Metrics on how you upsample an mp3

Sep 08, 2008 16:59

A little thought on mp3s, upsampling, ubuntu (gutsy) and ffmpeg.

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techy, unix, ubuntu, linux, nerdy

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ebel September 9 2008, 09:17:54 UTC
ah the black art of ffmpeg/mencoder/lame/gstreamer arcana.

One thing I liked about Rhythmbox, the Gnome Music Management player is that using HAL it was able to recognise what formats certain media players were able to play and would automagically convert files to that format if you tried to copy files on to them. e.g. when adding Oggs to my iPod, it automagically converted them to MP3s. Perhaps rhythmbox / HAL knows about your media player and can auto convert them? If not file a bug. :P

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tearsofzorro September 9 2008, 10:03:04 UTC
Rhythmbox doesn't seem to know how to talk to my Zen Micro. It's old enough that it uses an obscure protocol - see libnjb for details. The thing is, if you try and put on an mp3 with a weird sampling rate that your iPod can't play properly, will the software pick up on that or does it just make sure that you don't put oggs onto something that can't play them?

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ebel September 9 2008, 12:20:18 UTC
will the software pick up on that or does it just make sure that you don't put oggs onto something that can't play them?

I don't know. But I assume that it can look at sample rates and not just media format.

The advantage of having a media player that everyone else has is that all MP3s you download work for it. :)

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