我們在對音頻格式的轉換中，打交道最多的就是MP3了。如果你能徹底玩轉MP3，那么對你的音頻創作和對其他音頻格式的掌握會有很大的幫助。下面我們給大家介紹MP3制作軟件：LAME
要制作出高音質的MP3靠以前廣為流傳的MP3編碼器是不行的。LAME與一般MP3編碼器與眾不同，它支持幾乎所有能夠采用到MP3編碼中的技術，LAME支持CBR（固定碼率）和VBR（動態碼率，還有一個效果不是很出眾的ABR），LAME是MP3史上具有里程碑意義的軟件，LAME是一個Command line程序，象Dos程序一樣需要手工輸入，而且參數及其復雜，但可很方便的供其他程序調用，LAME同時也提供了一個DLL版本，但我們認為不如EXE版本的好，所以忽略不提。不要被LAME復雜的參數所嚇倒，文章中我們會提示如何操作來達到一勞永逸的效果。我們需要粗略的了解一下LAME的參數。
LAME其實真正要用到的參數就幾個而已。
VBR壓縮級別參數：[-V] 指定VBR的壓縮品質，范圍為0-9（數字越小品質越高），預設值為4。
碼率參數：[-b] 指定流量變動的下限，預設為32Kbps。[-B] 指定流量變動的上限，預設為320Kbps。注意 -b 和-B 的大小寫差異。如果使用在CBR編碼模式中，[-b]所指定的碼率就是固定碼率大小，可供指定的碼率大小可以為：16 24 32 40 48 56 64 80 96 112 128 160 192 224 256 320。
高品質編碼模式參數：[-h] 高品質編碼模式。這個選項在 VBR 壓縮模式中是預設開啟的。CBR編碼模式中是關閉的。
精度參數：[-q] 指定頻率資料量化時的精確度，范圍是為0-9（數字越小品質越高），預設值為2。如果在使用-q 0參數是覺得編碼速度慢得過份，請使用默認值。如果編碼的曲子是鋼琴或者小提琴、古箏二胡這類細節很豐富的樂器獨奏，我們推薦你就是耐着性子也要用-q 0參數，雖然慢點，但值得。
聲道模式參數：[-m] 立體聲壓縮模式，細分參數分別有 s:Stereo j:Joint Stereo f:Force ms_stereo m:Mono。當使用VBR編碼並把品質設為4-9和使用CBR編碼流量小於160 Kbps時，預設為j(Joint Stereo)。其余時候預設為s(Stereo)。
通過長期的使用，我們給出2個參數使用建議。
CBR 模式編碼的推薦參數：-b -m s -h （ 為碼率數值）。VBR 模式編碼推薦參數：-V 0。
在新版本的LAME中（3.90后），LAME提供了全新的--alt-preset系列預置參數，這組參數最大的好處就是不用再去記憶那些繁多的參數，而提供最佳化的選擇。
CBR模式：
--alt-preset insane 320kbps CBR模式，音質最好，體積最大。
VBR模式：
--alt-preset extreme 平均Bitrate范圍在192～256kbps之間，音質接近insane，體積小了一些，但比 －V 0 編碼效率要低。
--alt-preset系列參數提供比老參數更優秀的音質，但編碼效率卻低了很多，您需要更強勁的CPU支持才行，而相對比老參數提高相對不是很多，在乎您的取舍了，筆者傾向使用老參數。
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(轉)hifi級mp3制作和LAME參數設置2009-11-10 18:59:35
mp3也能hifi，hifi級mp3制作和LAME參數設置mp3也能HIF,hifi級mp3制作
對於我這樣的普通人來說，無損壓縮只能玩玩而已——雖然我的硬盤有160G，但是看到硬盤空間一個G一個G的減小，心里還是很不舒服。因此，我還是要聽MP3。
不要跟我提那些下載的128kbps MP3，大多數音質沒法聽。下面，我們請出的工具就是LAME。大家要問了，超級解霸等工具不是也可以壓MP3嗎？算了吧，一旦你使了LAME，這些軟件我保證你連看都不會再看一眼。那么，LAME有什么絕招呢？LAME的兩大神功就是VBR（動態流量編碼）和心理聲學模型。LAME可以說是將VBR的能力發揮到了極致。它將波形分割成50幀（30幀約1秒）一段，根據該段落內頻率的高低動態設置比特率，低頻使用相對低的比特率，高頻使用高比特率，這樣一來音質就得到了很大程度的保護。此外，LAME的心理聲學模型也是最出色的。就這樣，LAME將MP3的音質提高到了一個嶄新的階段，可以說LAME做出的MP3真正有着近似CD的音質了。但是LAME一開始只有命令行模式，使用不太方便，好在有人作出外殼程序，解決了這個問題。筆者現在使用的就是一個名為RazorLame的外殼，
首先我們設置一下LAME的參數，點擊LAME options。
里面有General, VBR, Advance和Expert等設置，要了解這些設置，我們還是需要首先了解一下LAME繁多的參數。
CBR（固定流量編碼）編碼時的基本參數：
CBR可以算是是最常用的的MP3編碼方式，其編碼流量可在32kbps-320kbps中選擇。我們從網上下載的MP3最常用的是128kbps，但是這個流量顯然是不夠的。如果你想做接近CD水准的MP3，推薦你用320Kbps的CBR（最高質量MP3），這類MP3音質最好，但是體積很大。如果你又想要小體積，那么還是不要用CBR了
-b參數：指定編碼的流量。LAME中可以使用的流量如下：
32 40 48 56 64 80 96 112 128 160 192 224 256 320。當然數字越大，體積越大，音質越好。這一點，體積與音質成正比。在波形靜音的部分，LAME會自動采用最小的流量。
-h參數：高品質編碼模式，可以增加音質，我們當然需要，一定要毫不猶豫用這個參數。這個選項在 VBR 壓縮模式中是預設開啟的。
-q參數：指定波形數據量化時的精確度，范圍為0-9，數字越低質量越好。筆者選擇2，因為LAME的開發者推薦這個參數。0理論上最好，但是開發者說這是個實驗型參數（不懂）。
因此，最強的MP3的命令行：-b 320 –h –q2。
VBR（動態流量編碼）編碼時的基本參數：
VBR編碼是LAME一大神功，可為你提供最佳的音質/體積比，所以筆者強烈推薦使用VBR。
-V參數：指定VBR的壓縮品質，范圍為0-9（數字越小品質越高），我們選擇2。
