msmith at labyrinth.net.au
Fri Dec 29 21:23:43 PST 2000
>> I don't know the details of how this is meant to work, though, except that
>> it's mostly codebook trickeruy.
>I believe the method is: just truncating the packets, and losing the
>high-frequency part of the music.
No, it's definately more than just cutting out the high-frequency bits
entirely. It's something not supported by any other widely-used formats, to
my knowledge. It also doesn't allow you to arbitrarily truncate the packet
at any point.
>"Fast bitrate scaling" is not something I've heard anyehere else, so
>I tought I'd like an explanation .
>> Further features specifically for surround sound may exist in the future,
>> too (channel coupling for multiple channels, etc).
>Yet the website doesn't mention _these_ facts.
It's a single paragraph mentioning features the format supports. Of COURSE
it doesn't say everything. That stuff isn't particularly aimed at a
technical audience, either.
>> >-- I don't think Vorbis can do sample-gran seeks (yeah, sure, we can
>> > emulate it, but anyone can, and it's not implemented at this time).
>> Since it's a transform codec, obviously the 'real' seeking is to the
>> nearest block (packet). Emulating it is possible (as it presumably is with
>> other transform codecs, though I don't know if anyone actually does it),
>> and the code does this. See ov_pcm_seek(). There's also a faster function
>> that just does 'rough' seeking.
>> Unlike some codecs (mp3 comes to mind...), vorbis is capable of producing
>> (as output) the same length (in samples) of data as the input - hence "fast
>> sample-granularity [...] decode".
>But that's something different from what you're saying.
It is? Vorbis (the codec to the neccesary extent, the supporting libraries
fully, and the apps - well, the apps would, except the user interfaces
don't lend themselves to doing stuff with sample granularity.
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