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Risultati sondaggio: Che formato utilizzi per la compressione audio? | |||
mp3 | 329 | 71.21% | |
wma | 21 | 4.55% | |
aac | 61 | 13.20% | |
apple lossless | 26 | 5.63% | |
sony atrac | 3 | 0.65% | |
ogg vorbis | 43 | 9.31% | |
flac/ape | 149 | 32.25% | |
wav | 24 | 5.19% | |
asf | 0 | 0% | |
altro formato compesso | 3 | 0.65% | |
altro formato non compresso | 3 | 0.65% | |
Sondaggio a risposta multipla Votanti: 462. Non puoi votare in questo sondaggio |
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21-01-2011, 09:12 | #421 |
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MP3 LAME
LAME development started around mid-1998. Mike Cheng started it as a patch against the 8hz-MP3 encoder sources. After some quality concerns raised by others, he decided to start from scratch based on the dist10 sources. His goal was only to speed up the dist10 sources, and leave its quality untouched. That branch (a patch against the reference sources) became Lame 2.0, and only on Lame 3.81 did we replaced of all dist10 code, making LAME no more only a patch. The project quickly became a team project. Mike Cheng eventually left leadership and started working on tooLame, an MP2 encoder. Mark Taylor became leader and started pursuing increased quality in addition to better speed. He can be considered the initiator of the LAME project in its current form. He released version 3.0 featuring gpsycho, a new psychoacoustic model he developed. In early 2003 Mark left project leadership, and since then the project has been lead through the cooperation of the active developers (currently 4 individuals). Today, LAME is considered the best MP3 encoder at mid-high bitrates and at VBR, mostly thanks to the dedicated work of its developers and the open source licensing model that allowed the project to tap into engineering resources from all around the world. Both quality and speed improvements are still happening, probably making LAME the only MP3 encoder still being actively developed. Ultima modifica di Khanattila : 21-01-2011 alle 09:28. |
21-01-2011, 09:17 | #422 |
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In ambito Open Source, progetto Xipth.org
-FLAC -VORBIS -CELT -SPEEX FLAC FLAC stands for Free Lossless Audio Codec, an audio format similar to MP3, but lossless, meaning that audio is compressed in FLAC without any loss in quality. This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player (or your car or home stereo, see supported devices) just like you would an MP3 file. FLAC stands out as the fastest and most widely supported lossless audio codec, and the only one that at once is non-proprietary, is unencumbered by patents, has an open-source reference implementation, has a well documented format and API, and has several other independent implementations. * The Compression ratio is calculated with the division of compressed size by uncompressed size * 100. So, lower is better. Encoding speed, Decoding speed and Compression ratio are based on each encoder's default settings. ** Flexibility refers to the amount of encoding choices offered to the users (Fast/low compression, Slow/high compression and everything inbetween) VORBIS Ogg Vorbis is a new audio compression format. It is roughly comparable to other formats used to store and play digital music, such as MP3, VQF, AAC, and other digital audio formats. It is different from these other formats because it is completely free, open, and unpatented. CELT The CELT codec is a compression algorithm for audio. Like MP3, Vorbis, and AAC it is suitable for transmitting music with high quality. Unlike these formats CELT imposes very little delay on the signal, even less than is typical for speech centric formats like Speex, GSM, or G.729. Using CELT application developers can build software that allows musicians to perform together across the Internet, or simply build great sounding telephony systems. Why shouldn't your telephone sound as good as your stereo? Like other open and unencumbered technology from the Xiph.Org Foundation CELT requires no royalties and no complicated licensing. 48 Kbit/s 64 Kbit/s SPEEX Speex is an Open Source/Free Software patent-free audio compression format designed for speech. The Speex Project aims to lower the barrier of entry for voice applications by providing a free alternative to expensive proprietary speech codecs. Moreover, Speex is well-adapted to Internet applications and provides useful features that are not present in most other codecs. Finally, Speex is part of the GNU Project and is available under the revised BSD license. Ultima modifica di Khanattila : 21-01-2011 alle 09:37. |
21-01-2011, 09:51 | #423 |
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Sempre in ambito Open Source, ma fuori dal progetto Xiph.org
MUSEPACK Musepack is an audio compression format with a strong emphasis on high quality. It's not lossless, but it is designed for transparency, so that you won't be able to hear differences between the original wave file and the much smaller MPC file. It is based on the MPEG-1 Layer-2 / MP2 algorithms, but since 1997 it has rapidly developed and vastly improved and is now at an advanced stage in which it contains heavily optimized and patentless code. |
22-01-2011, 22:36 | #424 |
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Finiamo la carrellata con uno dei progetti più ambiziosi in circolazione, ossia il lossy wav.
