TW201816777A - Electronic device and multi-frequency filter gain optimization method thereof - Google Patents
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本發明係有關於助聽器,特別是有關於一種電子裝置及其分頻濾波增益優化方法。 The present invention relates to hearing aids, and more particularly to an electronic device and its frequency division filter gain optimization method.
寬動態範圍壓縮(WDRC)的技術廣泛在助聽器的範圍被使用。經過長時間研究發現,啟動時間大約5ms能符合使用者需求,但是恢復時間隨著環境不同而所改變。第1圖係繪示進行寬動態範圍壓縮以轉換輸入音訊信號之聽力補償曲線的示意圖。曲線110(虛線部份)是指未經處理的輸入音訊信號之轉換曲線,即輸入音訊信號等於輸出音訊信號。曲線520(實線部份)是指輸入音訊信號經過寬動態範圍壓縮之處理的轉換曲線,且可依據輸入音訊信號之強弱而分為四個區域131~134。音訊信號之強度通常可用dB SPL(sound pressure level,聲壓程度)來表示。區域131係指高線性(high linear)區(例如大於90dB SPL),意即聽障人士的飽和聲壓與正常人一樣,不需放大。區域132係指壓縮(compression)區(例如介於55~90dB SPL),用以調節使用者聽域的動態範圍。區域133係指低線性(low linear)區(例如介於40~55db SPL),用以幫助聽障人士將 微弱的語音聲音放大。區域134係指擴充(expansion)區(例如小於40dB SPL),在此區域中之音訊信號的強度相當弱,輸入音訊信號可能為比語音聲音信號還小的噪音,不需放大太多。此外,在助聽器之輸出端亦會有一個音量限制器,用以限制輸出音訊信號的最大音量,例如限制於110dB SPL以內。 Wide dynamic range compression (WDRC) technology is widely used in the range of hearing aids. After a long period of research, it is found that the startup time is about 5ms to meet the user's needs, but the recovery time varies with the environment. Figure 1 is a schematic diagram showing the hearing compensation curve for wide dynamic range compression to convert the input audio signal. Curve 110 (the dotted line portion) refers to the conversion curve of the unprocessed input audio signal, that is, the input audio signal is equal to the output audio signal. The curve 520 (solid line part) refers to a conversion curve of the input audio signal subjected to wide dynamic range compression, and can be divided into four regions 131 to 134 according to the strength of the input audio signal. The intensity of the audio signal can usually be expressed in terms of dB SPL (sound pressure level). Region 131 refers to a high linear region (eg, greater than 90 dB SPL), meaning that the saturated sound pressure of a hearing impaired person is the same as that of a normal person, without amplification. Region 132 refers to a compression zone (eg, between 55 and 90 dB SPL) to adjust the dynamic range of the user's listening domain. Area 133 refers to the low linear area (eg, 40 to 55 db SPL) to help the hearing impaired to amplify the faint voice. Region 134 refers to an expansion region (e.g., less than 40 dB SPL) in which the intensity of the audio signal is rather weak, and the input audio signal may be less loud than the speech sound signal, without too much amplification. In addition, there is a volume limiter at the output of the hearing aid to limit the maximum volume of the output audio signal, for example, limited to 110dB SPL.
一般而言,聽障人士在配戴助聽器時,均會針對聽障人士的聽力衰減曲線對各自不同頻率進行增益補償。因為輸入聲音信號之各頻率有不同的增益,若將輸入音訊信號之畫分為不同頻帶(band)的數量過多,則每個頻帶的範圍均相對較小,例如可經過傅立葉轉換將輸入音訊信號從時域(time domain)轉換至頻域(frequency domain),此時可針對個別的頻率調整相應的增益,但相對地,傅立葉轉換的計算量非常大,也會造成助聽器中之音訊處理電路相當大的負擔。 In general, when hearing-impaired people wear hearing aids, they will compensate for the different frequencies of hearing-impaired hearing impairment curves. Since the frequencies of the input sound signals have different gains, if the picture of the input audio signal is divided into different numbers of bands, the range of each band is relatively small, for example, the input audio signal can be converted by Fourier transform. Switching from the time domain to the frequency domain, the corresponding gain can be adjusted for individual frequencies, but the calculation of the Fourier transform is relatively large, and the audio processing circuit in the hearing aid is equivalent. Big burden.
此外,除了助聽器之外,聽障人士亦有使用可攜式電子裝置(例如是智慧型手機及平板電腦)之需求,且在使用可攜式電子裝置時並未配戴助聽器。因為可攜式電子裝置之揚聲器的輸出特性並非針對聽障人士所設計,若在可攜式電子裝置上使用在助聽器上所使用的WDRC方法,則往往會在高頻部份(例如大於4KHz)的聲音在揚聲器輸出時會產生嘯叫聲,進而影響聽障人士在可攜式電子裝置上的使用者體驗。 In addition to hearing aids, hearing impaired people also have the need to use portable electronic devices (such as smart phones and tablets) and do not wear hearing aids when using portable electronic devices. Because the output characteristics of the speaker of the portable electronic device are not designed for the hearing impaired, if the WDRC method used on the hearing aid is used on the portable electronic device, it is often in the high frequency part (for example, greater than 4KHz). The sound of the sound will produce a howling sound when the speaker is output, which will affect the user experience of the hearing impaired person on the portable electronic device.
因此,需要一種電子裝置及其分頻濾波增益優化方法制方法以解決上述問題。 Therefore, there is a need for an electronic device and a method of dividing the frequency gain optimization method thereof to solve the above problems.
