amplitude C0 = 1. 其中n=0,1,2……N-1。 Where n = 0,1,2 ...... N-1.

单极性音频处理:施加一 Unipolar audio processing: applying a 的直流量将归一化的双极性音频信号转化为单极性音频输出信号,即sout(n)=out1(n)+12.]]>图3所示的为本发明在采用开平方预处理方法的指向性声源中的应用实例示意图,其具体实现方法如下:音频输入信号经过自适应信号幅度控制,按照上述的具体步骤实现载波的调制预处理;低通滤波用于自适应信号幅度控制处理后D/A转换器输出的低通滤波;开平方是为了补偿空气自解调模型中的平方环节,即采用预失真信号来补偿空气自解调模型中平方环节带来的信号畸变,达到降低失真的目的;40KHz载波为40KHz幅度范围为±1的正弦波,通过乘法器实现信号对载波的幅度调制;功放用于指向性声源输出信号的功率放大,要求其正常工作频率范围涵盖40KHz;换能器阵列采用单个换能器紧密排列而成,用于发射调制超声波,调制超声波在空气中进行自解调,便形成具有 Linear flow rate normalized bipolar signal into a unipolar audio output audio signal, i.e., as shown in sout (n) = out1 (n) +12.]]> The present invention in FIG. 3, an open square application examples directional speaker pretreatment process in a schematic, the specific method is as follows: an audio input signal through the adaptive amplitude control signal, according to the above specific pretreatment steps to achieve modulation of a carrier; for the adaptive low pass filtered signal controlling the amplitude of the low pass filter process after the D / a converter output; open square is to compensate for the air from the demodulated square segment model, i.e., predistortion signal to compensate for signal distortion model squared demodulated from the air link to bring , to reduce object distortion; 40KHz to 40KHz carrier amplitude range of ± 1 of a sine wave, the amplitude modulated carrier signal is implemented by a multiplier; a power amplifier for the output signal of the sound source directional amplifier, which requires normal operating frequency range covering 40KHz; transducer array using a single transducer closely arrayed, for emitting ultrasonic wave modulation, the modulated ultrasound self-demodulation in the air, form a 向性的声波。 Directional sound waves.

Claims (2)

1.用于指向性声源的自适应信号幅度控制方法,其特征在于包括下述步骤:(1)输入归一化:输入音频信号经A/D采集的数字量转化为+1~-1的范围;(2)载波调制预处理:在调制超声载波之前对1中归一化的数据进行预处理,其原理为响应不断变化的音频输入数据,固定超声载波电平,自适应调整调制指数,以达到最佳调制;(3)幅度分级处理:将再次归一化的数据按照幅度A0=0,0<A0<1/2,A0>1/2分为小幅度、中幅度和大幅度采取不同的处理方法,小幅度时控制载波输出为0,大幅度时线性衰减至中幅度,以中幅度的处理方法,控制调制指数在 1. The signal amplitude for the adaptive sound source directivity control method comprising the steps of: (1) a normalized input: the input digital audio signal by the A / D conversion is acquired +1 to -1 range; (2) carrier modulation pretreatment: 1 for the normalized data prior to modulation ultrasonic pretreated carrier, the principle of response to changing input audio data, fixed ultrasound carrier level, modulation index adaptively , to achieve the best modulation; (3) an amplitude classification treatment: the normalized data in accordance with the amplitude A0 = 0,0 <A0 <1/2, A0> 1/2 again divided into small amplitude, and amplitude substantially take a different approach, a small amplitude control carrier output is 0, while the linear decay to a significant amplitude to the amplitude of the processing method, the control modulation index (4)单极性音频输出处理,通过施加一1/2的直流量将归一化的双极性音频信号转化为单极性音频输信号。 (4) unipolar audio output processing, by applying a direct current 1/2 normalized bipolar signal into a unipolar audio input audio signal.
2.根据权利要求1所述的用于指向性声源的自适应信号幅度控制方法,其特征在于所述载波调制预处理可采用以下步骤实现:(1)对A/D转换并输入归一化的数据取绝对值,获取幅度值;(2)通过当前数据帧内的数据幅度值比较大小获取当前数据帧的峰值。 The signal magnitude control method for the adaptive directional sound source according to claim 1, wherein said carrier modulation pretreatment steps can be implemented: (1) the A / D converter and a normalized input taking the absolute value of the data, obtaining an amplitude value; (2) by comparing the current data frame size of the data peak amplitude value obtaining current data frame. (3)对当前数据帧内的数据以上一数据帧的峰值为标准进行实时归一化处理,即当前数据除以上一数据帧的峰值作为新的输出数据。 (3) data of the current data frame over a data frame in real time peak normalized to the standard, i.e., the current peak data other than the data frame as a new output data. (4)上述(3)中输出数据乘以1/2,满足最佳调制指数为1/2。 (4) above (3) in the output data by 1/2, 1/2 satisfies the optimal modulation index. (5)峰值更新,保存当前数据帧的峰值,准备下一帧数据处理。 (5) update the peak, the peak current data frame is stored, ready to process the next frame data. (6)重复1、2、3、4的处理步骤。 (6) repeating the processing steps 1, 2,.
CN 200610070580 2006-12-05 2006-12-05 A control method for self-adapting signal amplitude of directional sound source CN1964185A (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
CN 200610070580 CN1964185A (en) 2006-12-05 2006-12-05 A control method for self-adapting signal amplitude of directional sound source

