分類: 音频技术

  • Beginners Can Achieve Pro-Level Audio Quality Voice Enhancement Say Goodbye to Volume Swings & Noise

    Download Audacity:https://www.audacityteam.org/ 

    Download my Preset:

    The tool of removing ambient noise:https://www.lalal.ai/?fp_ref=samuel76

    I watched a YouTube video today and noticed the audio was really quiet, especially when watching outdoors—it was almost inaudible. It was this one:

    I tweaked her audio a bit, and it turned out like this.

    So, I thought, maybe a lot of you have similar needs for audio enhancement but don’t want it to be too complicated. After all, making videos is already time-consuming enough. So, after some experimenting, I came up with a super simple and beginner-friendly solution. You don’t need any audio tech background—just follow my steps, and you’ll definitely improve the sound quality of your content. Plus, all the tools are free software, and I’ll include all the tools and links on this episode’s webpage, with the URL in the description or comments.

    Before we start, please hit the little bell and subscribe to my channel.

    Let’s first get a quick overview of what’s going on. When you speak into a microphone, your vocal cords vibrate, causing various parts of your body to resonate, producing a continuous series of sound waves. We’ll just call this “speech” from now on. Before speech reaches the mic, it can include low-frequency sound waves starting at 80 Hz all the way up to high-frequency waves in the thousands of Hz. The energy of these sound waves varies a lot too—sometimes it’s a soft whisper, just a fraction of a decibel, and other times it’s loud, reaching 60 or 70 decibels. Quick sidenote: Hz and dB are physical units for measuring sound. Hz is the unit for frequency, which, in simple terms, affects the tone of the sound. Words like bright, deep, low, or sharp describe characteristics tied to frequency. dB measures sound pressure level, which you can think of as volume.

    In short, the speech coming out of your mouth has a huge range of frequencies and volumes, also called dynamic range. Even the best microphones have limits to their sensitivity, especially in dynamic range. A mic’s sensitivity is far lower than the dynamic range your speech can produce. That’s why you see some professional singers constantly adjusting the distance between their mouth and the mic while performing.

    When we record a show, it’s hard to keep adjusting the mic position like a singer. Usually, we fix the mic at a suitable distance from the mouth and lower the mic’s input level to prevent loud speech from overloading the mic or audio equipment’s dynamic limit. If the input exceeds the equipment’s dynamic limit, you get a squeaky, distorted sound—sometimes called clipping, or in layman’s terms, “audio popping.”

    So, obviously, to avoid popping during recording, we lower the input overall. But then, when the audio is recorded into a computer or phone, you’ll notice the overall volume is pretty low. Some parts might be loud, but others are super quiet. No matter how much you crank up your computer or phone’s volume, the sound won’t be as rich and clear as a radio host’s.

    Let me use that show I mentioned earlier as an example to show you how to achieve the final effect.

    Here, I’m using a free audio software called Audacity. It’s available for both Windows and Mac, super powerful, and the download and installation are really straightforward, so I’ll skip that part. Once installed, click here to download the preset files I prepared for you. On this episode’s webpage, find “Parameter Preset Download” and download the zip file. After downloading, unzip it to get the preset files. Then open Audacity, go to the File menu, select Open, and load your audio file.

    Next, we’re going to do six things:

    **First thing: Filtering**

    The goal here is to make the audio material clean and clear. It’s like using a juice filter—after filtering, you remove unwanted pulp and impurities, leaving just pure juice. During recording, signal amplification, and analog-to-digital conversion, the audio signal picks up some impurities, especially when recording in noisy environments. Filtering mainly reduces signals at certain frequencies.

    Based on my experience, for spoken audio, signals below 100 Hz or above 4,000 Hz are generally irrelevant to speech clarity. Reducing or completely removing these signals won’t affect how clear the speech sounds.

    First, select the audio segment you want to process. Click the Effect menu, go to EQ and Filters, find Filter Curve EQ, then click the Preset & Settings button and choose Import to load the preset file. From the preset files you downloaded, find “Vocal-Filter” and open it. You’ll see I completely filtered out signals below 100 Hz—this is called a low-cut. High frequencies between 4,000 and 8,000 Hz are slightly reduced. This range often includes a lot of environmental noise, sibilance, and mouth noises, but reducing it too much can dull the speech’s brightness, so I only apply a light reduction. Signals above 10,000 Hz get a high-cut, as this range includes some breathy sounds. If your show relies on a lot of breathy, whispery effects, you might want to manually boost this range, but for most speech-based content, a high-cut works fine. Click Preview to hear the processed effect. If you’re happy with it, click Apply to process the audio. Note: once you hit Apply, you can only undo it via the Edit menu’s Undo option. You can also apply the filter multiple times to the same audio for a more thorough effect.