本發明係提供一種電子裝置,包括:一音訊輸入級,用以接收一輸入音訊信號,並將該輸入音訊信號轉換為一輸入數位信號;一音訊處理電路,用以對該輸入電性信號執行一分頻濾波增益優化方法以產生一輸出數位信號;以及一音訊輸出級,用以將該輸出數位信號轉換為一輸出音訊信號並於該電子裝置之一揚聲器播放該輸出音訊信號,其中該分頻濾波增益優化方法係包括:取得一使用者之一聽力衰減曲線;計算該聽力衰減曲線相應的複數個適配頻率增益;對該輸入數位信號套用一分段帶通濾波器,其中該分段帶通濾波器包括用於不同頻帶之複數個帶通濾波器;計算該分段帶通濾波器所相應的一適配頻率關係矩陣;依據該複數個適配頻率增益及該適配頻率關係矩陣,計算各帶通濾波器之一增益;依據各帶通濾波器之該增益的相位以計算各帶濾波器之一濾波器特性及一補償增益;依據各帶通濾波器之該濾波器特性及該補償增益計算相應的一輸出信號;以及將各帶通濾波器所相應的該輸出信號合成為該輸出音訊信號。 The present invention provides an electronic device comprising: an audio input stage for receiving an input audio signal and converting the input audio signal into an input digital signal; an audio processing circuit for performing the input electrical signal a frequency division filter gain optimization method for generating an output digital signal; and an audio output stage for converting the output digital signal into an output audio signal and playing the output audio signal on a speaker of the electronic device, wherein the The frequency filter gain optimization method comprises: obtaining a hearing attenuation curve of a user; calculating a plurality of adapted frequency gains corresponding to the hearing attenuation curve; applying a segmented band pass filter to the input digital signal, wherein the segmentation The band pass filter includes a plurality of band pass filters for different frequency bands; calculating an adapted frequency relationship matrix corresponding to the segment band pass filter; and the plurality of adapted frequency gains and the adapted frequency relationship matrix Calculating a gain of each band pass filter; calculating the phase of each band filter according to the phase of the gain of each band pass filter A compensating gain and filter characteristics; calculating a respective filter output signal according to the characteristics of the respective band-pass filter and the compensating gain; and each of the band pass filter corresponding to the output signals of the audio signal for output.
本發明更提供一種分頻濾波增益優化方法,用於一電子裝置,其中該電子裝置包括一音訊輸入級、一音訊處理電路、及一音訊輸出級,該方法包括:該用該音訊輸入級接收一輸入音訊信號,並將該輸入音訊信號轉換為一輸入數位信號;取得一使用者之一聽力衰減曲線;計算該聽力衰減曲線相應的複數個適配頻率增益;對該輸入數位信號套用一分段帶通濾波器,其中該分段帶通濾波器包括用於不同頻帶之複數個帶通濾波器;計算該分段帶通濾波器所相應的一適配頻率關係矩 陣;依據該複數個適配頻率增益及該適配頻率關係矩陣,計算各帶通濾波器之一增益;依據各帶通濾波器之該增益的相位以計算各帶濾波器之一濾波器特性及一補償增益;依據各帶通濾波器之該濾波器特性及該補償增益計算相應的一輸出信號;將各帶通濾波器所相應的該輸出信號合成為一輸出音訊信號;以及利用該音訊輸出級播放該輸出音訊信號。 The present invention further provides a frequency division filter gain optimization method for an electronic device, wherein the electronic device includes an audio input stage, an audio processing circuit, and an audio output stage, the method comprising: receiving the audio input stage Inputting an audio signal, and converting the input audio signal into an input digital signal; obtaining a hearing attenuation curve of a user; calculating a plurality of adaptive frequency gains corresponding to the hearing attenuation curve; applying one point to the input digital signal a segment bandpass filter, wherein the segmented bandpass filter comprises a plurality of bandpass filters for different frequency bands; calculating an adapted frequency relationship matrix corresponding to the segmented bandpass filter; The frequency gain and the adapted frequency relationship matrix are used to calculate a gain of each band pass filter; the phase of the gain of each band pass filter is used to calculate a filter characteristic of each band filter and a compensation gain; The filter characteristic of the band pass filter and the compensation gain calculate a corresponding output signal; the corresponding output signals of the respective band pass filters are combined An output audio signal is used; and the output audio signal is played by the audio output stage.
本發明更提供一種分頻濾波增益優化方法,用於一電子裝置,其中該電子裝置包括一音訊輸入級、一音訊處理電路、及一音訊輸出級,該方法包括:該用該音訊輸入級接收一輸入音訊信號,並將該輸入音訊信號轉換為一輸入數位信號;取得一使用者之一聽力衰減曲線;計算該聽力衰減曲線相應的複數個適配頻率增益;對一輸入數位信號套用一分段帶通濾波器,其中該分段帶通濾波器包括一高頻帶通濾波器及複數個帶通濾波器;計算該複數個帶通濾波器所相應的一適配頻率關係矩陣;將該高頻帶通濾波器之一增益設定為一預設值;依據該複數個適配頻率增益及該適配頻率關係矩陣,計算各帶通濾波器之一增益;依據各帶通濾波器之該增益的相位以計算各帶濾波器之一濾波器特性及一補償增益;依據各帶通濾波器之該濾波器特性及該補償增益計算相應的一輸出信號;將各帶通濾波器及該高頻帶通濾波器所相應的該輸出信號合成為一輸出音訊信號;以及利用該音訊輸出級播放該輸出音訊信號。 The present invention further provides a frequency division filter gain optimization method for an electronic device, wherein the electronic device includes an audio input stage, an audio processing circuit, and an audio output stage, the method comprising: receiving the audio input stage Inputting an audio signal, and converting the input audio signal into an input digital signal; obtaining a hearing attenuation curve of a user; calculating a plurality of adaptive frequency gains corresponding to the hearing attenuation curve; applying one point to an input digital signal a segment bandpass filter, wherein the segmented bandpass filter comprises a high frequency band pass filter and a plurality of band pass filters; calculating an adapted frequency relationship matrix corresponding to the plurality of band pass filters; One of the band pass filters is set to a predetermined value; a gain of each of the band pass filters is calculated according to the plurality of adapted frequency gains and the adapted frequency relationship matrix; and the gain is determined according to each band pass filter Phase to calculate a filter characteristic of each band filter and a compensation gain; calculate corresponding to the filter characteristics of the band pass filters and the compensation gain An output signal; each of the band-pass filter and the high-frequency band-pass filter corresponding to the output signals into an output audio signal; and playback using the audio output stage of the output audio signal.
