Old 9th April 2007, 22:40   #1
Robert_B
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Live streaming with the lowest delay/latency?

The subject says it all, I guess.

The question is: How to stream live content with the lowest delay for listeners?

Back in the old days, buffering and slow internet connections were all over the place, so Shoutcast put in a buffer (I think it's a 1 MB buffer since a 128 kbps stream is delayed for half a minute or so)

I found out ShoutCast Server version 1.7.1 does NOT have a buffer, so what is put into it the listeners will hear almost instantly. And THAT is what I want

Of course, 1.7.1 is ancient technology and vulnerable to whatever attacks, and then some. And no AAC+ streaming possible either.

Oh the other side, Windows Media Server has a great option to disable the server-side buffer (for synchronized streams).

Of course, a stream will not start instantly but that is no problem. It's only a few seconds with the default Winamp settings most listeners have (a 128 kB buffer for streams).

A good example is to listen to a Live365 Live or Relay stream with Winamp (or any other player). Their proprietary Nanocaster "bit hurling" servers don't buffer a byte.. What comes in goes out.

So... why can't we have an option buffer/quick start. This is a (retro) feature request but I could not find the right topic so I will post it here.

To clarify, I am asking this because I run a live show with a chat room, and some listeners still hear the previous song for up to 20 seconds.

In a live/interactive environment, that sucks! Ok, Shoutcast DNAS 1.7.1 will do for now but as I said it's ancient and doesn't do AAC+

Please give us an option "to buffer or not"
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Old 9th April 2007, 22:44   #2
tuckerm
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Buffer is all on the client side and server side. If you are hosting your own server on your connection, The higher the bit-rate the lower the delay.
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Old 9th April 2007, 22:45   #3
Nick@ss
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he means the higher the bitrate the lower the delay i think
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Old 9th April 2007, 22:49   #4
djSpinnerCee
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Buffer size = time

The higher the bitrate, the less "time" will be buffered since most players buffer bytes irrespective of bitrate. The DNAS also buffers a fixed amount of data.

So, relatively speaking, a higher bitrate stream will not be as far behind the source as a lower bitrate will be.
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Old 9th April 2007, 22:50   #5
tuckerm
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Quote:
Originally posted by nick@ss
he means the higher the bitrate the lower the delay i think
Holy cow, I never caught that. Edited, Thanks Nick
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Old 9th April 2007, 22:55   #6
Robert_B
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Quote:
Originally posted by nick@ss
he means the higher the bitrate the lower the delay i think
You are right, nick@ss, and tucker was close :;

Current/newer versions of Shoutcast Server have a buffer to quick-start listeners and care for source drops.

But since my server is on the local network (hell, it's even on the same machine that takes the soundcard input) I would like the internal buffer turned OFF, so listeners do not have to wait 30 seconds or more to hear "what you play" (I agree, their Winamp or other player will account for some delay).

1.7.1 does NOT buffer, but I would like an option in a later version to turn it off. That's all.

Thanks for the quick reply, both of you.
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Old 9th April 2007, 22:59   #7
bored_womble
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how about you just use Icecast, it has a configurable buffer size so you can make any changes you like.

BW

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Old 9th April 2007, 23:06   #8
Robert_B
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Quote:
Originally posted by djSpinnerCee
Buffer size = time

[edited]The DNAS also buffers a fixed amount of data
Right... based on calculations that would be about a megabyte or so. It provides for a stable stream and quick start I agree but also causes latency for listeners.

Listeners will hear what's in the start of the buffer instead of what's playing now (up to 30 seconds later for a 64 kbps stream so I guess the buffer is 512 kB in the server)

All I request for is to be able to turn if OFF, even though it will cause more buffering on the client side. I'm talking about a LAN scenario here (or other synchronized content).

Thanks for the quick reply again.
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Old 9th April 2007, 23:26   #9
Robert_B
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Quote:
Originally posted by bored_womble
how about you just use Icecast, it has a configurable buffer size so you can make any changes you like.

BW
Thanks for that.

I will look into how Icecast is compatible with Winamp/WMP/QuickTime/RealPlayer on the client side.

As long as I can give listeners a .pls .asx or similar URL to listen without having to install plugins, that will do.

For now, I am looking into SimpleCast. I guess this is not the forum to talk about that, but I had good and bad experiences with OddCast back in the days (especially title streaming, through DSP stackers).

At home, I still use the good old ShoutCast 1.8.0 Fraunhofer based encoder for music (below 128 kbps, at least).

I guess AAC+ still has a long way to go, to take over mp3 streaming, including player support and URL standardisation (?spell)

Admitted, I love Windows Media because it makes things so easy but as a long time Winamp/Shoutcast user I prefer to do things the old school way..

Thanks anyway.
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Old 9th April 2007, 23:30   #10
bored_womble
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just so you know

i) Icecast can be used with the same DSP as Shoutcast

ii) it can be configured to work exactly the same way as Shoutcast for the client side.

BW

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Old 10th April 2007, 13:51   #11
FS-Randy
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I second the Icacast option -- it is really what you are looking for, no joke!
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Old 7th June 2007, 15:55   #12
Greg_E
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Pulled this up on a search...


I only have about 10 seconds delay on my system with 32K AAC+. Most of the delay seems to be the buffer in the client as it changes depending or whether I'm using Winamp, VLC, or WMP. However I just put up a low quality MP3 stream with a different encoder, this is delayed by almost 1 minute.

AAC+ encoder is the Orban Opticodec, MP3 encoder in Winamp with the 1.9.0 DSP plug-in.

Your mileage may vary!
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Old 7th June 2007, 17:07   #13
djSpinnerCee
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The delay is always there as the DNAS buffer will be there regardless of client.

It cannot be eliminated.

Decreasing server buffer sizes and stuff will reduce the delay, but also decrease reliability and increase rebuffering -- so expect more complaints from listeners who are far away or have little free bandwidth or are doing other things on their computer while they listen. Also expect this to be inconsistent to the extent that some listeners will have no trouble at all while at the same time some listeners will not be able to even get the stream started.
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Old 8th June 2007, 04:36   #14
bored_womble
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some source clients support the

icy-prebuffer:

tag, some do not. I think WinAMP does (oo shocker) but not sure about others. I believe it tells the Shoutcast server how much data quite obviously will be pushed into the buffer before normal streaming occurs.

You can see that when a client does not support the pre-buffer and it is connecting to a freshly started Shoutcast server any listeners connecting may not work for 5-10 seconds as the Shoutcast server does not have enough data to spool out.

BW

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