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我們在對音頻格式的轉(zhuǎn)換中,，打交道最多的就是MP3了,。如果你能徹底玩轉(zhuǎn)MP3,，那么對你的音頻創(chuàng)作和對其他音頻格式的掌握會(huì)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(zhì)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(shù)，LAME支持CBR（固定碼率）和VBR（動(dòng)態(tài)碼率,，還有一個(gè)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(gè)Command line程序,，象Dos程序一樣需要手工輸入,，而且參數(shù)及其復(fù)雜，但可很方便的供其他程序調(diào)用,，LAME同時(shí)也提供了一個(gè)DLL版本,，但我們認(rèn)為不如EXE版本的好，所以忽略不提,。不要被LAME復(fù)雜的參數(shù)所嚇倒,，文章中我們會(huì)提示如何操作來達(dá)到一勞永逸的效果。我們需要粗略的了解一下LAME的參數(shù),。 LAME其實(shí)真正要用到的參數(shù)就幾個(gè)而已,。 VBR壓縮級(jí)別參數(shù)：[-V] 指定VBR的壓縮品質(zhì)，范圍為0-9（數(shù)字越小品質(zhì)越高）,，預(yù)設(shè)值為4,。 碼率參數(shù)：[-b] 指定流量變動(dòng)的下限，預(yù)設(shè)為32Kbps,。[-B] 指定流量變動(dòng)的上限,，預(yù)設(shè)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(zhì)編碼模式參數(shù)：[-h] 高品質(zhì)編碼模式,。這個(gè)選項(xiàng)在 VBR 壓縮模式中是預(yù)設(shè)開啟的。CBR編碼模式中是關(guān)閉的,。 精度參數(shù)：[-q] 指定頻率資料量化時(shí)的精確度,，范圍是為0-9（數(shù)字越小品質(zhì)越高），預(yù)設(shè)值為2,。如果在使用-q 0參數(shù)是覺得編碼速度慢得過份,，請使用默認(rèn)值。如果編碼的曲子是鋼琴或者小提琴,、古箏二胡這類細(xì)節(jié)很豐富的樂器獨(dú)奏,，我們推薦你就是耐著性子也要用-q 0參數(shù),，雖然慢點(diǎn)，但值得,。 聲道模式參數(shù)：[-m] 立體聲壓縮模式,，細(xì)分參數(shù)分別有 s:Stereo j:Joint Stereo f:Force ms_stereo m:Mono。當(dāng)使用VBR編碼并把品質(zhì)設(shè)為4-9和使用CBR編碼流量小于160 Kbps時(shí),，預(yù)設(shè)為j(Joint Stereo),。其余時(shí)候預(yù)設(shè)為s(Stereo)。 通過長期的使用,，我們給出2個(gè)參數(shù)使用建議,。 CBR 模式編碼的推薦參數(shù)：-b -m s -h （ 為碼率數(shù)值）。VBR 模式編碼推薦參數(shù)：-V 0,。 在新版本的LAME中（3.90后）,，LAME提供了全新的--alt-preset系列預(yù)置參數(shù)，這組參數(shù)最大的好處就是不用再去記憶那些繁多的參數(shù),，而提供最佳化的選擇。 CBR模式： --alt-preset insane 320kbps CBR模式,，音質(zhì)最好,，體積最大。 VBR模式： --alt-preset extreme 平均Bitrate范圍在192～256kbps之間,，音質(zhì)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(shù)提供比老參數(shù)更優(yōu)秀的音質(zhì),，但編碼效率卻低了很多，您需要更強(qiáng)勁的CPU支持才行,，而相對比老參數(shù)提高相對不是很多,，在乎您的取舍了，筆者傾向使用老參數(shù),。 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> (轉(zhuǎn))hifi級(jí)mp3制作和LAME參數(shù)設(shè)置2009-11-10 18:59:35 mp3也能hifi,，hifi級(jí)mp3制作和LAME參數(shù)設(shè)置mp3也能HIF,hifi級(jí)mp3制作 對于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(gè)G一個(gè)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(shù)音質(zhì)沒法聽,。下面,，我們請出的工具就是LAME,。大家要問了，超級(jí)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(huì)再看一眼,。那么,，LAME有什么絕招呢？LAME的兩大神功就是VBR（動(dòng)態(tài)流量編碼）和心理聲學(xué)模型,。LAME可以說是將VBR的能力發(fā)揮到了極致,。它將波形分割成50幀（30幀約1秒）一段，根據(jù)該段落內(nèi)頻率的高低動(dòng)態(tài)設(shè)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(zhì)就得到了很大程度的保護(hù),。此外,，LAME的心理聲學(xué)模型也是最出色的。就這樣,，LAME將MP3的音質(zhì)提高到了一個(gè)嶄新的階段,，可以說LAME做出的MP3真正有著近似CD的音質(zhì)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(gè)問題,。筆者現(xiàn)在使用的就是一個(gè)名為RazorLame的外殼,， 首先我們設(shè)置一下LAME的參數(shù)，點(diǎn)擊LAME options,。 里面有General, VBR, Advance和Expert等設(shè)置,，要了解這些設(shè)置，我們還是需要首先了解一下LAME繁多的參數(shù),。 CBR（固定流量編碼）編碼時(shí)的基本參數(shù)： 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(wǎng)上下載的MP3最常用的是128kbps,，但是這個(gè)流量顯然是不夠的,。