110、120‧‧‧曲線 110, 120‧‧‧ Curve
131-134‧‧‧區域 131-134‧‧‧Area
200‧‧‧電子裝置 200‧‧‧Electronic devices
210‧‧‧音訊輸入級 210‧‧‧Optical input stage
211‧‧‧麥克風 211‧‧‧ microphone
212‧‧‧類比數位轉換器 212‧‧‧ Analog Digital Converter
220‧‧‧音訊處理電路 220‧‧‧Operation Processing Circuit
230‧‧‧音訊輸出級 230‧‧‧ audio output stage
231‧‧‧接收器 231‧‧‧ Receiver
232‧‧‧數位類比轉換器 232‧‧‧Digital Analog Converter
10‧‧‧輸入音訊信號 10‧‧‧ Input audio signal
11‧‧‧輸入電性信號 11‧‧‧ Input electrical signal
12‧‧‧輸入數位信號 12‧‧‧Input digital signal
14‧‧‧輸出數位信號 14‧‧‧Output digital signal
15‧‧‧輸出電性信號 15‧‧‧ Output electrical signal
16‧‧‧輸出音訊信號 16‧‧‧ Output audio signal
311-314、411-414‧‧‧曲線 311-314, 411-414‧‧‧ Curve
510-570、710-770、611-614‧‧‧方塊 510-570, 710-770, 611-614‧‧‧
第1圖係繪示進行寬動態範圍壓縮以轉換輸入音訊信號之聽力補償曲線的示意圖。 Figure 1 is a schematic diagram showing the hearing compensation curve for wide dynamic range compression to convert the input audio signal.
第2圖係顯示依據本發明一實施例中之助聽器的方塊圖。 Figure 2 is a block diagram showing a hearing aid in accordance with an embodiment of the present invention.
第3A及3B圖係顯示不同帶通濾波器之分布的示意圖。 Figures 3A and 3B show schematic diagrams of the distribution of different bandpass filters.
第4A及4B圖係顯示依據本發明一實施例中之不同帶通濾波器之分布的示意圖。 4A and 4B are diagrams showing the distribution of different band pass filters in accordance with an embodiment of the present invention.
第5圖係顯示依據本發明一實施例中之分頻濾波增益優化方法的流程圖。 Figure 5 is a flow chart showing a frequency division filter gain optimization method in accordance with an embodiment of the present invention.
第6圖係顯示依據本發明一實施例中輸入音訊信號經過各個帶通濾波器分別處理以合成輸出音訊信號之流程的示意圖。 Figure 6 is a diagram showing the flow of input audio signals processed by respective band pass filters to synthesize an output audio signal in accordance with an embodiment of the present invention.
第7圖係顯示依據本發明另一實施例中之分頻濾波增益優化方法的流程圖。 Figure 7 is a flow chart showing a frequency division filter gain optimization method in accordance with another embodiment of the present invention.
為使本發明之上述目的、特徵和優點能更明顯易懂,下文特舉一較佳實施例,並配合所附圖式,作詳細說明如下。 The above described objects, features and advantages of the present invention will become more apparent from the description of the appended claims.
第2圖係顯示依據本發明一實施例中之電子裝置200的方塊圖。在一實施例中,電子裝置200可為一智慧型手機、一平板電腦、或一可攜式電子裝置,但本發明並不以此為限。電子裝置200包括一音訊輸入級210、一音訊處理電路220、以及一音訊輸出級230。音訊輸入級210係包括一麥克風211及一類比數位轉換器(analog-to-digital converter,ADC)212。麥 克風211係用以接收一輸入音訊信號10(例如是一類比音訊信號),並該將該輸入音訊信號10轉換為一輸入電性信號11,類比數位轉換器112係將該輸入電性信號11轉換為一輸入數位信號12做為音訊處理電路220之輸入。 2 is a block diagram showing an electronic device 200 in accordance with an embodiment of the present invention. In an embodiment, the electronic device 200 can be a smart phone, a tablet computer, or a portable electronic device, but the invention is not limited thereto. The electronic device 200 includes an audio input stage 210, an audio processing circuit 220, and an audio output stage 230. The audio input stage 210 includes a microphone 211 and an analog-to-digital converter (ADC) 212. The microphone 211 is configured to receive an input audio signal 10 (for example, an analog audio signal), and convert the input audio signal 10 into an input electrical signal 11, and the analog digital converter 112 is configured to input the electrical signal 11 The input is converted to an input digital signal 12 as an input to the audio processing circuit 220.
