1232036 五、發明說明（1) --- 【發明所屬之技術領域】 本發明係有關一種訊號輸出技術，特別是關於一種訊 魂輪出之波形還原取樣方法，使輸出頻率適用於各種操作 頻率而不受限制。 【先前技術】 按，聲音或影像訊號之輸出皆涉及取樣技術，以聲音 輸出為例，播放裝置之標準取樣操作頻率通常為32 KHz、 48 KHz及96 KHz或22·05 KHz及44·1 KHz等，其中音訊之 取樣率或稱取樣頻率（Sampling Rate)係指在二秒之内對 聲音波形做記錄（取樣）的次數。 習知音訊輸出之取樣方法如第一圖所示，主機丨〇係透 過USB傳輸介面與一 USB揚聲器12連接，此主機與揚 聲器12之溝通頻率為12MHz，而揚聲器12之播放頻率為 1 2.288 MHz ’故揚聲器12必須將音頻昇頻至12·288ΜΗζ才 能進行處理。習知做法係在USB揚聲器丨2内設置一音訊資 料接收裝置13、一時脈合成器（ci〇ck Synthesizei〇14、 一數位類比轉換器（digital to analog converter，MC) 16及一放大器18，當主機l〇以48KHz之頻率對一聲音檔取 樣而輸出至USB揚聲器12時，係先由音訊資料接收裝置 接收，接著由時脈合成器14將操作頻率由12 MHz調整為 12· 2 88MHz後，而後才由數位類比轉換器16及放大器18將 數位訊號轉換為類比訊號及放大輸出。然而，此種音訊取 樣方法受揚聲器1 2之操作頻率所限制，必須考慮主機1 〇及 USB揚聲器12頻率同步之匹配問題，故必須在usb揚聲器121232036 V. Description of the invention (1) --- [Technical field to which the invention belongs] The present invention relates to a signal output technology, and in particular, to a waveform restoration sampling method of a signal soul wheel, so that the output frequency is suitable for various operating frequencies. Unlimited. [Previous technology] Pressing, sound or video signal output involves sampling technology. Taking sound output as an example, the standard sampling operation frequency of playback devices is usually 32 KHz, 48 KHz and 96 KHz or 22 · 05 KHz and 44.1 KHz. Etc. Among them, the sampling rate or sampling rate of the audio refers to the number of recordings (sampling) of the sound waveform within two seconds. The sampling method of the conventional audio output is shown in the first figure. The host is connected to a USB speaker 12 through a USB transmission interface. The communication frequency between the host and the speaker 12 is 12 MHz, and the playback frequency of the speaker 12 is 1 2.288 MHz. 'Therefore, the speaker 12 must up-convert the audio to 12.288 MHz, for processing. A conventional method is to set an audio data receiving device 13, a clock synthesizer 14, a digital to analog converter (MC) 16 and an amplifier 18 in the USB speaker 2. When the host 10 samples a sound file at a frequency of 48KHz and outputs it to the USB speaker 12, it is first received by the audio data receiving device, and then the operating frequency is adjusted by the clock synthesizer 14 from 12 MHz to 12 · 88 MHz. The digital analog converter 16 and the amplifier 18 then convert the digital signal into an analog signal and amplify the output. However, this audio sampling method is limited by the operating frequency of the speaker 12 and the frequency synchronization between the host 10 and the USB speaker 12 must be considered Matching problem, it must be in the usb speaker 12
五、發明說明（2) 内增設時脈合成器1 4以調整頻率；另一方面，不同操作頻 率便必須使用不同設定之時脈合成器1 4，使用上極為不 便。 另 ^知技術係將取樣率轉換器（s a m p 1 i n g r a t e converter)内建於揚聲器中，藉以將取樣率轉換之工作轉 而由揚聲器進行，然而，如此卻使得揚聲器之架構變得龐 大，且導致成本提高許多。 、因此，本發明即針對上述之問題，提出一種訊號輸出 之波形還原取樣方法，其係直接在主機端，以軟體或韌體 將取樣率進行轉換，以有效克服習知該等缺失。 【發明内容】 本發月之主要目的，係在提供一種訊號輸出之波形還 原取樣方法，藉以在提供等效訊號輸出之前提下，僅需使 =體即可轉換取樣率而還原訊號使其輸出，it而使主機 ^輸出頻率不需又標準操作頻率所限制，徹底決習 知頻率匹配問題。 一 肩®日f之另—目的，係在提供—種訊號輸出之波形還 '、本法，具有適用於各種操作頻率之優點者。 原取樣之再一目的，係在提供一種訊號輸出之波形還 效達成$ # i其係可省除時脈合成器及取樣率轉換器，有 ΐίί:效/視效配置成本之功效。 方法係吉述之目的，本發明訊號輸出之波形還原取樣 一聲音/ & # 、一主機端將一原始數位訊號還原取出而提供 聲曰像輸出裝置輸出，該方法包括下列步驟：首 1232036 五、發明說明（3) 先，在主機端對該原始數位訊號進行平滑處理（sm〇〇t 以重建=模擬訊號；而後以一取樣率對該模擬訊號進行 取樣，其中該取樣率係為該輸出袭置之操作頻率除以 輸倍數的值，進而經由取樣得到一還焉 :电 退原訊唬，以傳送至該 輸出裝置。 底下藉由具體實施例配合所附的圖式詳加說明， 容易瞭解本發明之目的、技術内容、特點及其所達成:功 效。 【實施方式】 本發明係於訊號輸出時’在主機端將原始數位訊號予 以平滑化而重建出一模擬訊號之後，接著再以一取樣率對 該模擬訊號進行取樣，進而產生一還原訊號而輸出，藉此 使主機端最終輸出頻率不需固定。 ，論聲音或影像訊號之輸出皆涉及取樣技術，一般而 a，聲音播放裝置之標準取樣頻率通常為44 1〖Hz、32 KHz、48 KHz及96 KHz等，而音訊之取樣率（sampHng rak)係指在一秒之内對聲音波形取樣的次數。本發明之 波形還原取樣方法適用於聲音及影像訊號，以下將以音訊 輸出為例來詳細說明本發明之方法。