-b參數： 指定流量變動的下限，預設為32Kbps。使用預設就可以了。
-B參數： 指定流量變動的上限，預設為320Kbps。推薦使用預設值
其他如-q參數與CBR相同。
筆者推薦VBR命令：-V2 q2
此外LAME還提供一種ABR的編碼方式，這種編碼將CBR通過VBR的方式壓縮，可以指定流量大小，參數為—abr
然后是一些共同參數：
-m參數：選擇立體聲輸出方式：有-ms (Stereo 立體聲) -mj (Joint Stereo 聯合立體聲) –mm (Mono 單聲道)等4種可以選擇。
為了簡化LAME繁多的參數，開發者又提供一組強大的預制參數-ap供選擇。這類參數是以--alt –present開頭，因此，最好的參數又有了新的選擇：
CBR參數：--alt-preset insane或者--alt-preset cbr 320。音質最好，體積最大。
VBR參數：.--alt-preset extreme。音質很好，體積小，筆者推薦並使用這一參數。
然后我們回到LAME options，首先要到General中指定輸出的MP3文件存放位置。Advance中都是一些實驗性參數，有興趣可以試試，說不定可以試出什么新的最優化參數來，其中有一個 Delete source file after encoding 的選項，選取之后，編碼完成后原始的波形文件會被自動刪除，非常方便。然后是核心——VBR的設置。這里你可以通過上面學到的知識進行設置，不錯吧。再后就是Expert——專家設置。這里面有一個Custom options。可以自己直接寫命令行，但是這一項好像不是給專家設計的——更像給懶人使用的，你只要把筆者的推薦CBR或VBR參數拷貝上去，然后在底下only use custom options的選項前打上勾就可以了，真是方便。最后是Audio processing，注意output sampling frequency一定要選擇44.1KHz。默認為32KHz，會引起音質的下降。最后，點擊編碼（Encode）就可以開始了。再耐心等待幾分鍾，我們的HIFI級MP3就出爐了。
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LAME問與答——兼談最新的編碼參數設置方案
1.LAME是什么？
LAME是目前最好的MP3編碼引擎。LAME（mitiok.ma.cx）編碼出來的MP3音色純厚、空間寬廣、低音清晰、細節表現良好，它獨創的心理音響模型技術保證了CD音頻還原的真實性，配合VBR和ABR參數，音質幾乎可以媲美CD音頻，但文件體積卻非常小。對於一個免費引擎，LAME的優勢不言而喻。
2.上邊提到的VBR和ABR是什么？還有CBR？
VBR（Variable Bitrate）動態比特率。也就是沒有固定的比特率，壓縮軟件在壓縮時根據音頻數據即時確定使用什么比特率，這是以質量為前提兼顧文件大小的方式，推薦編碼模式；
ABR（Average Bitrate）平均比特率，是VBR的一種插值參數。LAME針對CBR不佳的文件體積比和VBR生成文件大小不定的特點獨創了這種編碼模式。ABR在指定的文件大小內，以每50幀（30幀約1秒）為一段，低頻和不敏感頻率使用相對低的流量，高頻和大動態表現時使用高流量，可以做為VBR和CBR的一種折衷選擇。
CBR（Constant Bitrate），常數比特率，指文件從頭到尾都是一種位速率。相對於VBR和ABR來講，它壓縮出來的文件體積很大，而且音質相對於VBR和ABR不會有明顯的提高。
3.下載的壓縮包里怎么有兩種格式的LAME文件？它們有什么區別？哪一種比較好？
LAME分DLL和EXE兩種版本，DLL版本做為一個方便的接口程序在大多數抓軌軟件中都能看到（比如AltoMP3Maker），但由於可控性差，與具備豐富調節參數的EXE版相比，其壓縮出來的MP3效果稍遜一籌。
4.怎么EXE版本是命令行方式運行的程序？太難用了
針對這一點，網上出現了一些EXE版的外殼程序，比如RazorLAME（www.dors.de/razorLAME），它是Win窗口程序，通過它可以使我們在視窗界面下輕松調整各種參數，使繁瑣的壓縮過程簡單化。我們也可以用直接用EAC（目前最好的抓軌軟件，www.exactaudiocopy.de）來調用LAME.exe，可以在抓軌同時壓縮MP3，事半功倍。
5.我在一些網站學會了使用-V 0 -q 0這樣的終極參數，這下可以壓出最高品質MP3了
實際上象-V 0 -q 0這樣的參數可以壓縮出最高品質MP3的說法從來都不是LAME開發者所應允的。在LAME中，象0、1這樣的Level屬於試驗參數，如果用它壓縮MP3，非但不會提高音質（相對於Level2而言），反而會導入多余的噪音，所以以上的參數應該改為-V 2 -q 2。實際上象這樣的參數標准幾近淘汰，-ap參數將做為新的LAME參數標准逐漸流行。
6.-ap參數？沒聽說過
這種參數屬於預置參數。
--abr 128 -h --nspsytune --athtype 2 --lowpass 16 --ns-bass -8 --scale 0.93，面對上邊這組微調參數你會不會有一種暈菜的感覺呢@_@……正是為了簡化參數設置，避免各種不必要的試驗參數，LAME開發者精心調配出了-ap參數，它是一組代碼級參數（也就是說沒有微調參數可以實現與它相同的功能）。使用這種新的預置參數標准既可以壓縮出更高品質的MP3，又可以避免我們陷入微調參數的迷宮中。以下是-ap參數列表：
最高品質參數：
--alt-preset insane或者--alt-preset cbr 320
320k CBR，音質最好，文件體積最大
VBR參數：
1.--alt-preset extreme
220-270k左右的VBR，音質與上面參數相仿，但文件體積小25%，推薦此參數
2.--alt-preset fast extreme
音質比上面參數稍微差一些
3.--alt-preset standard
180-220k左右的VBR，在音質和文件大小之間比較好的平衡
4.--alt-preset fast standard
音質比上面參數稍微差一些
5.--alt-preset standard -Y
雖然品質稍差，但文件體積非常小
ABR參數：