音訊處理電路220係對該輸入數位信號12進行一分頻濾波增益優化方法及/或寬動態範圍壓縮處理以產生一輸出數位信號14。其中分頻濾波增益優化方法之細節將詳述於後。需了解的是上述寬動態範圍壓縮處理中包括了一預定寬動態範圍壓縮轉換曲線,其係針對各使用者之聽力特性之不同,預先進行各種聽量及頻率的聽力測量,進而獲得個別的寬動態範圍壓縮轉換曲線。此外,在輸入音訊信號之聲音強度產生變化時,音訊處理電路220亦會對電子裝置200之恢復時間進行相應的調整,進而讓聽障人士有更佳的使用者體驗。在一些實施例中,音訊處理電路220可以是一微控制器(microcontroller)、一處理器、一數位信號處理器(DSP)、或是應用導向之積體電路(ASIC),但本發明並不限於此。 The audio processing circuit 220 performs a frequency division filter gain optimization method and/or a wide dynamic range compression process on the input digital signal 12 to generate an output digital signal 14. The details of the crossover filter gain optimization method will be detailed later. It should be understood that the above wide dynamic range compression processing includes a predetermined wide dynamic range compression conversion curve, which is performed on various hearing and frequency hearing measurements in advance for each user's hearing characteristics, thereby obtaining individual widths. Dynamic range compression conversion curve. In addition, when the sound intensity of the input audio signal changes, the audio processing circuit 220 also adjusts the recovery time of the electronic device 200, thereby providing a better user experience for the hearing impaired. In some embodiments, the audio processing circuit 220 can be a microcontroller, a processor, a digital signal processor (DSP), or an application-oriented integrated circuit (ASIC), but the present invention is not Limited to this.
更進一步而言,音訊處理電路220在進行寬動態範圍壓縮時,會參考該輸入音訊信號相關的恢復時間因子以調整輸出音訊信號的延遲(即恢復時間)。音訊輸出級230例如包括一揚聲器231及一數位類比轉換器232。數位類比轉換器232係用以將音訊處理電路220所產生之輸出數位信號14轉換為輸出電性信號15。揚聲器231則可將輸出電性信號15轉換為輸出音訊信號16(例如是一類比音訊信號)並進行播放以供使用者聽取輸出音訊信號16。為了便於說明,在下面實施例中,均省略將音 訊信號與電性信號之間的轉換,而僅使用輸入音訊信號及輸出音訊信號進行說明。 Furthermore, when performing wide dynamic range compression, the audio processing circuit 220 refers to the recovery time factor associated with the input audio signal to adjust the delay (ie, recovery time) of the output audio signal. The audio output stage 230 includes, for example, a speaker 231 and a digital analog converter 232. The digital analog converter 232 is operative to convert the output digital signal 14 produced by the audio processing circuit 220 into an output electrical signal 15. The speaker 231 can convert the output electrical signal 15 into an output audio signal 16 (eg, an analog audio signal) and play it for the user to listen to the output audio signal 16. For convenience of explanation, in the following embodiments, the conversion between the audio signal and the electrical signal is omitted, and only the input audio signal and the output audio signal are used for explanation.
需注意的是,本發明之分頻濾波增益優化方法係可讓聽障人士利用其電子裝置(例如智慧型手機或平板電腦)聽取音訊信號時可達到使用助聽器之效果。然而,電子裝置中所配備的揚聲器往往都是全頻的,意即會將各種頻率的音訊信號都放大。相對地,助聽器中的接收器,其設計通常不會放大高頻(例如4KHz以上)的音訊信號。因此,若使用在助聽器中所使用的寬動態範圍壓縮處理在電子裝置上,則在電子裝置上的揚聲器很容易產生嘯叫聲,會降低聽障人士的使用者體驗。 It should be noted that the frequency-divided filter gain optimization method of the present invention can enable a hearing-impaired person to use an electronic device (such as a smart phone or a tablet) to listen to an audio signal to achieve the effect of using a hearing aid. However, the speakers provided in the electronic device are often full-frequency, which means that the audio signals of various frequencies are amplified. In contrast, receivers in hearing aids are typically designed to not amplify high frequency (eg, above 4 KHz) audio signals. Therefore, if the wide dynamic range compression process used in the hearing aid is used on the electronic device, the speaker on the electronic device can easily generate howling, which reduces the user experience of the hearing impaired.
第3A及3B圖係顯示不同帶通濾波器之分布的示意圖。舉例來說,傳統在使用時域的帶通濾波器時,會針對不同的頻帶範圍設置相應的帶通濾波器,如第3A圖中之用於低頻帶的帶通濾波器310及用於高頻帶的帶通濾波器311,以及第3B圖中用於低頻帶的帶通濾波器312及用於高頻帶的帶通濾波器313所示。然而,每個頻帶的中間頻率都必需維持相同的增益。然而,在高頻有增益時,其在不同頻帶之間的交界地帶的不連續性較為嚴重。 Figures 3A and 3B show schematic diagrams of the distribution of different bandpass filters. For example, when using a bandpass filter in the time domain, a corresponding bandpass filter is set for different frequency band ranges, such as the bandpass filter 310 for the low frequency band in FIG. 3A and for high The band pass filter 311 of the band, and the band pass filter 312 for the low band and the band pass filter 313 for the high band in Fig. 3B are shown. However, the intermediate frequency of each band must maintain the same gain. However, when there is gain at high frequencies, its discontinuity in the boundary between different frequency bands is more serious.
