Re: Audio delay under 10 seconds?
- From: "Neil Smith [MVP Digital Media]" <neil@xxxxxxxxxx>
- Date: Tue, 28 Feb 2006 11:11:56 GMT
On Mon, 27 Feb 2006 14:48:47 -0800, "Carlos M. Diaz-Perez"
<CarlosMDiazPerez@xxxxxxxxxxxxxxxxxxxxxxxxx> wrote:
Thanks Mike,
I had already read that article, but it doesn't answer my question...or
maybe it does and the answer is no, you can't broadcast audio with less than
10 seconds delay using Microsoft Windows Media Encoder...but WHY???
Yes, you can. I don't see anywhere you've posted your settings to
compare against the correct settings for lowest delay. The minimum, as
I stated yesterday, is about 3 seconds.
It seems to be a rather arbitrary amount of time the encoder chooses to
buffer content before it starts sending it out. This is a setting that should
be under the content producer's control.
*It Is* !!!
Many of us would readily trade-off the occasional hiccups during playback
That's a hard one to call - I find rebuffering far more irritating as
a listener than a 2 second live delay (which in any case I'm nor aware
of). Rebufferng leaves you with the feeling you've "missed a bit"
if this translated into near-real-time
streaming. Think of the case where a live event is being streamed, and in
particular a racing event. A 10 second delay on the encoder alone plus the 5
or 10s delay propagating through the network kills the experience for a user
that might be watching the same event on TV but listening to streaming audio
on the computer.
This is a very stretched example I'm afraid. Really - how many users
do this and *why* would you want to do it ??? It's just overkill.
While I understand where you're going on this, practically I'd have to
agree with the "stock" (sic) answer :
"WM codecs in general are optimised for quality over latency"
What this means in practice is they need a certain amount of latency
to be able to achive quite good encoding quality. As I said yeasterday
there *are* lower latency audio codecs, why not try them and see what
happens ?
If you *need* low latency you have to be looking at videoconferencing
type codecs, which WME doesn't support anyway.
But video conference codecs sacrifice quality and bitrate for playback
performance, which can be poor (think pronounced blocking artefacts in
Netmeeting video with any strong movement of the subject)
Try this during the next Nascar race at www.nascar.com and you'll see what I
mean....
But I get that equally with my Cable TV, about 0.75 second latency
over the terrestrial broadcast as the signal is encoded as MPEG2/MPEG4
sent down the wires and decoded by my cable box. 2 TVs in 2 rooms are
out of sync, clearly.
That's just the way the technology works - I don't phone my cableco to
complain about the delay.
Come on Microsoft! Am I the only one who feels this way?
Yes, you are - everybody else has understood this point and work
around it ;-))
All streaming solutions work this way, whether made by Realnetworks,
Apple, Macromedia or Microsoft. You cannot polish a turd, so to speak.
There are certain basic, physical, real-world limitations to do with
look ahead that are *required* to make codecs work well !
Cheers - Neil
.
- References:
- Re: Audio delay under 10 seconds?
- From: Mike Lowery
- Re: Audio delay under 10 seconds?
- Prev by Date: Re: Stream synchronization
- Next by Date: Re: Live microphone feed, Just want visualization in real time
- Previous by thread: Re: Audio delay under 10 seconds?
- Next by thread: Re: Audio delay under 10 seconds?
- Index(es):
Relevant Pages
|