--alt-preset <Bitrate>
（可用Bitrate：80、96、112、128、160、192、224、256、320）
CBR參數:
--alt-preset cbr <Bitrate>
（可用Bitrate：80、96、112、128、160、192、224、256、320）
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對MP3及音頻壓縮技術的一些誤解
1、mp3的音質很差？
錯。mp3作為當前音頻有損壓縮的“王者”，它的編碼技術已經幾近完美。很多人只是不清楚如何才能壓縮出高品質的mp3而已。2001年12月，世界上最優秀的mp3編碼器--LAME推出了革命性的版本3.90.2，針對lame壓縮參數過於煩瑣的情況，提供了幾個preset（預設）參數。現在只要使用LAME的standard（標准）模式進行壓縮，就能得到近似於CD的完美音質。
2、128kbps的mp3=CD音質？
錯。首先，所謂CD音質是一個帶有很大主觀性的名詞，基本上可以認為CD音質意味着在平均水平的聽音條件下能達到用光驅放CD的效果。但是根據這個定義，無數的試聽結果表明，不管用什么編碼器，什么樣的設置，128kbps的mp3都不能達到這個標准。關於這方面的主題可參http://ff123.net/,這是一個非常著名的國外音頻站點，對128kbps的mp3的測試有非常詳細的理論闡述。
3、mp3 192kbps CBR(固定比特速率) stereo(立體聲)編碼是音質與文件大小的最佳平衡設置？
錯。這一誤解有很深的根源。因為128kbps的mp3在音質上不能被“苛刻”的音樂愛好者接受，所以他們要尋求更好的設置。對Xing編碼器及Fraunhofer編碼器來說，直到現在它們在VBR(可變比特速率)和jointstereo(混合立體聲)的算法上都很失敗，所以很多人都認為CBR和stereo才是最佳的選擇，而且192kbps的mp3在文件大小上也是可以接受的。是LAME編碼器改變了這一切！LAME采用的VBR及智能的joint stereo算法非常優秀，已經沒什么理由再去使用CBR和stereo--這樣做只會浪費有限的bits。標准的VBR預定設置（即使用--alt-preset standard參數）生成的mp3文件的平均比特率也是192kbps，但它的音質要好過CBR 192kbps，在同等的比特率下其他的編碼器非其敵手（按：除了1、mpc--其音質在該bitrate左右好於mp3, 2、最近的oggenc 1.0--not tested yet）。
4、mp3 320kbps CBR Stereo是mp3音質的極限？
錯（或者說Not exactly true）。雖然320kbps是mp3標准的極限，但在320kbps下使用設計良好的Joint Stereo，能夠將節省下下的bits用於純粹的音樂部分（從而提高音質）。如果音源的立體聲分離度很低，使用完全的stereo是一種浪費。
5、VBR的音質不如CBR？
錯。設計良好的VBR算法不會將bits浪費在易於編碼的部分，節省下來的bits將用在對復雜的音頻部分進行編碼。這一誤解可能來自於較老的FhG Encoder的VBR算法及Xing VBR算法中存在的bug,對當前的lame編碼器來說,它的VBR算法已被協調得很好,不會有音質上的問題。
6、Joint Stereo 音質不佳？
錯。當前主流的encoder如lame、mppenc、oggenc、aacenc都使用了所謂smart joint stereo的技術，不會破壞stereo image,請參閱如下的兩個鏈接（E文,由編碼器的開發者解答）：
http://www.hydrogenaudio.org/forums/showthread.php?s=&threadid=1081 ;
http://www.hydrogenaudio.org/forums/showthread.php?s=&threadid=759 ;
更為技術性的解釋如下：
http://www.xiph.org/ogg/vorbis/doc/stereo.html ;
7、Blade是最佳的mp3編碼器？
錯。（似乎不用過多的解釋）Blade不推薦用於所有bitrate的mp3編碼，由於缺少相當多的功能，其音質較lame或FhG遜色很多。下面的兩個鏈接有助於了解blade的缺憾：
http://forums.afterdawn.com/thread_view.cfm/1914 ;
http://www.hydrogenaudio.org/forums/showthread.php?s=&threadid=463 ;
最新消息——Blade已經停止開發，其作者在主頁上聲明ogg是更好的選擇
8、wma在64kbps可達CD音質？
錯。不用我多費筆墨，不相信的話點擊下面的鏈接了解詳情:：
http://www.hydrogenaudio.org/forums/showthread.php?s=&threadid=1434 ;
http://forums.winamp.com/showthread.php?s=&threadid=89378 ;
另外，專門為winamp寫plugin的Peter也寫了篇文章:
Why not to use wma http://205.188.228.81/showthread.php?threadid=81838)
9、不同的音樂類型需要不同的編碼器及不同的參數?
錯。編碼器是在音頻信號級進行處理，不去分辨音樂類型。只要心理學模型與編碼算法正確，同一設置就適用於所有的音樂類型。詳情參見：
http://www.hydrogenaudio.org/forums/showthread.php?s=&threadid=1835
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小身材也要大味道——128kbps下如何設置Lame編碼參數
Lame MP3編碼引擎大家已經相當熟悉了，而且在APX參數推出以后，它的使用變得更加方便。但是很多朋友還是反映，Lame壓縮出來的MP3體積還是大了一點，降低壓縮波特比又怕效果不好，那么如何在底碼率下用Lame壓出效果相對比較好的曲目呢?