第4A及4B圖係顯示依據本發明一實施例中之不同帶通濾波器之分布的示意圖。在一實施例中,本發明係將過濾頻帶較大的帶通濾波器組合起來,可在不同的頻率有不同的增益,且在不同頻帶的交界區域的變化比較連續,如第4A圖中之用於低頻帶的帶通濾波器410及用於高頻帶的帶通濾波器411,以及第4B圖中用於低頻帶的帶通濾波器412及用於高頻帶 的帶通濾波器413所示。需注意的是,為了便於說明,在第4A及4B圖中係以兩個頻帶為例,在後述的實施例中,係以四個頻帶為例進行說明。相較於第3A及3B圖中之帶通濾波器,在第4A及4B圖中之帶通濾波器在頻帶兩側之斜率較為平緩。 4A and 4B are diagrams showing the distribution of different band pass filters in accordance with an embodiment of the present invention. In an embodiment, the present invention combines bandpass filters with larger filtering bands, which can have different gains at different frequencies, and the transitions in different boundary regions are relatively continuous, as shown in FIG. 4A. The band pass filter 410 for the low band and the band pass filter 411 for the high band, and the band pass filter 412 for the low band and the band pass filter 413 for the high band in Fig. 4B are shown. . It is to be noted that, for convenience of explanation, two bands are taken as an example in FIGS. 4A and 4B, and in the following-described embodiments, four bands are taken as an example for description. Compared to the bandpass filters in Figures 3A and 3B, the bandpass filters in Figures 4A and 4B have a flat slope on both sides of the band.
第5圖係顯示依據本發明一實施例中之分頻濾波增益優化方法的流程圖。在方塊510,取得使用者之一聽力衰減曲線。舉例來說,本發明係針對使用者(即聽障人士)之聽力檢測使用適配的五組頻率f 1~f 5進行測量,例如f 1=250Hz、f 2=500Hz、f 3=1000Hz、f 4=2000Hz、f 5=4000Hz,藉以確認聽障人士在個別適配頻率的衰減量H250、H500、H1000、H2000、及H4000。接著,本發明係利用內插法計算在其他適配頻率的衰減量,例如在750Hz、1500Hz、及3000Hz之衰減量H750、H1500、及H3000。舉例來說:H 750=0.5(H 500+H 1000) Figure 5 is a flow chart showing a frequency division filter gain optimization method in accordance with an embodiment of the present invention. At block 510, a hearing attenuation curve for one of the users is obtained. For example, the present invention measures the hearing detection of a user (ie, a hearing impaired person) using five sets of frequencies f 1 to f 5 that are adapted, such as f 1 =250 Hz , f 2 =500 Hz , f 3 = 1000 Hz , f 4 =2000 Hz , f 5 =4000 Hz , to confirm the attenuation of the hearing impairment at the individual adaptation frequencies H 250 , H 500 , H 1000 , H 2000 , and H 4000 . Next, the present invention calculates the attenuation at other adaptation frequencies using interpolation, such as attenuations H 750 , H 1500 , and H 3000 at 750 Hz, 1500 Hz, and 3000 Hz. For example: H 750 =0.5 ( H 500 + H 1000 )
H 1500=0.5(H 1000+H 2000) H 1500 =0.5 ( H 1000 + H 2000 )
H 3000=0.5(H 2000+H 4000) H 3000 = 0.5 ( H 2000 + H 4000 )
因此,可取得8個不同適配頻率的衰減量,並確認聽障人士之聽力衰減曲線。 Therefore, the attenuation of 8 different adaptation frequencies can be obtained, and the hearing attenuation curve of the hearing impaired person is confirmed.
在方塊520,進行一適配頻率增益處理。舉例來說,可針對不同的聽力衰退曲線搭配各種不同的適配增益法(半數增益法、1/3增益法、POGOII法、Berger法、NAL-R法...等等),藉以取得相對於測試頻率的增益值G250、G500、G750、G1000、G1500、G2000、G3000、及G4000。在一實施例中,本發明中係採用NAL-R法以計算聽力衰退曲線在不同測試頻率的增益 值,但本發明並不以此為限。 At block 520, an adaptive frequency gain process is performed. For example, different hearing gain curves can be used for different hearing gain methods (half gain method, 1/3 gain method, POGOII method, Berger method, NAL-R method, etc.) to obtain relative Gain values G 250 , G 500 , G 750 , G 1000 , G 1500 , G 2000 , G 3000 , and G 4000 at the test frequency. In an embodiment, the NAL-R method is used in the present invention to calculate the gain value of the hearing decline curve at different test frequencies, but the invention is not limited thereto.
在方塊530,套用一分段帶通濾波器。舉例來說,本發明係可使用傳統的有限脈衝響應(finite impulse response,FIR)帶通濾波器b c (k)。此有限脈衝響應帶通濾波器b c (k)的第k個係數搭配適合的視窗w(k)。分段帶通濾波器即包括了不同頻帶的帶通濾波器,例如各頻帶的帶通濾波器B c (k)=b c (k).w(k),其中第一頻帶B1為0~1000Hz,第二頻帶B2為1000~2000Hz,第三頻帶B3為2000~4000Hz,第四頻帶B4為4000~8000Hz。 At block 530, a segmented bandpass filter is applied. For example, the invention may use a conventional finite impulse response (finite impulse response, FIR) band-pass filter b c (k). The kth coefficient of this finite impulse response bandpass filter b c (k) is matched with the appropriate window w(k) . That segment comprises a band pass filter of band-pass filters of different frequency bands, each frequency band, for example band-pass filter B c (k) = b c (k). w(k) , wherein the first frequency band B1 is 0 to 1000 Hz, the second frequency band B2 is 1000 to 2000 Hz, the third frequency band B3 is 2000 to 4000 Hz, and the fourth frequency band B4 is 4000 to 8000 Hz.
