Buffers and real-time

distiller · 22nd October 2010, 18:44

I am new to Shoutcasting but I already tried versions 1 and 2 and read several FAQs. However, I don't understand the buffering. Currently I have 1.9.8 installed and I'm streaming mp3 to another computer over public network. I can get the delay (from sender to actually playing at receiver) to about 1 second. Nice. But how does the buffering system exactly work? I could figure out things by testing but I'd like to understand it fully.

I am trying to set up a high quality communications channel where I could speak to receiver with minimum delay. How to minimize the delay?

Is it possible to make the receiver always "real-time", to have receiver playing at the present moment in sender's playback when they join in? For me it seems to have up to something like 1 minute buffer, where the receiver starts hearing 1 minute from the past if they joined in later than the source playback actually began.

dotme · 22nd October 2010, 19:31

There's a built in buffer with the DNAS, and another in most players. The player buffer is end-user configurable, so you don't have a lot of control over it. If everyone leaves things to their defaults, a 128kbps stream usually is delayed 15-20 secs.

Shoutcast isn't skype - there will always be some delay and no two listeners will hear the same thing at the exact same moment.

distiller · 22nd October 2010, 19:41

Ok, but I do get it to only 750ms delay. Here's one way how I can do that:
1. start the server, source Winamp starts playing
2. (quickly) connect the receiver Winamp
3. now you can pause the source Winamp
4. wait until the receiver catches up to the point where the music stops
5. now when you play again the source, they are almost in sync!

So, how to force to have this server buffer to minimum? Obviously that buffer must be set somewhere, perhaps it's not user configurable.

22nd October 2010, 18:44	#1
distiller Junior Member Join Date: Oct 2010 Posts: 2	Buffers and real-time I am new to Shoutcasting but I already tried versions 1 and 2 and read several FAQs. However, I don't understand the buffering. Currently I have 1.9.8 installed and I'm streaming mp3 to another computer over public network. I can get the delay (from sender to actually playing at receiver) to about 1 second. Nice. But how does the buffering system exactly work? I could figure out things by testing but I'd like to understand it fully. I am trying to set up a high quality communications channel where I could speak to receiver with minimum delay. How to minimize the delay? Is it possible to make the receiver always "real-time", to have receiver playing at the present moment in sender's playback when they join in? For me it seems to have up to something like 1 minute buffer, where the receiver starts hearing 1 minute from the past if they joined in later than the source playback actually began.

22nd October 2010, 19:31	#2
dotme Moderator Join Date: Feb 2005 Location: USA Posts: 3,994	There's a built in buffer with the DNAS, and another in most players. The player buffer is end-user configurable, so you don't have a lot of control over it. If everyone leaves things to their defaults, a 128kbps stream usually is delayed 15-20 secs. Shoutcast isn't skype - there will always be some delay and no two listeners will hear the same thing at the exact same moment. Atlantic Sound Factory Licensed by LoudCity since 2005 \| Hosted by Fast Serv Stream Hosting

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22nd October 2010, 19:41	#3
distiller Junior Member Join Date: Oct 2010 Posts: 2	Ok, but I do get it to only 750ms delay. Here's one way how I can do that: 1. start the server, source Winamp starts playing 2. (quickly) connect the receiver Winamp 3. now you can pause the source Winamp 4. wait until the receiver catches up to the point where the music stops 5. now when you play again the source, they are almost in sync! So, how to force to have this server buffer to minimum? Obviously that buffer must be set somewhere, perhaps it's not user configurable.