其實一般來說，128kbps的編碼率下，任何編碼器都無法達到CD音質（M$所言，WMA在64kbps或96kpbs就能達到CD Quality是一個真實的謊言），對Lame來說，要想在128kbps超過那些專門為低bitrate作了優化的encoder如mp3pro、wma甚至ogg，冗長的參數是不可或缺的，這篇短文就為您進行詳細的解釋
1、Lame的版本的問題
Lame.exe的當前的最新穩定版是3.92，很多地方都可以提供下載，推薦使用。不過還有一個版本就是dibrom（Lame preset參數的開發者）編譯的3.90.2，Lame隨后的3.91、3.92版本有相當部分（特別是preset部分）是脫胎於此版的。這也是當前在preset參數設置下編碼最快的版本，下載鏈接如下http://www.hydrogenaudio.org/extra/Lame/Lame3.90.2-ICL.zip ;
Lame的開發速度很快，3.93的alpha版已經出過十幾個了。雖然內部測試版不推薦使用，但它的確修正了不少的錯誤（像對人們誤解最大的q0參數的修正），所以也提供一個下載鏈接，有興趣的朋友不妨一試:http://mitiok.free.fr/Lame-20020706.zip（這是最新7月6日版）。
2、參數設置
Lame的參數設置的爭論是最大的，我也有被千夫所指的經歷和准備……。下面的文字都是我在r3mix和Hydrogen論壇得來的信息的綜合:
a、對CBR：
--alt-preset cbr 128 或者
-h --nspsytune --athtype 2 --lowpass 16 --ns-bass -8 --scale 0.93
b、對ABR：
--alt-preset 128(該preset與--abr 128 -h --nspsytune --athtype 2 --lowpass 17.5 --ns-bass -6 --scale 0.93基本相當)
--abr 128 -h --nspsytune --athtype 2 --lowpass 16 --ns-bass -8 --scale 0.93
c、對VBR：
在128kbps下VBR沒有用武之地。
就音質來說，我認為，ABR>CBR。
小結：
r3mix論壇曾有一句話讓我印象很深刻: one can't talk about Lame without mentioning the version and settings. Lame的參數之多很為人詬病，preset的出現對懶人如我者是最大的福音，雖然128kbps不是我喜歡的bitrate，但不可否認這是internet上最流行的……。好像主題已經有點亂了，就此打住. 獨樂樂不如眾樂樂，讓我們一起研究、共享我們的知識，我們的音樂。
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（轉）lame 3.90.3 轉換mp3的所有參數2009-11-11 12:28:13| 分類： 默認分類 閱讀315 評論0 字號：大中小訂閱
LAME version 3.90.3 MMX (http://www.mp3dev.org/)
-- Compiled at http://www.hydrogenaudio.org
-- Check this website for up to date information on the --alt-presets
usage: lame [options] <infile> [outfile]
<infile> and/or <outfile> can be "-", which means stdin/stdout.
RECOMMENDED:
lame -h input.wav output.mp3
OPTIONS:
Input options:
-r input is raw pcm
-x force byte-swapping of input
-s sfreq sampling frequency of input file (kHz) - default 44.1 kHz
--bitwidth w input bit width is w (default 16)
--mp1input input file is a MPEG Layer I file
--mp2input input file is a MPEG Layer II file
--mp3input input file is a MPEG Layer III file
--nogap <file1> <file2> <...>
gapless encoding for a set of contiguous files
--nogapout <dir>
output dir for gapless encoding (must precede --nogap)
Operational options:
-m <mode> (s)tereo【立體聲】, (j)oint【聯合立體聲】, (f)orce, (m)ono or (a)auto
default is (s) or (j) depending on bitrate
force = force ms_stereo on all frames.
auto = jstereo, with varialbe mid/side threshold
-a downmix from stereo to mono file for mono encoding
-d allow channels to have different blocktypes
--freeformat produce a free format bitstream
--decode input=mp3 file, output=wav
-t disable writing wav header when using --decode
--comp <arg> choose bitrate to achive a compression ratio of <arg>
--scale <arg> scale input (multiply PCM data) by <arg>
--scale-l <arg> scale channel 0 (left) input (multiply PCM data) by <arg>
--scale-r <arg> scale channel 1 (right) input (multiply PCM data) by <arg>
--preset type type must be phone, voice, fm, tape, hifi, cd or studio
"--preset help" gives some more infos on these
--alt-preset type type must be "standard", "extreme", "insane",
or a value for an average desired bitrate and depending on
the value specified, appropriate quality settings will be
used.
--r3mix use high-quality VBR preset
Verbosity:
--disptime <arg>print progress report every arg seconds
-S don't print progress report, VBR histograms
--nohist disable VBR histogram display
--silent don't print anything on screen
--quiet don't print anything on screen
--verbose print a lot of useful information
Noise shaping & psycho acoustic algorithms:
-q <arg> <arg> = 0...9. Default -q 5
-q 0: Highest quality, very slow
-q 9: Poor quality, but fast
-h Same as -q 2. Recommended.