在方塊540,計算帶通濾波器B c (k)所相應的適配頻率關係矩陣。舉例來說,本發明係利用一取樣頻率f s 設計適配頻率的一弦波訊號(k),弦波訊號(k)可表示如下:
弦波訊號(k)係通過各頻帶的帶通濾波器,並計算其適配頻率關係矩陣,例如在上述步驟採用了4個頻帶的帶通濾波器及8個適配頻率增益,故適配頻率關係矩陣在此實施例中為一8x4矩陣。更進一步而言,若適配頻率增益之數量為M(例如為第一數量),頻帶之數量為N(例如為第二數量),則適配頻率關係矩陣之大小為M.N。在此實施例中M≠N,即第一數量不等於第二數量。 String signal ( k ) is a bandpass filter passing through each frequency band, and calculates an adaptive frequency relationship matrix. For example, in the above steps, a bandpass filter of 4 frequency bands and 8 adaptive frequency gains are used, so the frequency relationship matrix is adapted. In this embodiment it is an 8x4 matrix. Furthermore, if the number of adaptation frequency gains is M (for example, the first number) and the number of frequency bands is N (for example, the second number), the size of the adaptation frequency relationship matrix is M. N. In this embodiment, M ≠ N, that is, the first number is not equal to the second number.
舉例來說,適配頻率關係矩陣可表示如下:
即為一個振幅為1,振動頻率為f j 的訊號經過濾波器B i 所呈現的狀態。簡單來說,雖然各頻帶的帶通濾波器B c (k)是經過視窗w(k)計算而得,但實際上各個帶通濾波器兩側均會與其他的帶通濾波器有交界區,故需計算其相互影響,即上述的適配頻率關係矩陣。 That is, a signal whose amplitude is 1 and whose vibration frequency is f j passes through the state presented by the filter B i . In brief, although the bandpass filter B c (k) of each band is calculated by the window w(k) , in reality, each bandpass filter has a boundary with other bandpass filters on both sides. Therefore, it is necessary to calculate the mutual influence, that is, the above-mentioned adaptation frequency relationship matrix.
在方塊550,計算各帶通濾波器B c (k)之增益。舉例來說,轉換適配頻率增益可由下列矩陣表示:
簡單來說,分段帶通濾波器係以B c (k)表示,適配頻率關係矩陣係以表示,適配頻率增益係以表示,各帶通濾波器所需之增益為,且上述參數之關係式為: In simple terms, the segmented bandpass filter is represented by B c (k) , and the adaptive frequency relationship matrix is Said that the adaptation frequency gain is Indicates that the gain required for each bandpass filter is And the relationship of the above parameters is:
此時,各個帶通濾波器B c (k)所需的增益可用下式表示:
在方塊520及540中已分別計算出適配頻率增益及適配頻率關係矩陣,故各個帶通濾波器B c (k)所需的增益可依據已知的適配頻率增益及適配頻率關係矩陣計算而 得。 The adapted frequency gain has been calculated in blocks 520 and 540, respectively. And adaptive frequency relationship matrix Therefore, the gain required for each bandpass filter B c (k) Can be based on known adaptation frequency gain And adaptive frequency relationship matrix Calculated.
在方塊560,更新分段帶通濾波器之濾波器特徵及增益。舉例來說,需先確認各帶通濾波器之增益R i 之相位為相消或相長,例如:
接著,再更新分段濾波器特性及增益,並將每一個頻帶新的補償增益轉換為dB值,例如。 Then, update the segmentation filter characteristics. And gain And convert the new compensation gain for each band into a dB value, for example .
在方塊570,依據每一個頻段新的帶通濾波器特性,音訊處理電路220可調控輸入聲音訊號,將其分成N個頻帶,然後透過補償增益r i 調控WDRC的增益特性,最後將每一個頻帶的結果整合,成為電子裝置200之揚聲器231的輸出音訊信號。舉例來說,輸入音訊信號經過各個帶通濾波器分別處理以合成輸出音訊信號之流程係顯示於第6圖。 At block 570, new bandpass filter characteristics are selected for each band. The audio processing circuit 220 can adjust the input audio signal, divide it into N frequency bands, and then adjust the gain characteristic of the WDRC through the compensation gain r i , and finally integrate the result of each frequency band to become the output audio signal of the speaker 231 of the electronic device 200. . For example, the flow of the input audio signals processed by the respective band pass filters to synthesize the output audio signals is shown in FIG.
更進一步而言,各帶通濾波器具有相應的補償增益(例如r1~r4),且經過各帶通濾波器之音訊信號經過補償增益後,會進入相應的WDRC處理進行計算,例如方塊611-614中的WDRC1~WDRC4。最後,將WDRC1~WDRC4所產生之個別音訊信號合成為輸出音訊信號。 Furthermore, each band pass filter has a corresponding compensation gain (for example, r1~r4), and after the audio signal of each band pass filter passes the compensation gain, it enters the corresponding WDRC processing for calculation, for example, block 611- WDRC1~WDRC4 in 614. Finally, the individual audio signals generated by WDRC1~WDRC4 are combined into an output audio signal.