如果你想做接近CD水準(zhǔn)的MP3，推薦你用320Kbps的CBR（最高質(zhì)量MP3）,，這類MP3音質(zhì)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(shù)：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(dāng)然數(shù)字越大，體積越大,，音質(zhì)越好,。這一點(diǎn)，體積與音質(zhì)成正比,。在波形靜音的部分,，LAME會(huì)自動(dòng)采用最小的流量。 -h參數(shù)：高品質(zhì)編碼模式,，可以增加音質(zhì),，我們當(dāng)然需要，一定要毫不猶豫用這個(gè)參數(shù),。這個(gè)選項(xiàng)在 VBR 壓縮模式中是預(yù)設(shè)開啟的,。 -q參數(shù)：指定波形數(shù)據(jù)量化時(shí)的精確度，范圍為0-9,，數(shù)字越低質(zhì)量越好,。筆者選擇2，因?yàn)長AME的開發(fā)者推薦這個(gè)參數(shù),。0理論上最好,，但是開發(fā)者說這是個(gè)實(shí)驗(yàn)型參數(shù)（不懂）。 因此,，最強(qiáng)的MP3的命令行：-b 320 –h –q2。 VBR（動(dòng)態(tài)流量編碼）編碼時(shí)的基本參數(shù)： VBR編碼是LAME一大神功,，可為你提供最佳的音質(zhì)/體積比,，所以筆者強(qiáng)烈推薦使用VBR。 -V參數(shù)：指定VBR的壓縮品質(zhì),，范圍為0-9（數(shù)字越小品質(zhì)越高）,，我們選擇2。 -b參數(shù)： 指定流量變動(dòng)的下限,，預(yù)設(shè)為32Kbps,。使用預(yù)設(shè)就可以了。 -B參數(shù)： 指定流量變動(dòng)的上限,，預(yù)設(shè)為320Kbps,。推薦使用預(yù)設(shè)值 其他如-q參數(shù)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(shù)為—abr 然后是一些共同參數(shù)： -m參數(shù)：選擇立體聲輸出方式：有-ms (Stereo 立體聲) -mj (Joint Stereo 聯(lián)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(shù),，開發(fā)者又提供一組強(qiáng)大的預(yù)制參數(shù)-ap供選擇,。這類參數(shù)是以--alt –present開頭，因此,，最好的參數(shù)又有了新的選擇： CBR參數(shù)：--alt-preset insane或者--alt-preset cbr 320,。音質(zhì)最好，體積最大,。 VBR參數(shù)：.--alt-preset extreme,。音質(zhì)很好，體積小,，筆者推薦并使用這一參數(shù),。 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(shí)驗(yàn)性參數(shù),，有興趣可以試試，說不定可以試出什么新的最優(yōu)化參數(shù)來,，其中有一個(gè) Delete source file after encoding 的選項(xiàng),，選取之后，編碼完成后原始的波形文件會(huì)被自動(dòng)刪除,，非常方便,。然后是核心——VBR的設(shè)置。這里你可以通過上面學(xué)到的知識(shí)進(jìn)行設(shè)置,，不錯(cuò)吧,。再后就是Expert——專家設(shè)置。這里面有一個(gè)Custom options,?？梢宰约褐苯訉懨钚校沁@一項(xiàng)好像不是給專家設(shè)計(jì)的——更像給懶人使用的,，你只要把筆者的推薦CBR或VBR參數(shù)拷貝上去,，然后在底下only use custom options的選項(xiàng)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(rèn)為32KHz,，會(huì)引起音質(zhì)的下降,。最后，點(diǎn)擊編碼（Encode）就可以開始了,。再耐心等待幾分鐘,，我們的HIFI級(jí)MP3就出爐了。 ================================================= LAME問與答——兼談最新的編碼參數(shù)設(shè)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(xì)節(jié)表現(xiàn)良好,，它獨(dú)創(chuàng)的心理音響模型技術(shù)保證了CD音頻還原的真實(shí)性，配合VBR和ABR參數(shù),，音質(zhì)幾乎可以媲美CD音頻,，但文件體積卻非常小。對于一個(gè)免費(fèi)引擎,，LAME的優(yōu)勢不言而喻,。 2.上邊提到的VBR和ABR是什么？還有CBR,？ VBR（Variable Bitrate）動(dòng)態(tài)比特率,。也就是沒有固定的比特率，壓縮軟件在壓縮時(shí)根據(jù)音頻數(shù)據(jù)即時(shí)確定使用什么比特率,，這是以質(zhì)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(shù),。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(diǎn)獨(dú)創(chuàng)了這種編碼模式。ABR在指定的文件大小內(nèi),，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(dòng)態(tài)表現(xiàn)時(shí)使用高流量,，可以做為VBR和CBR的一種折衷選擇,。 CBR（Constant Bitrate），常數(shù)比特率,，指文件從頭到尾都是一種位速率。相對于VBR和ABR來講,，它壓縮出來的文件體積很大,，而且音質(zhì)相對于VBR和ABR不會(huì)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(qū)別,？哪一種比較好？ LAME分DLL和EXE兩種版本,，DLL版本做為一個(gè)方便的接口程序在大多數(shù)抓軌軟件中都能看到（比如AltoMP3Maker）,，但由于可控性差，與具備豐富調(diào)節(jié)參數(shù)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(yùn)行的程序,？太難用了 針對這一點(diǎn)，網(wǎng)上出現(xiàn)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(diào)整各種參數(shù),，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(diào)用LAME.exe,，可以在抓軌同時(shí)壓縮MP3,，事半功倍。 5.我在一些網(wǎng)站學(xué)會(huì)了使用-V 0 -q 0這樣的終極參數(shù),，這下可以壓出最高品質(zhì)MP3了 實(shí)際上象-V 0 -q 0這樣的參數(shù)可以壓縮出最高品質(zhì)MP3的說法從來都不是LAME開發(fā)者所應(yīng)允的,。在LAME中，象0,、1這樣的Level屬于試驗(yàn)參數(shù),，如果用它壓縮MP3，非但不會(huì)提高音質(zhì)（相對于Level2而言）,，反而會(huì)導(dǎo)入多余的噪音,，所以以上的參數(shù)應(yīng)該改為-V 2 -q 2。實(shí)際上象這樣的參數(shù)標(biāo)準(zhǔn)幾近淘汰,，-ap參數(shù)將做為新的LAME參數(shù)標(biāo)準(zhǔn)逐漸流行,。 6.-ap參數(shù)？沒聽說過 這種參數(shù)屬于預(yù)置參數(shù),。 --abr 128 -h --nspsytune --athtype 2 --lowpass 16 --ns-bass -8 --scale 0.93,，面對上邊這組微調(diào)參數(shù)你會(huì)不會(huì)有一種暈菜的感覺呢@_@……正是為了簡化參數(shù)設(shè)置，避免各種不必要的試驗(yàn)參數(shù),，LAME開發(fā)者精心調(diào)配出了-ap參數(shù),，它是一組代碼級(jí)參數(shù)（也就是說沒有微調(diào)參數(shù)可以實(shí)現(xiàn)與它相同的功能）。使用這種新的預(yù)置參數(shù)標(biāo)準(zhǔn)既可以壓縮出更高品質(zhì)的MP3,，又可以避免我們陷入微調(diào)參數(shù)的迷宮中,。以下是-ap參數(shù)列表： 最高品質(zhì)參數(shù)： --alt-preset insane或者--alt-preset cbr 320 320k CBR，音質(zhì)最好,，文件體積最大 VBR參數(shù)： 1.--alt-preset extreme 220-270k左右的VBR,，音質(zhì)與上面參數(shù)相仿，但文件體積小25%,，推薦此參數(shù) 2.--alt-preset fast extreme 音質(zhì)比上面參數(shù)稍微差一些 3.--alt-preset standard 180-220k左右的VBR,，在音質(zhì)和文件大小之間比較好的平衡 4.--alt-preset fast standard 音質(zhì)比上面參數(shù)稍微差一些 5.--alt-preset standard -Y 雖然品質(zhì)稍差，但文件體積非常小 ABR參數(shù)： --alt-preset （可用Bitrate：80,、96,、112、128,、160,、192、224,、256,、320） CBR參數(shù): --alt-preset cbr （可用Bitrate：80,、96,、112、128,、160,、192、224,、256,、320） ======================================================== 對MP3及音頻壓縮技術(shù)的一些誤解 1、mp3的音質(zhì)很差,？ 錯(cuò),。mp3作為當(dāng)前音頻有損壓縮的“王者”，它的編碼技術(shù)已經(jīng)幾近完美,。很多人只是不清楚如何才能壓縮出高品質(zhì)的mp3而已,。2001年12月，世界上最優(yōu)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(shù)過于煩瑣的情況,，提供了幾個(gè)preset（預(yù)設(shè)）參數(shù)。現(xiàn)在只要使用LAME的standard（標(biāo)準(zhǔn)）模式進(jìn)行壓縮,，就能得到近似于CD的完美音質(zhì),。 2、128kbps的mp3=CD音質(zhì),？ 錯(cuò),。首先，所謂CD音質(zhì)是一個(gè)帶有很大主觀性的名詞,，基本上可以認(rèn)為CD音質(zhì)意味著在平均水平的聽音條件下能達(dá)到用光驅(qū)放CD的效果。但是根據(jù)這個(gè)定義,，無數(shù)的試聽結(jié)果表明,，不管用什么編碼器，什么樣的設(shè)置，128kbps的mp3都不能達(dá)到這個(gè)標(biāo)準(zhǔn),。關(guān)于這方面的主題可參http:///,這是一個(gè)非常著名的國外音頻站點(diǎn),，對128kbps的mp3的測試有非常詳細(xì)的理論闡述。 3,、mp3 192kbps CBR(固定比特速率) stereo(立體聲)編碼是音質(zhì)與文件大小的最佳平衡設(shè)置,？ 錯(cuò)。這一誤解有很深的根源,。因?yàn)?28kbps的mp3在音質(zhì)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(shè)置。對Xing編碼器及Fraunhofer編碼器來說,，直到現(xiàn)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(rèn)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(yōu)秀，已經(jīng)沒什么理由再去使用CBR和stereo--這樣做只會(huì)浪費(fèi)有限的bits,。標(biāo)準(zhǔn)的VBR預(yù)定設(shè)置（即使用--alt-preset standard參數(shù)）生成的mp3文件的平均比特率也是192kbps,，但它的音質(zhì)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(zhì)在該bitrate左右好于mp3, 2,、最近的oggenc 1.0--not tested yet）。 