-f Same as -q 7. Fast, ok quality
CBR (constant bitrate, the default) options:
-b <bitrate> set the bitrate in kbps, default 128 kbps
ABR options:
--abr <bitrate> specify average bitrate desired (instead of quality)
VBR options:
-v use variable bitrate (VBR) (--vbr-old)
--vbr-old use old variable bitrate (VBR) routine
--vbr-new use new variable bitrate (VBR) routine
-V n quality setting for VBR. default n=4
0=high quality,bigger files. 9=smaller files
-b <bitrate> specify minimum allowed bitrate, default 32 kbps
-B <bitrate> specify maximum allowed bitrate, default 320 kbps
-F strictly enforce the -b option, for use with players that
do not support low bitrate mp3
-t disable writing LAME Tag
ATH related:
--noath turns ATH down to a flat noise floor
--athshort ignore GPSYCHO for short blocks, use ATH only
--athonly ignore GPSYCHO completely, use ATH only
--athtype n selects between different ATH types [0-5]
--athlower x lowers ATH by x dB
--athaa-type n ATH auto adjust types 1-3, else no adjustment
--athaa-loudapprox n n=1 total energy or n=2 equal loudness curve
--athaa-sensitivity x activation offset in -/+ dB for ATH auto-adjustment
PSY related:
--short use short blocks when appropriate
--noshort do not use short blocks
--allshort use only short blocks
--cwlimit <freq> compute tonality up to freq (in kHz) default 8.8717
--notemp disable temporal masking effect
--nspsytune experimental PSY tunings by Naoki Shibata
--nssafejoint M/S switching criterion
--nsmsfix <arg> M/S switching tuning [effective 0-3.5]
--ns-bass x adjust masking for sfbs 0 - 6 (long) 0 - 5 (short)
--ns-alto x adjust masking for sfbs 7 - 13 (long) 6 - 10 (short)
--ns-treble x adjust masking for sfbs 14 - 21 (long) 11 - 12 (short)
--ns-sfb21 x change ns-treble by x dB for sfb21
experimental switches:
-X n selects between different noise measurements
-Y lets LAME ignore noise in sfb21, like in CBR
MP3 header/stream options:
-e <emp> de-emphasis n/5/c (obsolete)
-c mark as copyright
-o mark as non-original
-p error protection. adds 16 bit checksum to every frame
(the checksum is computed correctly)
--nores disable the bit reservoir
--strictly-enforce-ISO comply as much as possible to ISO MPEG spec
Filter options:
-k keep ALL frequencies (disables all filters),【保留所有頻率，不使用過濾】
Can cause ringing and twinkling
--lowpass <freq> frequency(kHz), lowpass filter cutoff above freq
--lowpass-width <freq> frequency(kHz) - default 15% of lowpass freq
--highpass <freq> frequency(kHz), highpass filter cutoff below freq
--highpass-width <freq> frequency(kHz) - default 15% of highpass freq
--resample <sfreq> sampling frequency of output file(kHz)- default=automatic
ID3 tag options:
--tt <title> audio/song title (max 30 chars for version 1 tag)
--ta <artist> audio/song artist (max 30 chars for version 1 tag)
--tl <album> audio/song album (max 30 chars for version 1 tag)
--ty <year> audio/song year of issue (1 to 9999)
--tc <comment> user-defined text (max 30 chars for v1 tag, 28 for v1.1)
--tn <track> audio/song track number (1 to 255, creates v1.1 tag)
--tg <genre> audio/song genre (name or number in list)
--add-id3v2 force addition of version 2 tag
--id3v1-only add only a version 1 tag
--id3v2-only add only a version 2 tag
--space-id3v1 pad version 1 tag with spaces instead of nulls
--pad-id3v2 pad version 2 tag with extra 128 bytes
--genre-list print alphabetically sorted ID3 genre list and exit
Note: A version 2 tag will NOT be added unless one of the input fields
won't fit in a version 1 tag (e.g. the title string is longer than 30
characters), or the '--add-id3v2' or '--id3v2-only' options are used,
or output is redirected to stdout.
MPEG-1 layer III sample frequencies (kHz): 32 48 44.1
bitrates (kbps): 32 40 48 56 64 80 96 112 128 160 192 224 256 320
MPEG-2 layer III sample frequencies (kHz): 16 24 22.05
bitrates (kbps): 8 16 24 32 40 48 56 64 80 96 112 128 144 160
MPEG-2.5 layer III sample frequencies (kHz): 8 12 11.025
bitrates (kbps): 8 16 24 32 40 48 56 64 80 96 112 128 144 160
我個人在foobar0.83中的lame里使用的參數為：
-m s -q 0 -b 320 --noath -k - %d
LAME其實真正要用到的參數就幾個而已。
VBR壓縮級別參數：[-V] 指定VBR的壓縮品質，范圍為0-9（數字越小品質越高），預設值為4。
碼率參數：[-b] 指定流量變動的下限，預設為32Kbps。[-B] 指定流量變動的上限，預設為320Kbps。注意 -b 和-B 的大小寫差異。如果使用在CBR編碼模式中，[-b]所指定的碼率就是固定碼率大小，可供指定的碼率大小可以為：16 24 32 40 48 56 64 80 96 112 128 160 192 224 256 320。
高品質編碼模式參數：[-h] 高品質編碼模式。這個選項在 VBR 壓縮模式中是預設開啟的。CBR編碼模式中是關閉的。
精度參數：[-q] 指定頻率資料量化時的精確度，范圍是為0-9（數字越小品質越高），預設值為2。如果在使用-q 0參數是覺得編碼速度慢得過份，請使用默認值。如果編碼的曲子是鋼琴或者小提琴、古箏二胡這類細節很豐富的樂器獨奏，我們推薦你就是耐着性子也要用-q 0參數，雖然慢點，但值得。
聲道模式參數：[-m] 立體聲壓縮模式，細分參數分別有 s:Stereo j:Joint Stereo f:Force ms_stereo m:Mono。當使用VBR編碼並把品質設為4-9和使用CBR編碼流量小於160 Kbps時，預設為j(Joint Stereo)。其余時候預設為s(Stereo)。
通過長期的使用，我們給出2個參數使用建議。
CBR 模式編碼的推薦參數：-b -m s -h （ 為碼率數值）。VBR 模式編碼推薦參數：-V 0。
在新版本的LAME中（3.90后），LAME提供了全新的--alt-preset系列預置參數，這組參數最大的好處就是不用再去記憶那些繁多的參數，而提供最佳化的選擇。
CBR模式：
--alt-preset insane 320kbps CBR模式，音質最好，體積最大。
VBR模式：
--alt-preset extreme 平均Bitrate范圍在192～256kbps之間，音質接近insane，體積小了一些，但比 －V 0 編碼效率要低。
--alt-preset系列參數提供比老參數更優秀的音質，但編碼效率卻低了很多，您需要更強勁的CPU支持才行，而相對比老參數提高相對不是很多，在乎您的取舍了，筆者傾向使用老參數。
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lame3.90.3 Full command line switch reference2009-11-11 13:06:58| 分類： 默認分類 閱讀17 評論0 字號：大中小 訂閱
Full command line switch reference
note: Options which could exist without beeing documented here are considered as experimental ones. Such experimental options should usually not be used.