第7圖係顯示依據本發明另一實施例中之分頻濾波增益優化方法的流程圖。在方塊710,取得複數個適配頻率增益。舉例來說,取得該複數個適配頻率增益可用兩種方法實現。第一種方法是預先將該複數個適配頻率增益儲存於電子裝 置200之一非揮發性記憶體(未繪示)。這些預先儲存的適配頻率增益可符合大多數聽障人士所需求的各頻率增益。第二種方法是取得使用者之一聽力衰減曲線。舉例來說,可針對使用者(即聽障人士)之聽力檢測使用適配的五組頻率f 1~f 5進行測量,例如f 1=250Hz、f 2=500Hz、f 3=1000Hz、f 4=2000Hz、f 5=4000Hz,藉以確認聽障人士在個別適配頻率的衰減量H250、H500、H1000、H2000、及H4000。接著,本發明係利用內插法計算在其他適配頻率的衰減量,例如在750Hz、1500Hz、及3000Hz之衰減量H750、H1500、及H3000。舉例來說:H 750=0.5(H 500+H 1000) Figure 7 is a flow chart showing a frequency division filter gain optimization method in accordance with another embodiment of the present invention. At block 710, a plurality of adapted frequency gains are obtained. For example, obtaining the plurality of adapted frequency gains can be implemented in two ways. The first method is to store the plurality of adaptive frequency gains in a non-volatile memory (not shown) of the electronic device 200 in advance. These pre-stored adapted frequency gains are compatible with the various frequency gains required by most hearing impaired people. The second method is to obtain one of the user's hearing attenuation curves. For example, the hearing detection for the user (ie the hearing impaired) can be measured using the adapted five sets of frequencies f 1 ~ f 5 , eg f 1 =250 Hz , f 2 =500 Hz , f 3 =1000 Hz , f 4 =2000 Hz , f 5 =4000 Hz , to confirm the attenuation of the hearing impairment at the individual adaptation frequencies H 250 , H 500 , H 1000 , H 2000 , and H 4000 . Next, the present invention calculates the attenuation at other adaptation frequencies using interpolation, such as attenuations H 750 , H 1500 , and H 3000 at 750 Hz, 1500 Hz, and 3000 Hz. For example: H 750 =0.5 ( H 500 + H 1000 )
H 1500=0.5(H 1000+H 2000) H 1500 =0.5 ( H 1000 + H 2000 )
H 3000=0.5(H 2000+H 4000) H 3000 = 0.5 ( H 2000 + H 4000 )
因此,可取得8個不同適配頻率的衰減量,並確認聽障人士之聽力衰減曲線。 Therefore, the attenuation of 8 different adaptation frequencies can be obtained, and the hearing attenuation curve of the hearing impaired person is confirmed.
,接著,可對所取得的聽力衰減曲線進行一適配頻率增益處理。舉例來說,可針對不同的聽力衰退曲線搭配各種不同的適配增益法(半數增益法、1/3增益法、POGOII法、Berger法、NAL-R法...等等),藉以取得相對於測試頻率的增益值G250、G500、G750、G1000、G1500、G2000、G3000、及G4000。在一實施例中,本發明中係採用NAL-R法以計算聽力衰退曲線在不同測試頻率的增益值,但本發明並不以此為限。 Then, an adaptive frequency gain process can be performed on the obtained hearing attenuation curve. For example, different hearing gain curves can be used for different hearing gain methods (half gain method, 1/3 gain method, POGOII method, Berger method, NAL-R method, etc.) to obtain relative Gain values G 250 , G 500 , G 750 , G 1000 , G 1500 , G 2000 , G 3000 , and G 4000 at the test frequency. In an embodiment, the NAL-R method is used in the present invention to calculate the gain value of the hearing decline curve at different test frequencies, but the invention is not limited thereto.
在方塊730,套用一分段帶通濾波器。舉例來說,本發明係可使用傳統的有限脈衝響應(finite impulse response,FIR)帶通濾波器b c (k)。此有限脈衝響應帶通濾波器 b c (k)的第k個係數搭配適合的視窗w(k)。分段帶通濾波器即包括了不同頻帶的帶通濾波器,例如各頻帶的帶通濾波器B c (k)=b c (k).w(k),其中第一頻帶B1為0~1000Hz,第二頻帶B2為1000~2000Hz,第三頻帶B3為2000~4000Hz,第四頻帶B4為4000~8000Hz。 At block 730, a segmented bandpass filter is applied. For example, the invention may use a conventional finite impulse response (finite impulse response, FIR) band-pass filter b c (k). The kth coefficient of this finite impulse response bandpass filter b c (k) is matched with the appropriate window w(k) . That segment comprises a band pass filter of band-pass filters of different frequency bands, each frequency band, for example band-pass filter B c (k) = b c (k). w(k) , wherein the first frequency band B1 is 0 to 1000 Hz, the second frequency band B2 is 1000 to 2000 Hz, the third frequency band B3 is 2000 to 4000 Hz, and the fourth frequency band B4 is 4000 to 8000 Hz.
在方塊740,計算帶通濾波器B c (k)所相應的Mx(N-1)適配頻率關係矩陣。舉例來說,本發明係利用一取樣頻率f s 設計適配頻率的一弦波訊號(k),弦波訊號(k)可表示如下:
因為4KHz以上的高頻信號對於電子裝置之揚聲器231來說,易產生嘯叫聲,故需對輸出音訊信號在高頻部份的增益有所限制。更進一步而言,輸出音訊信號在高頻部份與輸入音訊信號相同,即高頻部份的增益不變,故高頻部份之增益可用一8x1矩陣表示(即8個適配頻率增益搭配4KHz之頻帶):
此外,並計算4KHz以下之音訊信號所相應的適配頻率關係矩陣,例如可用一8x3矩陣表示,意即8個適配頻率增益搭配4KHz以下的3個頻帶,若適配頻率增益之數量為M、帶通濾波器之數量為N,則相應於各帶通濾波器(未包含高頻帶通濾波器)之適配頻率關係矩陣例如可表示為Mx(N-1)矩陣:
弦波訊號(k)係通過各頻帶的帶通濾波器,並計算其適配頻率關係矩陣,例如在上述步驟採用了4KHz以下3個頻帶的帶通濾波器及8個適配頻率增益,故適配頻率關係矩陣在此實施例中為一8x3矩陣。 String signal ( k ) is a bandpass filter that passes through each frequency band, and calculates an adaptive frequency relationship matrix. For example, in the above steps, a bandpass filter of three frequency bands below 4 kHz and eight adaptive frequency gains are used, so the frequency is adapted. The relationship matrix is an 8x3 matrix in this embodiment.