LossyWAV is based on the lossyFLAC idea proposed by David Robinson at Hydrogenaudio, which is a method of carefully reducing the bitdepth of samples, therefore utilising the wasted bits feature of the FLAC lossless codec. The aim is to transparently reduce audio bit depth (by making some lower significant bits (lsb's) zero), consequently taking advantage of FLAC's detection of consistently-zeroed lower significant bits within each single frame and significantly increasing coding efficiency.[1] In this way the user can enjoy audio encoded using the same codec (which may be all important from a hardware compatibility perspective) at a reduced bitrate compared to the lossless version. Nick Currie ported the original MATLAB implementation to Delphi (Many thanks CodeGear for Turbo Explorer!!) with a liberal sprinkling of IA-32 and x87 Assembly Language for speed. Subsequently, lossyFLAC proved itself to work with other lossless codecs, so the application name was changed to lossyWAV. Since then, Nick has heavily developed and built upon lossyWAV, with valuable tuning performed by Horst Albrecht at Hydrogenaudio. Although the current lossyWAV implementation has built on David's original method, the method itself still very much belongs to its author. Indicative bitrate reduction It must be stressed that lossyWAV is a pure variable bit-depth pre-processor in that the overall sample size remains the same after processing but the number of significant bits used for the samples in a codec-block can change on a block-by-block basis. Bits-to-remove from the audio data are calculated on a block-by-block basis (codec-block length = 512 samples, 11.6msec @ 44.1kHz) using overlapping fast Fourier Transform (FFT) analyses of at least two lengths (default quality preset (-q 5) = 32, 64 & 1024 samples). After some manipulation, the results of each FFT analysis for a specific codec-block are then grouped and the minimum value used to determine bits-to-remove for the whole codec-block. Bit removal adds white noise to the output, however the level of the added noise associated with the removal of a number of bits has been pre-calculated and the number of bits to remove will depend on the level of the noise floor of the codec-block in question. Each sample in the codec-block is then rounded such that the first <bits-to-remove> lsb's are zero. In this way the wasted bits feature of FLAC et al. is exploited. P:S. SI potrebbe fare una cernita dei miei post precedenti e vederne di farne una guida Ultima modifica di Khanattila : 22-01-2011 alle 22:40. |
22-01-2011, 22:48 | #425 |
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@ Khanattila:
i tuoi post sono interessantissimi e molto chiari. Decidi se permettermi di mettere tutto nel primo post, specificando chiaramente che il merito di questa miniguida è tutto tuo, oppure se preferisci aprire un post ad hoc. Io sono molto interessata e ho letto con attenzione, ammetto che del lossy wav ignoravo tutto.
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22-01-2011, 23:24 | #426 | |
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Quote:
P.s. Penso che pure il CELT sia un formato poco conosciuto |
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23-01-2011, 02:05 | #427 |
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Mai sentito, sto leggendo ora su wiki.
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23-01-2011, 11:32 | #428 |
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23-01-2011, 15:08 | #429 |
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Come vuoi tu, puoi anche chiedere a Kewell se ti dà direttamente il primo post, non ho ancora capito come funziona ma magari si può, così che la discussione possa essere gestita completamente da te.
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25-01-2011, 08:26 | #430 |
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Ho letto molti messaggi di questo tipo, inclusi un paio con "gli sviluppatori sostengono che ma secondo me no". Ora, tu effettui test ABX per decidere oppure vai a orecchio, giusto una curiosità?
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25-01-2011, 10:20 | #431 |
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Penso la prima visto che mp3 sorpra i 192kpbs scala in maniera orribile
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25-01-2011, 14:31 | #432 |
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25-01-2011, 15:10 | #433 |
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Ovvio che io vada a orecchio, un mp3 256 vs lossless perde di netto. Per questo non lo uso.
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25-01-2011, 15:17 | #434 | |
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Quote:
Per me queste affermazioni confinano con la disinformazione. Senza offesa, le preferenze personali quando identificate chiaramente come tali sono sempre e comunque sacrosante. Altro è affermare cose che vengono lette da altri e tendenzialmente rischiano di essere decontestualizzate. Meno male che c'è Hydrogenaudio. Quindi, tu sostieni di essere in grado di effettuare un ABX al 100% di un LAME V0? Su musica normale e non su campioni selezionati? |
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25-01-2011, 15:23 | #435 | |
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Quote:
Ultima modifica di gorman : 25-01-2011 alle 15:33. |
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25-01-2011, 15:35 | #436 | |
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E sì, sono in grado di percepire differenze tra un 256 e un flac rippato da cd e credo che tutti possano riconoscere la differenza quando dotati di opportuni strumenti e indirizzati sui tagli da considerare. Per il 320 CBR il discorso invece è diverso. Per il resto le inesattezze che leggo su hydrogen sono palesi quando leggo accostare il q5-q6 vorbis al flac, ho letto cose su quel forum che potrebbero essere vere solo quando i test siano condotti con auricolari di bassa lega o addirittura stock. Oppure quando i test ABX sono fatti con soggetti poco preparati o non dotati della necessaria attenzione. In questo caso alzo le mani, ma qui in sezione tutti sanno come conduco i miei test e la cura che ci metto. Non sono un utente distratto o poco attento insomma, tutt'altro. Sono malata e fissata Per il resto la differenza di spazio tra un borbis 350 e un mp3 256 non è elevata, ma la differenza anche oggettiva è innegabile anche se i grafici non dicono mai nulla sull'effettiva qualità audio. Nessuno dei due codec riproduce il grafico della canzone non compressa, entrambi mostrano tagli, forse qui entra anche il gusto personale. Per me il vorbis suona decisamente meglio Mp3 lame 256: Vorbis 350: File originale: Da: http://www.head-fi.org/forum/thread/...ate-on-p-7/210
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25-01-2011, 15:39 | #437 | |
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25-01-2011, 15:40 | #438 | |||
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In passaggi specifici, quindi? Che contengono artefatti?
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Ciao! PS Io gli ABX li ho fatti. Sia con Ultimate Ears UE-10 Pro, sia con Beyerdynamic DT-880. Amplificate con un Porta Corda HA-1 (che non è al top ma non certo il peggio). I risultati... sconcertanti |
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25-01-2011, 15:42 | #439 |
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Ecco qua:
http://www.hwupgrade.it/forum/showpo...&postcount=421 Devo riorganizzare tutto il mattone postato post indietro |
25-01-2011, 15:47 | #440 | |
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Quote:
Io non percepisco tutte queste differenze. Io utilizzo mp3 e ne sono soddisfatto. |
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Tutti gli orari sono GMT +1. Ora sono le: 18:49.