search for: prebuffers

What is the best way to pick a prebuffering length for a streaming audio application using UDP transport? I'm using Speex in a VoIP application with RTP transport, currently with a fixed 500ms prebuffer on the playback side. However, I'd like something a bit more adaptive to accomodate high-jitter connections. For example, in one test configuration there is a very low average

Ok, this is a silly question, but what does the jitter buffer do? I'm really new to audio, so please bear with me. From what I gather (primarily from the list archive), the jitter buffer is a wrapper around the Speex decoder. I give it the packets I receive, in whatever order I receive them, and then it gives me back a clean stream of audio samples. But what I don't entirely

Hrm, yes it does sound like a bug in WinAmp. Even if it does prebuffer, the length counter should be going beyond 0:01 since you've obviously been listening longer than that after the first second. =P (I don't think the counter resets after buffering, but I could be wrong.) Can you see if you can duplicate it by getting someone else to tune in to the 'cast with WinAmp?

Ah, I'm sorry, I have read the manual and believe I have a reasonably good grasp on how to use the Speex encoder and decoder altogether. In fact I've been using it with great success in my P2P SIP/RTP VoIP application for almost a year now; it's been working wonderfully and I can't thank you enough. However, the manual makes no mention of the jitter buffer, nor does it (so

I strongly suggest you start by reading the Speex manual (you can skip the technical parts about CELP). If you still ask questions, then post them. Jean-Marc Le mardi 14 juin 2005 ? 22:30 -0700, David Barrett a ?crit : > Ok, this is a silly question, but what does the jitter buffer do? I'm > really new to audio, so please bear with me. > > From what I gather (primarily from

Sounds like WinAmp isn't storing a large enough buffer, or Icecast isn't putting a long enough buffer between what it recieves from Oddcast & what it transmits to clients. I've never had such a problem, though. Could be your connection? >===== Original Message From John Farnsworth <si@darkness.nu> ===== >Hrm, but it works fine (and the mountpoint is /dnumusic.ogg).

Noticing a strange issue, i'm not sure where this lies. I'm using oddcastDSP to stream Ogg to icecast2, which seems to be working ok, but using winamp (v2) to stream from icecast, winamp is constantly prebuffering, displaying a length of 0:01. Is this a known bug in either Winamp or icecast, or a configuration issue perhaps? I have a feeling this is Winamp, but I was wondering if anyone

Have you looked at the Speex (adaptive) jitter buffer? See speex_jitter.h Jean-Marc Le mardi 14 juin 2005 ? 17:50 -0700, David Barrett a ?crit : > What is the best way to pick a prebuffering length for a streaming audio > application using UDP transport? > > I'm using Speex in a VoIP application with RTP transport, currently with > a fixed 500ms prebuffer on the playback

On 19/02/02 03:06, Jack Moffitt shaped the electrons to say: > > The point is, is this list related to icecast2 development? I made some > > changes to the server and sent a patch to Jack Moffitt, but I'm not sure > > he's still mainaining the server. > > You're in the right place, and I still maintain the server. I don't > remember your patch right

I send a patch again the last CVS version. The changes are: - Configurable prebuffer, in seconds. - prebuffering parameter in configurable in <limits>...<prebuffer>seconds</prebuffer>... - Created a new function (send_client_queue) to send the client->queue. Better modularisation. - Moved queue_lenght verification to send_client_queue(). - instead of disconnecting when

> Noticing a strange issue, i'm not sure where this lies. I'm using > oddcastDSP to stream Ogg to icecast2, which seems to be working ok, but > using winamp (v2) to stream from icecast, winamp is constantly > prebuffering, displaying a length of 0:01. Is this a known bug in either > Winamp or icecast, or a configuration issue perhaps? I have a feeling > this is Winamp,

> Just in case, I copy the mail and attach the patch. > > I am working in my second version (not included in the patch), I would like > to hear some comments: > > - prebuffering parameter (used in previous patch) in now configurable in > <limits>...<prebuffer>seconds</prebuffer>... > > - instead of disconnecting when the client-queue > 25, I

> - Configurable prebuffer, in seconds. > > - prebuffering parameter in configurable in > <limits>...<prebuffer>seconds</prebuffer>... We're in the process of redoing some of this. We will prebuffer some etc, so in the end this functionality will be there. I apologize that this patch won't get applied for that reason. > - Created a new function

Hi, Just wanted to clarify, can icecast2 not read OR broadcast any mp3, or can it just not broadcast mp3 (but still play from a playlist)? I know the quality would be dropped if it had to reencode them though. And what's up with that POS Winamp3, it doesn't even play gg streams?!? I use xmms, but we have winamp3 on a demo computer, it just sits and keeps prebuffering. I need something

Hi, find enclosed a patch to implement faststart (or prebuffering). Basically, it stores a number of "refbuf"s, configurable by seconds. When a new client connects, it adds the prebuffers to que clients' queue so they are delivered as fast as permitted by the connection. As result, the players prebuffers are filled up faster giving a better "response". I tried to minimise the impact, both in source code and execution. If faststart is configured to zero, it has also...

The problem with WinAMP 3 is that it will accept only ogg data with the old MIME type "application/x-ogg", but current icecast sources emit the new and correct mime type "application-ogg". WinAMP must reject the ogg data because it didn't match the MIME type it expected, so yes, it just sits there prebuffering, and prebuffering, and prebuffering... I fixed this by

We're running a setup with M3W on a PC in Sri Lanka, encoding to a server in the UK. What factors govern the time delay between things being said to the microphone in SL and heard by a client connecting to the server, and how can we reduce it as much as possible? Currently we're experiencing 20-30 secs or more.

Hi all, I am ricardo galli, I've juscribed just a couple of days ago. I wanted to stay as a lurker for a longer period, but this list has almost no traffic, so I wake up due to the silence. The point is, is this list related to icecast2 development? I made some changes to the server and sent a patch to Jack Moffitt, but I'm not sure he's still mainaining the server. If he

> > Is it true ? And if it is, is there any way to change that behavior ? > > Yes. Icecast always sends data at the data rate. This behavior will > not change in the 1.3.x branch. We will however try and make > prebuffering as fast as possible in 2.x (it's currently done the same > way, but we plan to add this feature). Doesn't icecast send the data out as fast as

Dear Group, I'm running the following setup; Yoper v2, Kernel 2.6.8.1-7, Wine 20040914 on a Toshiba Satellite Pro 4600. The Toshiba has a Yamaha AC-XG. In addition I have a USB Plantronics DSP100. After some tweaking I got wine to install x-ten lite; I have pasted my config file; [WinMM] ; Wine supports the following sound drivers: ; winearts.drv ; for KDE ; winealsa.drv ; for