本發明所配合之硬體 如第二圖所示，一主機端2〇係透過一傳輸介面連接有一聲 音輸出裝置，聲音輸出裝置係為一揚聲器（speaker)22， ，傳輸介面通常為USB、IEEE1 394或藍芽（bluetooth)，且 虽傳輸介面為USB1· 1時，揚聲器22之操作頻率係為12 MHz ’傳輸介面為USB2· 0時，揚聲器22之操作頻率係為30V. Description of the invention (2) A clock synthesizer 14 is added to adjust the frequency; on the other hand, different operating frequencies must use a clock synthesizer 14 with different settings, which is extremely inconvenient to use. Another known technology is that the sample rate converter (samp 1 ingrate converter) is built into the speaker, so that the work of converting the sample rate is performed by the speaker. However, this makes the speaker structure large and causes cost. Improve a lot. Therefore, the present invention is directed to the above-mentioned problems, and proposes a method for waveform reduction sampling of signal output, which directly converts the sampling rate by software or firmware on the host side to effectively overcome these shortcomings. [Summary of the invention] The main purpose of this month is to provide a method for waveform reduction sampling of signal output, so as to raise it before providing equivalent signal output, only need to make the body to convert the sampling rate and restore the signal to output , It makes the host ^ output frequency need not be limited by the standard operating frequency, thoroughly determine the frequency matching problem. The other purpose is to provide waveforms of signal output. This method has the advantage of being suitable for various operating frequencies. Another purpose of the original sampling is to provide a signal output waveform that is effective to achieve $ # i It can save the clock synthesizer and sampling rate converter, which has the effect of cost / effect configuration cost. The method is for the purpose of description. The waveform of the signal output of the present invention is sampled with a sound / amplifier, and a host terminal restores an original digital signal to provide audio and video output device output. The method includes the following steps: First 1232036 3. Description of the invention (3) First, the original digital signal is smoothed on the host side (smOOt = reconstruction = analog signal; then the analog signal is sampled at a sampling rate, where the sampling rate is the output Divide the operating frequency by the value of the input multiplier, and then get a response through sampling: the electric signal is sent back to the output device. The detailed description below with specific examples in conjunction with the attached drawings is easy. Understand the purpose, technical content, characteristics and achievement of the present invention and its effect: [Embodiment] The present invention is to 'smooth the original digital signal on the host side to reconstruct an analog signal when the signal is output, and then use the The analog signal is sampled at a sampling rate, and then a restored signal is generated and output, so that the final output frequency of the host does not need to be fixed. The output of sound or video signals involves sampling technology. Generally, a, the standard sampling frequency of sound playback devices is usually 44 1 [Hz, 32 KHz, 48 KHz, 96 KHz, etc., and the sampling rate of audio (sampHng rak) refers to The number of times the sound waveform is sampled in one second. The waveform reduction sampling method of the present invention is suitable for sound and video signals. The following will take the audio output as an example to describe the method of the present invention in detail. As shown in the two figures, a host end 20 is connected to a sound output device through a transmission interface. The sound output device is a speaker 22. The transmission interface is usually USB, IEEE1 394, or bluetooth, and Although the transmission interface is USB1 · 1, the operating frequency of the speaker 22 is 12 MHz. 'When the transmission interface is USB2 · 0, the operating frequency of the speaker 22 is 30.