4,、mp3 320kbps CBR Stereo是mp3音質(zhì)的極限,？ 錯(cuò)（或者說Not exactly true）。雖然320kbps是mp3標(biāo)準(zhǔn)的極限,，但在320kbps下使用設(shè)計(jì)良好的Joint Stereo,，能夠?qū)⒐?jié)省下下的bits用于純粹的音樂部分（從而提高音質(zhì)）。如果音源的立體聲分離度很低,，使用完全的stereo是一種浪費(fèi),。 5、VBR的音質(zhì)不如CBR,？ 錯(cuò),。設(shè)計(jì)良好的VBR算法不會(huì)將bits浪費(fèi)在易于編碼的部分，節(jié)省下來的bits將用在對復(fù)雜的音頻部分進(jìn)行編碼,。這一誤解可能來自于較老的FhG Encoder的VBR算法及Xing VBR算法中存在的bug,對當(dāng)前的lame編碼器來說,它的VBR算法已被協(xié)調(diào)得很好,不會(huì)有音質(zhì)上的問題,。 6、Joint Stereo 音質(zhì)不佳,？ 錯(cuò),。當(dāng)前主流的encoder如lame,、mppenc、oggenc,、aacenc都使用了所謂smart joint stereo的技術(shù),，不會(huì)破壞stereo image,請參閱如下的兩個(gè)鏈接（E文,由編碼器的開發(fā)者解答）： http://www./forums/showthread.php?s=&threadid=1081 ; http://www./forums/showthread.php?s=&threadid=759 ; 更為技術(shù)性的解釋如下： http://www./ogg/vorbis/doc/stereo.html ; 7、Blade是最佳的mp3編碼器,？ 錯(cuò),。（似乎不用過多的解釋）Blade不推薦用于所有bitrate的mp3編碼，由于缺少相當(dāng)多的功能,，其音質(zhì)較lame或FhG遜色很多,。下面的兩個(gè)鏈接有助于了解blade的缺憾： http://forums./thread_view.cfm/1914 ; http://www./forums/showthread.php?s=&threadid=463 ; 最新消息——Blade已經(jīng)停止開發(fā)，其作者在主頁上聲明ogg是更好的選擇 8,、wma在64kbps可達(dá)CD音質(zhì),？ 錯(cuò)。不用我多費(fèi)筆墨,，不相信的話點(diǎn)擊下面的鏈接了解詳情:： http://www./forums/showthread.php?s=&threadid=1434 ; http://forums./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(shù)? 錯(cuò),。編碼器是在音頻信號(hào)級(jí)進(jìn)行處理,，不去分辨音樂類型。只要心理學(xué)模型與編碼算法正確,，同一設(shè)置就適用于所有的音樂類型,。詳情參見： http://www./forums/showthread.php?s=&threadid=1835 ====================================================== 小身材也要大味道——128kbps下如何設(shè)置Lame編碼參數(shù) Lame MP3編碼引擎大家已經(jīng)相當(dāng)熟悉了，而且在APX參數(shù)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(diǎn),，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(shí)一般來說，128kbps的編碼率下,，任何編碼器都無法達(dá)到CD音質(zhì)（M$所言,，WMA在64kbps或96kpbs就能達(dá)到CD Quality是一個(gè)真實(shí)的謊言），對Lame來說,，要想在128kbps超過那些專門為低bitrate作了優(yōu)化的encoder如mp3pro,、wma甚至ogg，冗長的參數(shù)是不可或缺的,，這篇短文就為您進(jìn)行詳細(xì)的解釋 1,、Lame的版本的問題 Lame.exe的當(dāng)前的最新穩(wěn)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(gè)版本就是dibrom（Lame preset參數(shù)的開發(fā)者）編譯的3.90.2,，Lame隨后的3.91、3.92版本有相當(dāng)部分（特別是preset部分）是脫胎于此版的,。這也是當(dāng)前在preset參數(shù)設(shè)置下編碼最快的版本，下載鏈接如下http://www./extra/Lame/Lame3.90.2-ICL.zip ; Lame的開發(fā)速度很快,，3.93的alpha版已經(jīng)出過十幾個(gè)了,。雖然內(nèi)部測試版不推薦使用，但它的確修正了不少的錯(cuò)誤（像對人們誤解最大的q0參數(shù)的修正）,，所以也提供一個(gè)下載鏈接,，有興趣的朋友不妨一試:http://mitiok./Lame-20020706.zip（這是最新7月6日版）。 2,、參數(shù)設(shè)置 Lame的參數(shù)設(shè)置的爭論是最大的,，我也有被千夫所指的經(jīng)歷和準(zhǔn)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(dāng)) --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(zhì)來說,，我認(rèn)為，ABR>CBR,。 小結(jié)： 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(shù)之多很為人詬病,，preset的出現(xiàn)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(rèn)這是internet上最流行的……,。好像主題已經(jīng)有點(diǎn)亂了，就此打住. 獨(dú)樂樂不如眾樂樂,，讓我們一起研究,、共享我們的知識(shí)，我們的音樂,。 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> ?。