    Now, the audio no longer has that background hum or popping mic sounds, and mouth noises like sibilance are reduced.

    **Second thing: Dynamic Compression**

    Dynamic refers to the range between the quietest and loudest parts of the audio. Compressing the dynamics means narrowing the gap between the loudest and quietest volumes. The main goal is to make the audio’s loudness more consistent, so it doesn’t feel like it’s jumping from loud to soft or far to near.

    How do we do this? One way is to select the audio segment you want to adjust, go to the Effect menu, choose Volume and Compression, open Amplify, and enter a negative value to reduce volume or a positive value to increase it, then hit Apply. This method is flexible and lets you control every detail, but if the audio is long with lots of variations, it can be time-consuming.

    In Audacity, we can use the Compressor for this. Select the audio to process, go to the Effect menu, choose Volume and Compression, find Compressor, and open it. Click the Presets & Settings button, then Import, and from the downloaded preset files, find “VocalStartUp,” open it, and click Apply.

    After processing, the audio is noticeably louder, and details are slightly more pronounced. If some parts are still too loud, you can use the first method to adjust them individually.

    When using the Compressor, besides reducing the dynamic range, you’ll also get some enhanced details and a slight change in sound quality. Applying the Compressor multiple times might produce unexpected effects, so listen carefully to the changes before deciding whether to keep the multi-processed result.

    **Third thing: Distortion**

    You might have heard of electric guitar distortion effects. Here, we’re using a distortion effect on the voice—not to mimic a guitar, but to add harmonic components with a subtle distortion effect, enhancing the audio’s brightness and fullness.

    In Audacity, go to the Effect menu, find Distortion and Modulation, select Distortion, and open it. Click Presets & Settings, then Import. From the downloaded preset files, find “Voice Exciton” and open it. The parameters are: Distortion Type set to Hard Clipping, Clipping Level set between -6 and -10, Drive can be adjusted but defaults to 50, and Make-up Gain should be set to 0 to avoid extra volume boost. Then click Apply.

    Listen to it now. The audio sounds quite different.

    **Fifth thing: Loudness Normalization**

    Many platforms and industries have their own loudness standards. For YouTube, the recommended loudness is no lower than -14 dB LUFS. I won’t dive into what that means here—just follow the steps.

    Select the audio to process, go to the Effect menu, choose Volume and Compression, find Loudness Normalization, and open it. Click Presets & Settings, then Import. From the downloaded preset files, find “YouTube,” open it, and click Apply.

    **Sixth thing: Limiting**

    This is a standard safety process for audio. We’ll use Audacity’s Limiter. Select the audio, go to the Effect menu, choose Volume and Compression, find Limiter, open it, set the Threshold (dB) to -1.0, and click Apply.

    Go to the File menu, choose Export Audio, select Export to Computer, and set the audio file format as recommended here.

    Then click Export.

    At this point, you’ve completed all the speech processing. Doesn’t it sound much louder and clearer than before?

    Although the overall audio is louder, the background noise might get amplified too. What to do? Thanks to AI advancements, I recommend a super practical online tool for removing ambient noise called LALAL.AI.

    Open this website, click here to switch to Chinese, then click Log In. For first-time users, register with your email or another login method. After registering and logging in, go to Products, select Voice Remover, drag your audio file into the page, wait a moment, and it’s done.

    How’s that? Doesn’t it feel like something from a big professional media outlet?

    Alright, that’s it for today’s video. All the tools and links are on this episode’s webpage. If you run into any issues while processing your audio, feel free to ask in the comments, and I’ll reply as soon as I can. If you found this helpful, don’t forget to subscribe, like, and save. Thanks for watching!

    — 

    This translation maintains the original structure and intent, using conversational English suitable for video narration. Let me know if you need further adjustments!

  • 1分鐘搞定-小白也能做出媲美專業主播的音質|全免費聲音美顏技術大爆光|告別忽大忽小|噪聲淹沒|製作清晰洪亮富有磁性的語音教程

    Audacity 下載:https://www.audacityteam.org/  打開官網以後點擊首頁的Download字樣就行

    參數預置下載:

    噪音消除工具鏈接 :https://www.lalal.ai/?fp_ref=samuel76

    今天看了一個油管視頻,發現聲音小,特別是在室外看,幾乎聽不清,就是這個:

