AW: [vorbis-dev] PlusV algorithm -> CBR
Rob D
rddone at att.net
Thu Jan 30 05:17:00 PST 2003
Even if you want a buffer for less time, you may be out of luck.
In windows, the OS/sound card caches 2-3 seconds of music when you send WAV
data to the card.
The sound starts nearly immediately when you send it the first PCM packet,
but the OS/driver accepts several seconds worth of data before it waits to
return.
There may be a way to tell windows to use a smaller (.25 seconds or so)
buffer, and let the player buffer the data in a way to allow nearly
immediate EQ/volume control, but I am unaware.
Rob D
-----Original Message-----
From: owner-vorbis-dev at xiph.org [mailto:owner-vorbis-dev at xiph.org]On
Behalf Of Geoff Shang
Sent: Thursday, January 30, 2003 3:52 AM
To: vorbis-dev at xiph.org
Subject: Re: AW: [vorbis-dev] PlusV algorithm -> CBR
<p>On Wed, 29 Jan 2003, Hannes wrote:
> I know it would mean a great deal of damage to vorbis quality, but would
it
> really be so bad considering the possibilities? And if, how big would the
> bandwidth adaptation window really have to be - 2 seconds? thats about 1.9
> seconds too long, if this adds on my streaming latency, sorry.
Most players have a much bigger buffer than 2 seconds. Even some testing
I've done with MP3 and mpg123 tends to produce latency of at least 1
second. Vorbis is not a low-latency codec, like it or not. I can't
remember the exact figure, but there's a delay that vorbis can not go
below. that's just fact. It's the nature of the beast.
Geoff.
<p>--- >8 ----
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