ㄞD(zhuǎn)）lame 3.90.3 轉(zhuǎn)換mp3的所有參數(shù)2009-11-11 12:28:13| 分類： 默認(rèn)分類 閱讀315 評(píng)論0 字號(hào)：大中小訂閱 LAME version 3.90.3 MMX (http://www./) -- Compiled at http://www. -- Check this website for up to date information on the --alt-presets usage: lame [options] [outfile] and/or 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 gapless encoding for a set of contiguous files --nogapout output dir for gapless encoding (must precede --nogap) Operational options: -m (s)tereo【立體聲】, (j)oint【聯(lián)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 choose bitrate to achive a compression ratio of --scale scale input (multiply PCM data) by --scale-l scale channel 0 (left) input (multiply PCM data) by --scale-r scale channel 1 (right) input (multiply PCM data) by --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 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 = 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 set the bitrate in kbps, default 128 kbps ABR options: --abr 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 specify minimum allowed bitrate, default 32 kbps -B 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 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 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 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 frequency(kHz), lowpass filter cutoff above freq --lowpass-width frequency(kHz) - default 15% of lowpass freq --highpass frequency(kHz), highpass filter cutoff below freq --highpass-width frequency(kHz) - default 15% of highpass freq --resample sampling frequency of output file(kHz)- default=automatic ID3 tag options: --tt audio/song title (max 30 chars for version 1 tag) --ta audio/song artist (max 30 chars for version 1 tag) --tl audio/song album (max 30 chars for version 1 tag) --ty audio/song year of issue (1 to 9999) --tc user-defined text (max 30 chars for v1 tag, 28 for v1.1) --tn audio/song track number (1 to 255, creates v1.1 tag) --tg 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(gè)人在foobar0.83中的lame里使用的參數(shù)為： -m s -q 0 -b 320 --noath -k - %d LAME其實(shí)真正要用到的參數(shù)就幾個(gè)而已,。 VBR壓縮級(jí)別參數(shù)：[-V] 指定VBR的壓縮品質(zhì),，范圍為0-9（數(shù)字越小品質(zhì)越高），預(yù)設(shè)值為4,。 碼率參數(shù)：[-b] 指定流量變動(dòng)的下限,，預(yù)設(shè)為32Kbps。[-B] 指定流量變動(dòng)的上限,，預(yù)設(shè)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(zhì)編碼模式參數(shù)：[-h] 高品質(zhì)編碼模式。這個(gè)選項(xiàng)在 VBR 壓縮模式中是預(yù)設(shè)開啟的,。CBR編碼模式中是關(guān)閉的,。 精度參數(shù)：[-q] 指定頻率資料量化時(shí)的精確度，范圍是為0-9（數(shù)字越小品質(zhì)越高）,，預(yù)設(shè)值為2,。如果在使用-q 0參數(shù)是覺得編碼速度慢得過份，請使用默認(rèn)值,。如果編碼的曲子是鋼琴或者小提琴,、古箏二胡這類細(xì)節(jié)很豐富的樂器獨(dú)奏，我們推薦你就是耐著性子也要用-q 0參數(shù),，雖然慢點(diǎn),，但值得。 聲道模式參數(shù)：[-m] 立體聲壓縮模式,，細(xì)分參數(shù)分別有 s:Stereo j:Joint Stereo f:Force ms_stereo m:Mono,。當(dāng)使用VBR編碼并把品質(zhì)設(shè)為4-9和使用CBR編碼流量小于160 Kbps時(shí)，預(yù)設(shè)為j(Joint Stereo),。其余時(shí)候預(yù)設(shè)為s(Stereo),。 