    我順手處理了一下她的聲音,就變成了這樣

    於是我想,可能有許多朋友也有這樣的聲音美化需求,但是又不想太麻煩,畢竟大家製作視頻已經夠費時間的了。所以,經過一些嘗試,我總結了一個非常簡單容易上手的方案,不需要任何音頻技術基礎,只要你跟著我的提示一步步操作,一定能改善自己的節目的音質。並且所有的工具全部都使用免費軟件,我也會把所有工具和鏈接放在本期的網頁裏,網址放在說明欄或者留言裏。

    開始之前,請你先點擊小鈴鐺,訂閱我的頻道。

    讓我們先來大致了解一下事情的前因後果吧,當你對著麥克風開始講話的時候,你的聲帶開始震動,並帶動整個身體的各部分共振起來,從而發出一串連續的聲波,後面我們統一稱它為語音。在語音傳遞到話筒之前,它可能包含了從80赫茲的低頻聲波到幾千赫茲的高頻聲波。而且聲波的能量變化也會很大,有時可能是輕聲細語,只有零點幾db,有時又可能很洪亮,甚至達到六十到七十db。快速科普一下哈,這裏說的赫茲和db都是衡量聲音參數的物理單位,hz是震動頻率的單位,頻率簡單理解就是那些影響到聲音音色的參數,我們常說明亮、渾厚、低沈、尖銳等等,這些形容表示的就是頻率值表現出來的特性;db是聲壓級單位,聲壓級可以簡單理解為音量大小。

    總之,從人嘴裏發出的語音,無論是頻率範圍,還是音量範圍,也叫動態範圍,都非常的大,而即使是最好的麥克風,它的敏感度都是有一定限度的,特別是在動態範圍方面,麥克風的靈敏度遠遠低於語音所能產生的動態範圍,這就是為什麽你會看到,有些專業的歌手在演唱的時候會時不時調整麥克風與嘴巴之間的距離。

    當我們錄製節目的時候,當然很難像歌手那樣不斷調整麥克風位置,我們通常會把麥克風固定在一個與嘴巴適合的距離,然後通過適當減小麥克風輸入的量來限製最大的輸入,以防止大聲的說話的時候,突然沖破麥克風和音頻設備的動態極限,一旦最大輸入超出音頻設備的動態極限,我們會得到一個吱吱響的破損的音頻,這種現象有時候稱為溢出,俗稱爆音。

    於是,很顯然,為了保證錄音的時候不爆音,我們會整體減小輸入,這樣,聲音錄進電腦或者手機以後,你會發現,整體的音量都是偏小的,經管有些地方可能比很大,但可能其他地方都很小,無論你如何把電腦或手機的音量調到最大,聲音也不會像電臺主播那樣渾厚清晰。

    我就以開始提到的那個節目為例,給大家展示如何處理成最終的效果。

    這裏我使用一款免費的音頻軟件叫做Audacity,這個軟件有windows版本也有Mac版本,功能非常強大,下載和安裝的過程非常簡單,這裏就跳過了。安裝完成以後,點這裏下載我為大家製作好的預置文件。在本期節目的網頁裏,找到「參數預置下載」,下載這個壓縮包,下載完成後,解壓縮,得到一些預置文件。然後打開Audacity 軟件,從File文件菜單中選擇Open,打開音頻文件。

    接下來我們要做這麽六件事:

    第一件事,濾波

    做這件事的目的是讓聲音素材變得幹凈清晰。它的原理類似於果汁過濾器,過濾以後可以把不要的果皮雜質去掉,只過留下純凈的果汁。聲音從錄製,到信號放大以及模擬數字轉換等等過程中,音頻信號會被添加一些雜質,特別是當你在比較嘈雜的環境中錄音時尤其明顯。濾波主要是降低某些頻率的信號。

    根據我的經驗,對於人講話的語音材料而言,低於100hz或高於4000hz的部分基本上是與語言清晰程度無關的信號,降低或者完全去除這些信號,不會影響到語音清晰度。

    首先選擇要處理的音頻部分,點擊Effect菜單,選擇這裏的EQ and  Filters,找到Filter Curve EQ,然後點擊這個Preset & Settings按鈕,選擇Import導入預置文件,在剛才下載的預置文件中,找到Vocal-Filter,打開它。可以看到,我把100赫茲以下的信號完全濾掉了,這叫做低切處理。高頻4000到8000赫茲的部分有一點減弱,這一部分通常包含比較多的環境噪音,過強的齒音,口舌噪音等等,但這一部分如果衰減太多減少語音亮澤感,所以只做輕微的衰減。1萬赫茲以上的信號我做了一個高切,其實這部分包含一些氣聲,如果你的節目是那種需要很多氣聲的悄悄話節目,你可能需要手動提升一下,不過大部分語言類節目這樣高切是可以的。點擊這個Preview可以預覽處理以後的效果,如果覺得滿意可以點擊Apply按鈕對聲音進行處理。這裏需要註意一下,當你點擊Apply處理以後,你只能通過Edit菜單的Undo來取消它,你也可以對一段聲音重復多次進行處理,產生更加徹底的處理操作。