1232036 MHz，而當傳輸介面為IEEE 1 394時，揚聲器22之操作頻率 則為400 MHz，以下將以USB1.1傳輸介面之12MHz操作頻率 為例，說明在音訊輸出時，於主機端2 0運用本發明之方法 對音波進行取樣加以還原而輸出之工作流程。 第三圖係為本發明於音訊輸出之音波還原取樣方法示 意圖，此音波還原取樣方法係以軟體或韌體在主機端2 〇内 實現，包括下列步驟：首先，主機端20以一可配合原始數 位音訊DSWin頻率之頻率f s 1對該原始數位音訊j)SWin進行平 滑處理（smooth)，以便將原始數位音訊DSWin重建成為一平 滑之模擬音訊；該頻率fsl通常為44. 1 KHz、32 KHz、48 KHz或96 KHz，且頻率fsl必須與原始數位音訊DSWin輸入之 頻率耦合，例如DSWin 416KHz者，則fsl以16KHz重建；若 DSWin為4 8KHz者，則fsl以48KHz重建。在以頻率fsl重建出一 模擬音訊之後，接著係以一取樣率U對該模擬音訊進行取 樣，進而經由取樣得到一還原音訊DSW〇ut，提供輸出至 器22。 卑 其中’該取樣率fsZ之值係為揚聲器22之操作頻率除以 一傳輸倍數的值所求得，此取樣率之計算方法係操作頻率 =知為12MHz，令傳輸倍數為256，則以12MHz除以256，可 知到取樣率fsZ為46· 875KHz。其中，由於揚聲器22為 l|SBl.l傳輸介面者，因此當揚聲器以與主機端2〇電連接之 後丄本發作之軟體即主動告知主機端20以46· 875KHz為取 rH’黎另外，傳輸倍數通常係定義為2n倍’如128、256及 512倍等’256倍係較常使用者，然當不限定為256倍。及1232036 MHz, and when the transmission interface is IEEE 1 394, the operating frequency of the speaker 22 is 400 MHz. The following will take the 12MHz operating frequency of the USB1.1 transmission interface as an example to illustrate the use of the host 20 at the audio output. The method of the present invention is a work flow for sampling and restoring sound waves and outputting them. The third figure is a schematic diagram of the sound wave reduction sampling method at the audio output of the present invention. The sound wave reduction sampling method is implemented by software or firmware in the host terminal 20, and includes the following steps: First, the host terminal 20 Digital audio DSWin frequency fs 1 The original digital audio j) SWin is smoothed in order to reconstruct the original digital audio DSWin into a smooth analog audio; the frequency fsl is usually 44.1 KHz, 32 KHz, 48 KHz or 96 KHz, and the frequency fsl must be coupled with the frequency of the original digital audio DSWin input. For example, DSWin 416KHz, fsl is rebuilt at 16KHz; if DSWin is 4 8KHz, fsl is rebuilt at 48KHz. After an analog audio is reconstructed at the frequency fsl, the analog audio is sampled at a sampling rate U, and then a restored audio DSWout is obtained through sampling, and provided to the output device 22. The value of the sampling rate fsZ is obtained by dividing the operating frequency of the speaker 22 by a transmission multiple. The calculation method of this sampling rate is the operating frequency = known as 12MHz, and the transmission multiple is 256. Dividing by 256, it can be seen that the sampling rate fsZ is 46 · 875KHz. Among them, since the speaker 22 is a l | SBl.l transmission interface, after the speaker is electrically connected to the host terminal 20, the software that initiates the attack will actively inform the host terminal 20 to take 46 · 875KHz as rH '. In addition, the transmission Multiples are usually defined as 2n times, such as 128, 256, and 512 times. 256 times are more common users, but they are not limited to 256 times. and
第8頁 五、發明說明（5) 取係可在尚未重取土所使^之取，率fs2，其求 器22之操作頻| = '、D數位音訊之前，先藉由偵测揚聲 是在重建= 操作頻率而計算出取樣率；Ϊ 前，先偵測揚聲模擬音訊進行取樣之步驟 算出取樣率。。之#作頻率，且依據該操作頻率而計 謂.ί^ί發明之方法於主機端2G將—原始數位音訊 ν旋m:還原數位音訊之後，該還原數位音 聲器2r、yn 聲器22。請同時參第四圖所示，揚 聲㈣通“糸内建有-音訊資料接收裝置23、-數位類比 轉換器（digital to analog converter，DAC)24 及一放大 器26 士’藉以當主機端2〇以適當頻帛對一聲音檔取樣而輸出 至揚聲器22時，係先由音訊資料接收裝置23接收該聲音 檔，再由數位類比轉換器24將聲音訊號由數位型式轉換為 人耳可聽到之類比音訊，且由放大器26放大類比訊號後而 播放出來。 因此，於訊號輸出時，本發明係在主機端重建出一模 擬訊號之後，藉由直接以一取樣率對該模擬訊號進行取樣 而產生一還原訊號提供聲音/影像輸出裝置輸出，以藉此 在提供等效音訊/影訊輸出之前提下，僅需使用軟體即可 轉換取樣率而還原訊號使其輸出，進而使主機端最終輸出 頻率不需受標準操作頻率所限制。故本發明無習知揚聲器 與主機之頻率匹配問題，不僅具有適用於各種操作頻率之 優點，同時玎達到省除時脈合成器及取樣率轉換器之功 1232036 五、發明說明（6) 效，進而有效降低音效配置之成本。 以上所述係藉由實施例說明本發明之特點，其目的在 使熟習該技術者能暸解本發明之内容並據以實施，而非限 定本發明之專利範圍，故，凡其他未脫離本發明所揭示之 精神所完成之等效修飾或修改，仍應包含在以下所述之申 請專利範圍中。