| switch | parameter |
| -a | downmix stereo file to mono |
| --abr | average bitrate encoding |
| --allshort | use short blocks only |
| --athlower | lower the ATH |
| --athonly | ATH only |
| --athshort | ATH only for short blocks |
| --athtype | select ATH type |
| -b | bitrate (8...320) |
| -B | max VBR/ABR bitrate (8...320) |
| --bitwidth | input bit width |
| -c | copyright |
| --comp | choose compression ratio |
| --cwlimit | tonality limit |
| -d | block type control |
| --decode | decoding only |
| --disptime | time between display updates |
| -e | de-emphasis (n, 5, c) |
| -f | fast mode |
| -F | strictly enforce the -b option |
| --freeformat | free format bitstream |
| -h | high quality |
| --help | help |
| --highpass | highpass filtering frequency in kHz |
| --highpass-width | width of highpass filtering in kHz |
| -k | full bandwidth |
| --lowpass | lowpass filtering frequency in kHz |
| --lowpass-width | width of lowpass filtering in kHz |
| -m | stereo mode (s, j, f, m) |
| --mp1input | MPEG Layer I input file |
| --mp2input | MPEG Layer II input file |
| --mp3input | MPEG Layer III input file |
| --noath | disable ATH |
| --nohist | disable histogram display |
| --nores | disable bit reservoir |
| --noshort | disable short blocks frames |
| --notemp | disable temporal masking |
| -o | non-original |
| -p | error protection |
| --preset | use built-in preset |
| --alt-preset | use updated and much higher quality "alternate" presets |
| --priority | OS/2 process priority control |
| -q | algorithm quality selection |
| --quiet | silent operation |
| -r | input file is raw pcm |
| --resample | output sampling frequency in kHz (encoding only) |
| --r3mix | r3mix VBR preset |
| -s | sampling frequency in kHz |
| -S | silent operation |
| --scale | scale input |
| --scale-l | scale input channel 0 (left) |
| --scale-r | scale input channel 1 (right) |
| --short | use short blocks |
| --silent | silent operation |
| --strictly-enforce-ISO | strict ISO compliance |
| -t | disable INFO/WAV header |
| -V | VBR quality setting (0...9) |
| --vbr-new | new VBR mode |
| --vbr-old | older VBR mode |
| --verbose | verbosity |
| -x | swapbytes |
| -X | change quality measure |
* -a downmix
Mix the stereo input file to mono and encode as mono.
The downmix is calculated as the sum of the left and right channel, attenuated by 6 dB.
This option is only needed in the case of raw PCM stereo input (because LAME cannot determine the number of channels in the input file).
To encode a stereo PCM input file as mono, use "lame -m s -a".
For WAV and AIFF input files, using "-m m" will always produce a mono .mp3 file from both mono and stereo input.
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* --abr n average bitrate encoding
Turns on encoding with a targeted average bitrate of n kbits, allowing to use frames of different sizes. The allowed range of n is 8-310, you can use any integer value within that range.
It can be combined with the -b and -B switches like:
lame --abr 123 -b 64 -B 192 a.wav a.mp3
which would limit the allowed frame sizes between 64 and 192 kbits.
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* --allshort use short blocks only
Use only short blocks, no long ones.
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* --athlower n lower the ATH
Lower the ATH (absolute threshold of hearing) by n dB.
Normally, humans are unable to hear any sound below this threshold, but for music recorded at very low level this option might be usefull.
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* --athonly ATH only
This option causes LAME to ignore the output of the psy-model and only use masking from the ATH (absolute threshold of hearing). Might be useful at very high bitrates or for testing the ATH.
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* --athshort ATH only for short blocks
Ignore psychoacoustic model for short blocks, use ATH only.
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* --athtype 0/1/2 select ATH type
The Absolute Threshold of Hearing is the minimum threshold under which humans are unable to hear any sound. In the past, LAME was using ATH shape 0 which is the Painter & Spanias formula. Tests have shown that this formula is innacurate for the 13-22 kHz area, leading to audible artifacts in some cases. Shape 1 was thus implemented, which is over sensitive, leading to very high bitrates. Shape 2 formula was accurately modelized from real data in order to real optimal quality while not wasting bitrate. In CBR and ABR modes, LAME uses ATH shape 2 by default.
In VBR mode, LAME is adapting its shape according to the -V value, going gradually from the 0 shape at -V9 up to shape 2 at -V0.
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* -b n bitrate
For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)
n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320
For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)
n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160
Default is 128 kbs for MPEG1 and 64 kbs for MPEG2.
When used with variable bitrate encoding (VBR), -b specifies the minimum bitrate to be used. However, in order to avoid wasted space, the smallest frame size available will be used during silences.
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* -B n maximum VBR/ABR bitrate
For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)
n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320
For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)
n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160
Specifies the maximum allowed bitrate when using VBR/ABR
The use of -B is NOT RECOMMENDED. A 128kbs CBR bitstream, because of the bit reservoir, can actually have frames which use as many bits as a 320kbs frame. VBR modes minimize the use of the bit reservoir, and thus need to allow 320kbs frames to get the same flexibility as CBR streams.
note: If you own an mp3 hardware player build upon a MAS 3503 chip, you must set maximum bitrate to no more than 224 kpbs.
* --bitwidth 8/16/24/32 input bit width
Required only for raw PCM input files. Otherwise it will be determined from the header of the input file.
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* -c copyright
Mark the encoded file as being copyrighted.
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* --comp choose compression ratio
Instead of choosing bitrate, using this option, user can choose compression ratio to achieve.
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* --cwlimit n tonality limit
Compute tonality up to freq (in kHz). Default setting is 8.8717.
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* -d block type control
Allows the left and right channels to use different block size types.
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* --decode decoding only
Uses LAME for decoding to a wav file. The input file can be any input type supported by encoding, including layer I,II,III (MP3) and OGG files. In case of MPEG files, LAME uses a bugfixed version of mpglib for decoding.
If -t is used (disable wav header), Lame will output raw pcm in native endian format. You can use -x to swap bytes order.
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* --disptime n time between display updates
Set the delay in seconds between two display updates.