    現在這個聲音沒有了背景裏嗡嗡的聲音和噴麥的砰砰聲,唇齒口舌的雜音相對減弱了一些。 

    第二件事,動態壓縮

    動態就是指聲音最小的音量到最大的音量之間的範圍,對動態進行壓縮其實就是縮小最大音量和最小音量之間的差距。主要的目的是使整個聲音的聽起來響度比較穩定,而不會感覺忽大忽小,忽遠忽近。

    我們怎麽做呢,一種方法是直接選中一塊要調整的音頻,然後在Effect菜單選擇Volume and Compression,打開裏面的Amplify振幅調整,輸入負值減小音量,輸入正數增加音量,點擊Apply應用。這個辦法比較靈活,可以自由控製每一個細節,但是如果素材時長很長,變化很多,處理起來也比較費時。

    在Audacity中,我們可以使用壓縮器來進行處理。首先選中要處理的音頻,點擊Effect菜單,選擇Volume and Compression 音量和壓縮,找到Compressor,打開它,然後點擊Presets & settings按鈕,然後點擊Import導入,在剛才下載的預置文件中找到 VocalStartUp文件,打開,點擊Apply應用。

    經過處理後,聲音明顯變得更加響了,細節會稍微突出一些。如果發現處理以後有個別音量還是特被突出,可以使用前一種辦法單獨調整。

    使用壓縮器進行處理的時候,聲音除了動態範圍減小外,還會得到一些細節的提升,也會有音質的改變,多次使用壓縮器進行處理,可能會得到意外的效果,所以請仔細聽遍聲音的變化再決定是否要保留多次處理的效果。

    第三件事,失真處理

    大家可能聽過電吉他失真效果,這裏用到失真效果器來處理人聲,目的不是要追求電吉他效果,而是通過少量的失真效果,為音頻增加諧波成分,提升音頻的亮澤度和飽滿感。

    在Audacity中選擇Effect菜單,找到Distortion and Modulation菜單,從裏面選擇Distortion打開,點擊Presets & settings,然後點擊Import導入。在下載的預置文件裏找到Voice Exciton文件,打開。這裏幾個參數分別為Distortion Type: 失真類型選擇 Hard Clipping,Clipping Level設置在-6到-10之間,Drive 可以適當調整,默認50,Make-up Gain建議調到0,也就是不要額外增大音量。然後點擊Apply

    試聽一下。現在這個聲音聽起來大不一樣了。

    第五件事,響度標準化

    許多平臺和行業都有自己的節目響度標準。以YouTube為例,建議的響度不低於-14db LUFS,對於這個數值這裏就不多解釋了,直接操作。

    選中要處理的音頻,點擊Effect菜單,選擇Volume and Compression 音量和壓縮,找到Loudness Normalization ,打開它,然後點擊Presets & settings按鈕,然後點擊Import導入。在剛才下載的預置文件中找到 YouTube文件,打開它,點擊Apply。

    第六件事,限幅

    這是一個聲音安全化標準處理。通過Audacity的限製器 Limiter 來操作。選中要處理的音頻,點擊Effect菜單,選擇Volume and Compression 音量和壓縮,找到Limiter ,打開它,然後把Threshold(dB)調到-1.0,點擊Apply。

    點擊file菜單,選擇Export audio 導出音頻,。選擇Export to computer 導出到電腦裏,建議按照這裏的參數設定音頻文件格式。

    然後點擊Export 導出。

    到目前為止,你已經完成了全部的語音處理,聽聽是不是比之前更加響亮和清晰了呢?

    雖然整體聲音變響亮了,但是背景噪音也跟著一起被放大了,怎麽辦,感謝AI技術的發展,這裏給大家推薦一個超實用的在線去除環境噪音工具,叫做LALA AI.

    打開這個網址,點這裏可以選中文,然後點登陸,第一次選擇郵箱註冊或者其他登陸形式都可以。註冊並登陸成功以後,在產品中選擇聲音移除器,在這個頁面中,把音頻文件直接拖進來,稍等片刻,搞定。

    怎麽樣,有沒有專業大媒體那種感覺了?

    好了,今天的視頻就到這裏了,所有工具和鏈接我都放在這期視頻的網頁裏了,如果你在處理你的語音過程中遇到什麽問題歡迎在留言區提問,我會第一時間回答你。如果你覺得有所收獲,請別忘訂閱點贊收藏。感謝觀看!