Audio Format Transcoding for Compatibility?
brandorf asks: "With the multitude of compressed audio formats that are available today, (MP3, Ogg, AAC, and FLAC to name very few) our music libraries start to spread across quite a few different formats. While this isn't a problem for desktop/media PC use, as programs like Winamp or iTunes have plugins available for almost every format. However, when it's time to start using a portable unit, it's unavoidable that some files will get transcoded. Have there been any studies or experiments as to how similar the codecs really are? Will transcoding from Format A to B sound worse than going from A to C? What's your experience with this?"
http://www.hydrogenaudio.org/forums/index.php?show topic=32440
The site insists on proper ABX tests too, not some thirteen olds insisting they can tell the difference between FLAC and Monkey's Audio codecs.
Use a lossless format for archival purposes (any format really since you won't use it on your portable), then use MP3 for everything else. MP3 is the only thing that pretty much every portable can play. OGG and Windows Media are a close second but I would never consider Windows Media format.
I've noticed significant reduction in lows and an unsettling amount of distortion when I go from vinyl to wax cylinder.
"There are no facts, only interpretations." --Friedrich Nietzsche.
If you have a lossless source, the quality of the derived audio should be as good as an original rip of the CD itself. That's the whole point of lossless encoding. If you're going from lossy->lossy, then any transcoding will introduce garbage, but how much garbage depends on how good the original source is. .anacron
"Hasn't this been asked a thousand times already"
No, RTFA. I'll give you a little clue: read the bit about A to B and A to C.
"Derp de derp."
Seriously. Encode your music in a lossless format, then transcode it to whatever lossy format you use on the go. Sure, it's much bigger, but it will be bit-for-bit accurate! Even if you can't tell the difference on your $50 computer speakers and bundled iPod headphones, you can feel good because you know it's better.
Also, it will get you laid.
Love, your hard drive manufacturer.
encode from mp3 to flac ... ;)
Electronic Music Made Using Linux http://soundcloud.com/polyp
Hard drive space is plentiful. Just rip everything to a lossless format, such as FLAC, or even .wav or .aiff if you can't be bothered with the hassle, then make a convenient shell script to convert everything into another format as and when required. That way, you get the best sounding MP3s or Ogg Vorbis files with none of the bad side effects of transcoding, and as soon as any given codec is improved or replaced by a better one, you won't have to worry about not taking advantage of the shiny new algorithms.
Essentially the same on a $20 boom box maybe. Listen to it on a decent hi-fi. On my sub-$1000 rig (Yamaha HTR-5150 reciever, a pair Boston Acoustic CR-9's), lossy sounds noticeably worse than non-lossy. With my Linux box as the audio source, and an S/PDIF-out (so you can't claim it's my crappy sound card's fault for a shitty signal), if I'm actually listening (as opposed to having music just "in the background" for a conversation) I can certainly hear the difference between 160kbps Vorbis and FLAC. Hell, I can often tell what compression method is used (between Vorbis and MP3) by the types of artifacts. MP3's, for instance, have a huge drop in harmonic complexity (VERY noticeable in violins and cello!) and a limp, flat, soundfield. Vorbis, though not as terrible as MP3, is a hair... soft and muddy.
Both work rather well for portable units with cheap headphones. But for my home system, it's FLAC and CD-Audio.
Lex orandi, lex credendi.
Requantizing audio of a given format to reduce its bitrate is likely to cause less of a problem than switching formats.
Simply put, each format has different criteria on what information is thrown away and what is not. Thus, for example, something that MP3 may keep but AAC throws away will not be present if you transcode from AAC to MP3, IN ADDITION to losing anything that AAC keeps but MP3 throws away. The same holds true in reverse.
retrorocket.o not found, launch anyway?
Different lossy codecs throw away different parts of the audio stream. Trasncoding from one lossy format to another is essentially throwing away the superset of those parts. If you encode at a high enough bitrate, a "lossy" codec throws away almost nothing. If you think that there is no discernable difference, then how do you explain the fact that neither of the lossy formats throw away the entire superset to begin with for superior compression? Codecs have something called a "bitrate". The higher the bitrate, the less they throw away. If you set the bitrate high enough, they become extremely conservative at what they throw away.
On my sub-$1000 rig (Yamaha HTR-5150 reciever, a pair Boston Acoustic CR-9's), lossy sounds noticeably worse than non-lossy.
That statement makes no sense whatsoever. Just digitally adjusting the volume down a little on your recording is "lossy" coding, since you can't recover the original signal from the reduced volume signal. Does that sound worse? I don't think so.
The point is: if you set the bitrate for a lossy codec high enough, you won't hear a difference. If you set it even higher, you won't hear a difference even if you re-encode in another lossy codec once. Etc. As a rule of thumb, the point at which that happens still is before you hit the bitrate of lossless audio codecs.
If you hear a difference, either you are using a bad codec, or you are setting the bitrate too low, or there is something else wrong with your setup.
I only speak for myself, but I know that I consider it silly encoding anything above about 150kbps ABR Ogg Vorbis, because above that, I can't hear the difference. I've been used to listening to 128kbps CBR MP3s for so long that I can't tell the difference any more. Sure, they're crappy PC speakers, but why bother with anything better if I simply cannot tell the difference. A musical friend of mine tried to test this - he played me a song twice, at different bitrates - I couldn't hear any difference. Surely you're listening to the wrong sort of music if you get distracted by the bitrate.. I mean, if it's *really* low then there's obvious reason, even I can tell at the extreme low end of the bitrate spectrum. I'd rather have manageable filesize and reasonable quality. However, I demand more digital music players to be ogg vorbis compatible! What I wouldn't give for one of those sleek HP jukebox jobbies.. *drools*