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
CN 200610070580 CN1964185A (en) 2006-12-05 2006-12-05 A control method for self-adapting signal amplitude of directional sound source

Publications (1)

Publication Number Publication Date
CN1964185A true CN1964185A (en) 2007-05-16

Family

ID=38083123

Family Applications (1)

Application Number Title Priority Date Filing Date
CN 200610070580 CN1964185A (en) 2006-12-05 2006-12-05 A control method for self-adapting signal amplitude of directional sound source

Country Status (1)

Country Link
CN (1) CN1964185A (en)

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN101743706B (en) 2007-06-04 2013-05-01 艾比奎蒂数字公司 Method and apparatus for implementing a digital signal quality metric
CN107645694A (en) * 2017-08-29 2018-01-30 山东科技大学 A kind of orientation acoustic emission apparatus and method for bird repellent

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN101743706B (en) 2007-06-04 2013-05-01 艾比奎蒂数字公司 Method and apparatus for implementing a digital signal quality metric
CN107645694A (en) * 2017-08-29 2018-01-30 山东科技大学 A kind of orientation acoustic emission apparatus and method for bird repellent
CN107645694B (en) * 2017-08-29 2018-09-21 山东科技大学 A kind of orientation acoustic emission apparatus and method for bird repellent

Similar Documents

Publication Publication Date Title
CN101903941B (en) Noise cancellation system with lower rate emulation
JP2007011330A (en) System for adaptive enhancement of speech signal
CN1149536C (en) Noise suppressor
CN103460716B (en) For the method and apparatus of Audio Signal Processing
EP1865494B1 (en) Engine sound processing device
US8005233B2 (en) Bass enhancement for audio
JP5431369B2 (en) System and method for reducing power consumption for audio playback
EP1720249A1 (en) Audio enhancement system and method
BRPI0518278B1 (en) Method and apparatus for controling a particular sound feature of an audio signal
US7394908B2 (en) Apparatus and method for generating harmonics in an audio signal
EP1210845A1 (en) Modulator processing for a parametric speaker system
WO2003019846A2 (en) Dynamic carrier system for parametric arrays
WO2000045379A3 (en) Enhancing perceptual performance of sbr and related hfr coding methods by adaptive noise-floor addition and noise substitution limiting
JP2007143157A (en) Superdirectional speaker system and signal processing method
US9741333B2 (en) Noise cancellation system
CN101006747A (en) Low frequency phase matching for microphones
CN1640190A (en) Dynamic range compression using digital frequency warping
EP1480494B1 (en) Feedback suppression in sound signal processing using frequency translation
CN1689295A (en) Broadband predistortion linearization method and system
US20020173950A1 (en) Circuit for improving the intelligibility of audio signals containing speech
US8229135B2 (en) Audio enhancement method and system
CN104012001A (en) Bass enhancement system
JP4245060B2 (en) Sound masking system, masking sound generation method and program
JP2792853B2 (en) Transmission method and device for audio signals
CN104081664B (en) Method and apparatus for having the circuit of low IC power consumption and HDR

Legal Events

Date Code Title Description
C06 Publication
C10 Request of examination as to substance
C12 Rejection of an application for a patent