search for: rebuffered

I have a question about PLC used by OPUS. We are using OPUS for streaming music in some application. Due to some network problems, we are running into problems related to clock drifts. This causes jitter buffer underflows at the receiver and hence we need to rebuffer the jitter buffer periodically. We are looking at the root cause of this problem (drift), but that's besides the point.

After I installed the latest nightly build, it looks like the fallback works without rebuffering. I do see one problem -- when the main mountpoint stream restarts, the fallback continues to play. Once I stop the fallback mountpoint stream, the main mountpoint stream restarts, but only after client rebuffering Also, during the mountpoint switchovers, sometimes the meta-data is lost in the

Hi, <p>I was streaming Ogg Vorbis to a mount point which didn't end .ogg ogg123 played the stream without any problems, but other clients (xmms, Audion) just kept rebuffering and failed to detect that it as an Ogg encoded stream. Surely the client should use the MIME type provided by the server rather than rely on the suffix in a URL ? <p>Is there any attempt to encourage

Hello, I am running icecast 1.3.11 under OpenBSD 2.8-STABLE. On a first look, everything works fine, but after ~1.5 hours of streaming, clients start losing the stream and have to rebuffer every couple minutes. While this happens, the prepuffering starts to take a couple times longer than it did before. It seems to work fine again now that I reconnected both client and streamer, but it appears

Uaaargh ... between now and the last mail, everything's gone kaploohey. But I doubt that it concerns icecast too much in this case - the streaming computer has a cpu load of 90-100% for no reason (it's an athlon 1ghz with only winamp and ssh and the system monitor running under win98). The playback on the streaming computer sounds good, though, but what reaches the client side is no more

6/28/01 1:59 PM jack@xiph.org Dear Jack: >The way I've suggested in the past is to use your m3u or pls files for >this purpose. > >Just make your m3u file ahve: > >http://www.mysite.com/ad.mp3 >http://www.mysite.com:8000/stream Currently I have it set this way: http://www.mysite.com:8000/playlist.pls which forces open an mp3 player. Do I need to include it in the

While maintaining the Relay-Server for Radio Subether [1] the official Chaos Communication Camp [2] Radio Station I noticed some weird behaviour. We don’t have QoS for the Relay-Path, so the Relay started pretty soon to be unstable. Especially the higher bandwidth streams are very often unavailable. To monitor this behaviour I was constantly reloading the "List MountPoints" page AND

I dont know if this is a qustion that I should ask so let me know if it should be directed somewhere else. <p>BTW thanks to everyone who help me in that last thread. It is now working with the config file for ices. <p>Anyway my stream needs to rebuffer about every three seconds. Is it a bandwidth issue? or are there any settings that I can change to help improve this?

un@dom.de wrote: >>i've also linked the stream into dir.xiph.org for testing, >>it shows up, when i click the link, wmp buffers... :( > > > if someone would like to check: > http://dir.xiph.org/index.php?sgenre=&stype=&search=aporee > (don't care about the content ;) > thx, u. You seem to be streaming at all sorts of different bitrates... I'm

Here's yet another ogg123 patch that: 1) Adds a command-line parameter "--prebuffer n" or "-p n" that decodes "n" chunks into the buffer before even forking off the writer thread. 2) Moves the buffer_shutdown call in ogg123.c to its proper place. 3) Doesn't use signals ;) This way, the default behavior is to start playing immediately, while allowing the user

New to the list (and forum, but waiting for a moderator to approve - hence my question here) but not new to icecast. I have a question - I'm running an Airtime install with Icecast - and yes I asked them first but this has flummoxed them over at Sourcefabric - anyone had a problem with a HQ (192kbs) stream rebuffering every few seconds during a FTP/SSH upload to a remote streaming server?

So, I've googled around, and can't seem to find anything regarding usage of limit-rate inside of the mount tags. So... <mount> blah... mount settings here <limit-rate>(How to set the number here?</limit-rate> </mount> Any help would be appreciated. Nicholas -------------- next part -------------- An HTML attachment was scrubbed... URL:

> > Can you refresh your sources from the CVS, and try again? <p>OK, I got it to compile this time using the new sources from CVS. I am able to run the program and I heard myself talking through a microphone, but the client keeps rebuffering every 10 seconds or so. Not sure if this is because of some settings I need to adjust or if it's because the computer is too slow (200

Hi, I have a few problems with Winamp 2.8. (I can't use or recommend Winamp3 because of all the various problems I've had with it. It seems that Ogg streaming is broken on ports other than port 80. However, streaming in general seems not to work correctly. If I point the winamp client at an icecast or gnump3d server, it locates the file or stream fine, buffers a bit, and starts

On Fri, Jan 7, 2011 at 7:36 PM, Brian Willoughby <brianw at sounds.wa.com> wrote: > This thread has raised several good topics. ?It's surprising that the > FLAC-Dev list has been silent for years, and now suddenly there are several > good ideas to discuss. I'll take credit for this, toot toot toot :D > > On Jan 7, 2011, at 15:04, David Richards wrote: >>

> They should. A large number of them are horribly broken (I don't know about > audion, but in the case of both xmms and winamp, the plugin API is designed > such that the _plugin_ has to do the HTTP streaming itself - and the plugin > has to decide whether to handle the play request _before_ it actually starts > the request. I believe this was fixed in winamp3, but winamp3

On 11/06/13 07:16, "Thomas B. R?cker" wrote: > Hi, >> Other datail: all users have always "Lag: 0" on Admin interface > > That doesn't need to be a problem, but without full understanding it's > hard to say. that is just the difference between the latest source offset and client offset, at 0, just like on the main trunk, the listener is at the

On Jul 16, 2009, at 12:53 AM, Aaron Gray wrote: > Chris, > > The basic idea of using templates inconjunction with inlining is for > efficiency. > > 6,500 virtual calls outputting bytes out of 10000 calls, and the > rest 1,750 being words to output 10,000 of code does not entice me > to use virtual calls. I understand that you say that, but I can't bring myself

...to each other without problem. However, when I try to connect with a client, I run into problems: with Winamp 2.8, I get an [HTTP/1.0 200 OK] message, but then the stream is apparently dropped before buffering. With XMMS 1.2.7, the client connects, the stream is buffered, but then dropped, and rebuffered, ad infinitum. Any ideas?? from logfiles: [Icecast access.log] 10.10.20.21 - - [08/Apr/2003:12:13:17 -0700] "GET /ogg HTTP/1.0" 200 9643333 "- " "xmms/1.2.7" 805 10.10.20.11 - - [08/Apr/2003:12:14:33 -0700] "GET /ogg HTTP/1.0" 200 10455570 "- " &q...

This thread has raised several good topics. It's surprising that the FLAC-Dev list has been silent for years, and now suddenly there are several good ideas to discuss. On Jan 7, 2011, at 15:04, David Richards wrote: > I am interested in streaming lossless audio, FLAC is probably the best > option for that. Currently the OggFLAC way of doing it mostly works > with a few hacks in