通過長期的使用，我們給出2個(gè)參數(shù)使用建議,。 CBR 模式編碼的推薦參數(shù)：-b -m s -h （ 為碼率數(shù)值）,。VBR 模式編碼推薦參數(shù)：-V 0。 在新版本的LAME中（3.90后）,，LAME提供了全新的--alt-preset系列預(yù)置參數(shù),，這組參數(shù)最大的好處就是不用再去記憶那些繁多的參數(shù)，而提供最佳化的選擇,。 CBR模式： --alt-preset insane 320kbps CBR模式,，音質(zhì)最好，體積最大,。 VBR模式： --alt-preset extreme 平均Bitrate范圍在192～256kbps之間,，音質(zhì)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(shù)提供比老參數(shù)更優(yōu)秀的音質(zhì)，但編碼效率卻低了很多,，您需要更強(qiáng)勁的CPU支持才行,，而相對比老參數(shù)提高相對不是很多,，在乎您的取舍了，筆者傾向使用老參數(shù),。 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> lame3.90.3 Full command line switch reference2009-11-11 13:06:58| 分類： 默認(rèn)分類 閱讀17 評(píng)論0 字號(hào)：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. switchparameter -adownmix stereo file to mono --abraverage bitrate encoding --allshortuse short blocks only --athlowerlower the ATH --athonlyATH only --athshortATH only for short blocks --athtypeselect ATH type -bbitrate (8...320) -Bmax VBR/ABR bitrate (8...320) --bitwidthinput bit width -ccopyright --compchoose compression ratio --cwlimittonality limit -dblock type control --decodedecoding only --disptimetime between display updates -ede-emphasis (n, 5, c) -ffast mode -Fstrictly enforce the -b option --freeformatfree format bitstream -hhigh quality --helphelp --highpasshighpass filtering frequency in kHz --highpass-widthwidth of highpass filtering in kHz -kfull bandwidth --lowpasslowpass filtering frequency in kHz --lowpass-widthwidth of lowpass filtering in kHz -mstereo mode (s, j, f, m) --mp1inputMPEG Layer I input file --mp2inputMPEG Layer II input file --mp3inputMPEG Layer III input file --noathdisable ATH --nohistdisable histogram display --noresdisable bit reservoir --noshortdisable short blocks frames --notempdisable temporal masking -onon-original -perror protection --presetuse built-in preset --alt-presetuse updated and much higher quality 'alternate' presets --priorityOS/2 process priority control -qalgorithm quality selection --quietsilent operation -rinput file is raw pcm --resampleoutput sampling frequency in kHz (encoding only) --r3mixr3mix VBR preset -ssampling frequency in kHz -Ssilent operation --scalescale input --scale-lscale input channel 0 (left) --scale-rscale input channel 1 (right) --shortuse short blocks --silentsilent operation --strictly-enforce-ISOstrict ISO compliance -tdisable INFO/WAV header -VVBR quality setting (0...9) --vbr-newnew VBR mode --vbr-oldolder VBR mode --verboseverbosity -xswapbytes -Xchange 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. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --allshort use short blocks only Use only short blocks, no long ones. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --athshort ATH only for short blocks Ignore psychoacoustic model for short blocks, use ATH only. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * -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. -------------------------------------------------------------------------------- * -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. -------------------------------------------------------------------------------- * -c copyright Mark the encoded file as being copyrighted. -------------------------------------------------------------------------------- * --comp choose compression ratio Instead of choosing bitrate, using this option, user can choose compression ratio to achieve. -------------------------------------------------------------------------------- * --cwlimit n tonality limit Compute tonality up to freq (in kHz). Default setting is 8.8717. -------------------------------------------------------------------------------- * -d block type control Allows the left and right channels to use different block size types. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --disptime n time between display updates Set the delay in seconds between two display updates. -------------------------------------------------------------------------------- * -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. -------------------------------------------------------------------------------- * -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. -------------------------------------------------------------------------------- * -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). -------------------------------------------------------------------------------- * --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 -------------------------------------------------------------------------------- * -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. -------------------------------------------------------------------------------- * --help help Display a list of all available options. -------------------------------------------------------------------------------- * --highpass highpass filtering frequency in kHz Set an highpass filtering frequency. Frequencies below the specified one will be cutoff. -------------------------------------------------------------------------------- * --highpass-width width of highpass filtering in kHz Set the width of the highpass filter. The default value is 15% of the highpass frequency. -------------------------------------------------------------------------------- * -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! -------------------------------------------------------------------------------- * --lowpass lowpass filtering frequency in kHz Set a lowpass filtering frequency. Frequencies above the specified one will be cutoff. -------------------------------------------------------------------------------- * --lowpass-width width of lowpass filtering in kHz Set the width of the lowpass filter. The default value is 15% of the lowpass frequency. -------------------------------------------------------------------------------- * -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. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --nores disable bit reservoir Disable the bit reservoir. Each frame will then become independent from previous ones, but the quality will be lower. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --notemp disable temporal masking Don't make use of the temporal masking effect. -------------------------------------------------------------------------------- * -o non-original Mark the encoded file as being a copy. -------------------------------------------------------------------------------- * -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. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * -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. -------------------------------------------------------------------------------- * -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. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --r3mix r3mix VBR preset Uses r3mix VBR preset. See www.r3mix.net for more details. -------------------------------------------------------------------------------- * -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. -------------------------------------------------------------------------------- * -S / --silent / --quiet silent operation Don't print progress report. -------------------------------------------------------------------------------- * --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 Use with care, since most MP3 decoders will truncate data which decodes to values greater than 32768. -------------------------------------------------------------------------------- * --short use short blocks Let LAME use short blocks when appropriate. It is the default setting. -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * -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. -------------------------------------------------------------------------------- * -V 0...9 VBR quality setting Enable VBR (Variable BitRate) and specifies the value of VBR quality. default=4 0=highest quality. -------------------------------------------------------------------------------- * --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). -------------------------------------------------------------------------------- * --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. -------------------------------------------------------------------------------- * --verbose verbosity Print a lot of information on screen. -------------------------------------------------------------------------------- * -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. -------------------------------------------------------------------------------- * -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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