Page 8 V. Description of the invention (5) It can be taken before the soil is retrieved again, the rate is fs2, and the operating frequency of the seeker 22 | = ', D digital audio, by detecting the speaker first The sampling rate is calculated after reconstruction = operating frequency; Ϊ Before, the steps of detecting speaker analog audio and sampling are used to calculate the sampling rate. . ## work frequency, and count according to the operating frequency. Ί ^ ί Invented method on the host 2G will be-original digital audio ν spin m: After the digital audio is restored, the restored digital sounder 2r, yn sounder 22 . Please also refer to the fourth figure at the same time, the speaker is connected "with built-in audio data receiving device 23,-digital to analog converter (DAC) 24 and an amplifier 26 +" to be the host side 2 〇 When a sound file is sampled at an appropriate frequency and output to the speaker 22, the audio file is first received by the audio data receiving device 23, and then the digital analog converter 24 converts the sound signal from a digital type to a human ear. The analog audio is amplified by the amplifier 26 and played out. Therefore, when the signal is output, the invention reconstructs an analog signal on the host side and generates the analog signal by sampling the analog signal directly at a sampling rate. A restored signal provides the output of the audio / video output device, so that before the equivalent audio / video output is provided, only the software can be used to convert the sampling rate and restore the signal to output, so that the final output frequency of the host is not changed. Need to be limited by the standard operating frequency. Therefore, the present invention has no conventional frequency matching problem between the speaker and the host, and not only has a suitable frequency for various operating frequencies. Advantages, at the same time, it can save the work of clock synthesizer and sample rate converter. 1232036 V. Description of the invention (6) effect, which can effectively reduce the cost of sound effect configuration. The above is the description of the features of the present invention through the embodiment. The purpose is to enable those skilled in the art to understand and implement the content of the present invention, rather than to limit the patent scope of the present invention. Therefore, other equivalent modifications or modifications made without departing from the spirit disclosed by the present invention are still It should be included in the scope of patent application described below.
第10頁 1232036 圖式簡單說明 圖式說明： 第一圖為習知音訊取樣輸出之硬體裝置示意圖。 第二圖為本發明音訊取樣輸出之硬體裝置示意圖。 第三圖為本發明音訊輸出之取樣方法示意圖。 第四圖為本發明之揚聲器連接有數位類比轉換器之示意 圖。 圖號說明： 10 主機 12揚聲器 1 3音訊資料接收裝置 1 4 時脈合成器 20 主機端 22揚聲器 23音訊資料接收裝置 2 4數位類比轉換器Page 10 1232036 Brief description of the diagrams Description of the diagrams: The first diagram is a schematic diagram of the hardware device of the conventional audio sampling output. The second figure is a schematic diagram of a hardware device for audio sampling output according to the present invention. The third figure is a schematic diagram of a sampling method for audio output according to the present invention. The fourth figure is a schematic diagram of a digital analog converter connected to the speaker of the present invention. Description of figure number: 10 main unit 12 speakers 1 3 audio data receiving device 1 4 clock synthesizer 20 host side 22 speakers 23 audio data receiving device 2 4 digital analog converter
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