即為一個振幅為1,振動頻率為f j 的訊號經過濾波器B i 所呈現的狀態。簡單來說,雖然各頻帶的帶通濾波器B c (k)是經過視窗w(k)計算而得,但實際上各個帶通濾波器兩側均會與其他的帶通濾波器有交界區,故需計算其相互影響,即上述的適配頻率關係矩陣。 That is, a signal whose amplitude is 1 and whose vibration frequency is f j passes through the state presented by the filter B i . In brief, although the bandpass filter B c (k) of each band is calculated by the window w(k) , in reality, each bandpass filter has a boundary with other bandpass filters on both sides. Therefore, it is necessary to calculate the mutual influence, that is, the above-mentioned adaptation frequency relationship matrix.
在方塊750,計算各帶通濾波器B c (k)之增益。舉例來說,轉換適配頻率增益可由下列矩陣表示:
簡單來說,分段帶通濾波器係以B c (k)表示,適配頻率關係矩陣係以表示,適配頻率增益係以表示,各帶通濾波器所需之增益為,且上述參數之關係式為: In simple terms, the segmented bandpass filter is represented by B c (k) , and the adaptive frequency relationship matrix is Said that the adaptation frequency gain is Indicates that the gain required for each bandpass filter is And the relationship of the above parameters is:
此時,各個帶通濾波器B c (k)所需的增益可用下式表示:
第四頻帶(4KHz以上)的增益則固定為1。 The gain of the fourth frequency band (above 4 kHz) is fixed at 1.
在方塊720及740中已分別計算出適配頻率增益及適配頻率關係矩陣,故各個帶通濾波器B c (k)所需的增益可依據已知的適配頻率增益及適配頻率關係矩陣計算而得。 The adapted frequency gain has been calculated in blocks 720 and 740, respectively. And adaptive frequency relationship matrix Therefore, the gain required for each bandpass filter B c (k) Can be based on known adaptation frequency gain And adaptive frequency relationship matrix Calculated.
在方塊760,更新分段帶通濾波器之濾波器特徵及補償增益。舉例來說,需先確認各帶通濾波器之增益R i 之相位為相消或相長,例如:
接著,再更新分段濾波器特性及增益,並將每一個頻帶新的補償增益轉換為dB值,例如。 Then, update the segmentation filter characteristics. And gain And convert the new compensation gain for each band into a dB value, for example .
在方塊770,合成輸出音訊信號。更進一步而言,依據每一個頻段新的帶通濾波器特性,音訊處理電路220可調控輸入聲音訊號,將其分成N個頻帶,然後透過補償增益r i 調控WDRC的增益特性,最後將每一個頻帶之帶通濾波器的輸出信號合成為電子裝置200之揚聲器231的輸出音訊信號。舉例來說,輸入音訊信號經過各個帶通濾波器分別處理以合成輸出音訊信號之流程係顯示於第6圖。 At block 770, the output audio signal is synthesized. Further, based on the new bandpass filter characteristics of each band The audio processing circuit 220 can adjust the input audio signal, divide it into N frequency bands, and then adjust the gain characteristic of the WDRC through the compensation gain r i , and finally synthesize the output signal of the band pass filter of each frequency band into the speaker of the electronic device 200. 231 output audio signal. For example, the flow of the input audio signals processed by the respective band pass filters to synthesize the output audio signals is shown in FIG.
更進一步而言,各帶通濾波器具有相應的補償增益(例如r1~r4),且經過各帶通濾波器之音訊信號經過補償增 益後,會進入相應的WDRC處理進行計算,例如方塊611-614中的WDRC1~WDRC4。最後,將WDRC1~WDRC4所產生之個別音訊信號合成為輸出音訊信號。 Furthermore, each band pass filter has a corresponding compensation gain (for example, r1~r4), and after the audio signal of each band pass filter passes the compensation gain, it enters the corresponding WDRC processing for calculation, for example, block 611- WDRC1~WDRC4 in 614. Finally, the individual audio signals generated by WDRC1~WDRC4 are combined into an output audio signal.
相較於本案第5圖,本案第7圖中之分頻濾波增益優化方法的流程圖更能針對電子裝置上之揚聲器的特性對高頻音訊信號另外進行特別處理,使得高頻音訊信號不會在揚聲器播放時產生嘯叫聲,而且更可針對除了高頻信號之外的部份進行補償增益之優化。 Compared with the fifth picture of the present case, the flow chart of the frequency-dividing filter gain optimization method in the seventh picture of the present case can further specially process the high-frequency audio signal for the characteristics of the speaker on the electronic device, so that the high-frequency audio signal does not The whistling sound is generated when the speaker is played, and the compensation gain is optimized for the portion other than the high frequency signal.
本發明雖以較佳實施例揭露如上,然其並非用以限定本發明的範圍,任何所屬技術領域中具有通常知識者,在不脫離本發明之精神和範圍內,當可做些許的更動與潤飾,因此本發明之保護範圍當視後附之申請專利範圍所界定者為準。 The present invention has been disclosed in the above preferred embodiments, and is not intended to limit the scope of the present invention. Any one of ordinary skill in the art can make a few changes without departing from the spirit and scope of the invention. The scope of protection of the present invention is therefore defined by the scope of the appended claims.
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