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* -e n/5/c de-emphasis
n = (none, default)
5 = 0/15 microseconds
c = citt j.17
All this does is set a flag in the bitstream. If you have a PCM input file where one of the above types of (obsolete) emphasis has been applied, you can set this flag in LAME. Then the mp3 decoder should de-emphasize the output during playback, although most decoders ignore this flag.
A better solution would be to apply the de-emphasis with a standalone utility before encoding, and then encode without -e.
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* -f fast mode
This switch forces the encoder to use a faster encoding mode, but with a lower quality. The behaviour is the same as the -q7 switch.
Noise shaping will be disabled, but psycho acoustics will still be computed for bit allocation and pre-echo detection.
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* -F strictly enforce the -b option
This is mainly for use with hardware players that do not support low bitrate mp3.
Without this option, the minimum bitrate will be ignored for passages of analog silence, ie when the music level is below the absolute threshold of human hearing (ATH).
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* --freeformat free format bitstream
Produces a free format bitstream. With this option, you can use -b with any bitrate higher than 8 kbps.
However, even if an mp3 decoder is required to support free bitrates at least up to 320 kbps, many players are unable to deal with it.
Tests have shown that the following decoders support free format:
FreeAmp up to 440 kbps
in_mpg123 up to 560 kbps
l3dec up to 310 kbps
LAME up to 560 kbps
MAD up to 640 kbps
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* -h high quality
Use some quality improvements. Encoding will be slower, but the result will be of higher quality. The behaviour is the same as the -q2 switch.
This switch is always enabled when using VBR.
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* --help help
Display a list of all available options.
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* --highpass highpass filtering frequency in kHz
Set an highpass filtering frequency. Frequencies below the specified one will be cutoff.
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* --highpass-width width of highpass filtering in kHz
Set the width of the highpass filter. The default value is 15% of the highpass frequency.
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* -k full bandwidth
Tells the encoder to use full bandwidth and to disable all filters. By default, the encoder uses some highpass filtering at low bitrates, in order to keep a good quality by giving more bits to more important frequencies.
Increasing the bandwidth from the default setting might produce ringing artefacts at low bitrates. Use with care!
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* --lowpass lowpass filtering frequency in kHz
Set a lowpass filtering frequency. Frequencies above the specified one will be cutoff.
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* --lowpass-width width of lowpass filtering in kHz
Set the width of the lowpass filter. The default value is 15% of the lowpass frequency.
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* -m s/j/f/d/m stereo mode
Joint-stereo is the default mode for stereo files with VBR when -V is more than 4 or fixed bitrates of 160kbs or less. At higher fixed bitrates or higher VBR settings, the default is stereo.
stereo
In this mode, the encoder makes no use of potentially existing correlations between the two input channels. It can, however, negotiate the bit demand between both channel, i.e. give one channel more bits if the other contains silence or needs less bits because of a lower complexity.
joint stereo
In this mode, the encoder will make use of a correlation between both channels. The signal will be matrixed into a sum ("mid"), computed by L+R, and difference ("side") signal, computed by L-R, and more bits are allocated to the mid channel.
This will effectively increase the bandwidth if the signal does not have too much stereo separation, thus giving a significant gain in encoding quality.
Using mid/side stereo inappropriately can result in audible compression artifacts. To much switching between mid/side and regular stereo can also sound bad. To determine when to switch to mid/side stereo, LAME uses a much more sophisticated algorithm than that described in the ISO documentation, and thus is safe to use in joint stereo mode.
forced joint stereo
This mode will force MS joint stereo on all frames. It's slightly faster than joint stereo, but it should be used only if you are sure that every frame of the input file has very little stereo separation.
dual channels
In this mode, the 2 channels will be totally indenpendently encoded. Each channel will have exactly half of the bitrate. This mode is designed for applications like dual languages encoding (ex: English in one channel and French in the other). Using this encoding mode for regular stereo files will result in a lower quality encoding.
mono
The input will be encoded as a mono signal. If it was a stereo signal, it will be downsampled to mono. The downmix is calculated as the sum of the left and right channel, attenuated by 6 dB.
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* --mp1input MPEG Layer I input file
Assume the input file is a MPEG Layer I file.
If the filename ends in ".mp1" or ".mpg" LAME will assume it is a MPEG Layer I file. For stdin or Layer I files which do not end in .mp1 or .mpg you need to use this switch.
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* --mp2input MPEG Layer II input file
Assume the input file is a MPEG Layer II (ie MP2) file.
If the filename ends in ".mp2" LAME will assume it is a MPEG Layer II file. For stdin or Layer II files which do not end in .mp2 you need to use this switch.
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* --mp3input MPEG Layer III input file
Assume the input file is a MP3 file. Usefull for downsampling from one mp3 to another. As an example, it can be usefull for streaming through an IceCast server.
If the filename ends in ".mp3" LAME will assume it is an MP3 file. For stdin or MP3 files which do not end in .mp3 you need to use this switch.
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* --noath disable ATH
Disable any use of the ATH (absolute threshold of hearing) for masking. Normally, humans are unable to hear any sound below this threshold.
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* --nohist disable histogram display
By default, LAME will display a bitrate histogram while producing VBR mp3 files. This will disable that feature.
Histogram display might not be available on your release.
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* --nores disable bit reservoir
Disable the bit reservoir. Each frame will then become independent from previous ones, but the quality will be lower.
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* --noshort disable short blocks frames
Encode all frames using long blocks only. This could increase quality when encoding at very low bitrates, but can produce serious pre-echo artefacts.
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* --notemp disable temporal masking
Don't make use of the temporal masking effect.
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* -o non-original
Mark the encoded file as being a copy.
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* -p error protection
Turn on CRC error protection.
It will add a cyclic redundancy check (CRC) code in each frame, allowing to detect transmission errors that could occur on the MP3 stream. However, it takes 16 bits per frame that would otherwise be used for encoding, and then will slightly reduce the sound quality.
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* --preset presetName use built-in preset
Use one of the built-in presets (phone, phon+, lw, mw-eu, mw-us, sw, fm, voice, radio, tape, hifi, cd, studio).
"--preset help" gives more information about the used options in these presets.
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* --alt-preset presetName use updated and much higher quality "alternate" presets
Use one of the built-in alternate presets (standard, fast standard, extreme, fast extreme, insane, or the abr/cbr modes).
"--alt-preset help" gives more information about the usage possibilities for these presets.
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* --priority 0...4 OS/2 process priority control
With this option, LAME will run with a different process priority under IBM OS/2.
This will greatly improve system responsiveness, since OS/2 will have more free time to properly update the screen and poll the keyboard/mouse. It should make quite a difference overall, especially on slower machines. LAME's performance impact should be minimal.
0 (Low priority)
Priority 0 assumes "IDLE" class, with delta 0.
LAME will have the lowest priority possible, and the encoding may be suspended very frequently by user interaction.
1 (Medium priority)
Priority 1 assumes "IDLE" class, with delta +31.
LAME won't interfere at all with what you're doing.
Recommended if you have a slower machine.
2 (Regular priority)
Priority 2 assumes "REGULAR" class, with delta -31.
LAME won't interfere with your activity. It'll run just like a regular process, but will spare just a bit of idle time for the system. Recommended for most users.
3 (High priority)
Priority 3 assumes "REGULAR" class, with delta 0.
LAME will run with a priority a bit higher than a normal process.
Good if you're just running LAME by itself or with moderate user interaction.
4 (Maximum priority)
Priority 4 assumes "REGULAR" class, with delta +31.
LAME will run with a very high priority, and may interfere with the machine response.
Recommended if you only intend to run LAME by itself, or if you have a fast processor.
Priority 1 or 2 is recommended for most users.
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* -q 0..9 algorithm quality selection
Bitrate is of course the main influence on quality. The higher the bitrate, the higher the quality. But for a given bitrate, we have a choice of algorithms to determine the best scalefactors and huffman encoding (noise shaping).
-q 0: use slowest & best possible version of all algorithms. -q 0 and -q 1 are slow and may not produce significantly higher quality.
-q 2: recommended. Same as -h.
-q 5: default value. Good speed, reasonable quality.
-q 7: same as -f. Very fast, ok quality. (psycho acoustics are used for pre-echo & M/S, but no noise shaping is done.
-q 9: disables almost all algorithms including psy-model. poor quality.
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* -r input file is raw pcm
Assume the input file is raw pcm. Sampling rate and mono/stereo/jstereo must be specified on the command line. Without -r, LAME will perform several fseek()'s on the input file looking for WAV and AIFF headers.
Might not be available on your release.
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* --resample 8/11.025/12/16/22.05/24/32/44.1/48 output sampling frequency in kHz
Select ouptut sampling frequency (for encoding only).
If not specified, LAME will automatically resample the input when using high compression ratios.
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* --r3mix r3mix VBR preset
Uses r3mix VBR preset.
See www.r3mix.net for more details.
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* -s 8/11.025/12/16/22.05/24/32/44.1/48 sampling frequency
Required only for raw PCM input files. Otherwise it will be determined from the header of the input file.
LAME will automatically resample the input file to one of the supported MP3 samplerates if necessary.
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* -S / --silent / --quiet silent operation
Don't print progress report.
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* --scale n scales input by n
* --scale-l n scales input channel 0 (left) by n
* --scale-r n scales input channel 1 (right) by n
Scales input by n. This just multiplies the PCM data (after it has been converted to floating point) by n.
n > 1: increase volume
n = 1: no effect
n < 1: reduce volume
Use with care, since most MP3 decoders will truncate data which decodes to values greater than 32768.
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* --short use short blocks
Let LAME use short blocks when appropriate. It is the default setting.
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* --strictly-enforce-ISO strict ISO compliance
With this option, LAME will enforce the 7680 bit limitation on total frame size.
This results in many wasted bits for high bitrate encodings but will ensure strict ISO compatibility. This compatibility might be important for hardware players.
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* -t disable INFO/WAV header
Disable writing of the INFO Tag on encoding.
This tag in embedded in frame 0 of the MP3 file. It includes some information about the encoding options of the file, and in VBR it lets VBR aware players correctly seek and compute playing times of VBR files.
When '--decode' is specified (decode to WAV), this flag will disable writing of the WAV header. The output will be raw pcm, native endian format. Use -x to swap bytes.
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* -V 0...9 VBR quality setting
Enable VBR (Variable BitRate) and specifies the value of VBR quality.
default=4
0=highest quality.
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* --vbr-new new VBR mode
Invokes the newest VBR algorithm. During the development of version 3.90, considerable tuning was done on this algorithm, and it is now considered to be on par with the original --vbr-old.
It has the added advantage of being very fast (over twice as fast as --vbr-old).
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* --vbr-old older VBR mode
Invokes the oldest, most tested VBR algorithm. It produces very good quality files, though is not very fast. This has, up through v3.89, been considered the "workhorse" VBR algorithm.
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* --verbose verbosity
Print a lot of information on screen.
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* -x swapbytes
Swap bytes in the input file or ouptut file when using --decode.
For sorting out little endian/big endian type problems. If your encodings sounds like static, try this first.
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* -X 0...7 change quality measure
When LAME searches for a "good" quantization, it has to compare the actual one with the best one found so far. The comparison says which one is better, the best so far or the actual. The -X parameter selects between different approaches to make this decision, -X0 beeing the default mode:
-X0
The criterions are (in order of importance):
* less distorted scalefactor bands
* the sum of noise over the thresholds is lower
* the total noise is lower
-X1
The actual is better if the maximum noise over all scalefactor bands is less than the best so far .
-X2
The actual is better if the total sum of noise is lower than the best so far.
-X3
The actual is better if the total sum of noise is lower than the best so far and the maximum noise over all scalefactor bands is less than the best so far plus 2db.
-X4
Not yet documented.
-X5
The criterions are (in order of importance):
* the sum of noise over the thresholds is lower
* the total sum of noise is lower
-X6
The criterions are (in order of importance):
* the sum of noise over the thresholds is lower
* the maximum noise over all scalefactor bands is lower
* the total sum of noise is lower
-X7
The criterions are:
* less distorted scalefactor bands
or
* the sum of noise over the thresholds is lower
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