search for: prebuff

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,...

...ackets (as I'm witnessing), what advantage does the jitter buffer offer over going straight to the Speex decoder? 2) Assuming samples arrive at an average rate of 22KHz, but arrive in a highly irregular fashion, is there any way to ensure regular playback other than to just wait some "prebuffer" duration before beginning playback? How do I pick the smallest prebuffer duration to accomodate a given connection's jitter? 3) Assuming I want to deliver samples at a rate of 22KHz, what's the best graularity at which to encode and broadcast? Granted, I need to stay beneath...

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 &...

...buffer as you suggest, and I see how it can impove playback quality in high packet-loss situations considerably, but I haven't yet wrapped my head around what benefit it offers in reliable, high-jitter environments. So far as I can tell, the only solution to jittery transport is an adequate prebuffer, and thus I'm looking for advice on how to determine what "adequate" means. Likewise, I can easily broadcast anywhere from 33ms to 500ms audio packets (I currently use 50ms), but I'd like to hear your real-world advice on what the ideal packet size is I should be using. Tha...

...ssing), what advantage does the jitter buffer offer over > going straight to the Speex decoder? > > 2) Assuming samples arrive at an average rate of 22KHz, but arrive in a > highly irregular fashion, is there any way to ensure regular playback > other than to just wait some "prebuffer" duration before beginning > playback? How do I pick the smallest prebuffer duration to accomodate a > given connection's jitter? > > 3) Assuming I want to deliver samples at a rate of 22KHz, what's the > best graularity at which to encode and broadcast? Granted,...

...m 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). It just >constantly says it's prebuffering. It doesn't affect the audio output of >it at all, it just never scrolls the songtitle/etc. Just prebuffer, flash >of title, prebuffer again. > >On Thu, 15 May 2003, Jack Moffitt wrote: > >> > Noticing a strange issue, i'm not sure where this lies. I'm us...

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 knew for sure. --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.ice...

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 side. However, I'd like > something a bit more adaptive to accomodate high-jitter c...

...n the right place, and I still maintain the server. I don't > remember your patch right off-hand but I'm going to look for it now. 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 discard packages wich are older than prebuffer*2. All clients I tried have recovered very well, even with a p...

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 d...

> 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 knew for sure. Sounds like you forgot to name your mountpoint ending in .ogg. Winamp assumes everythign...

> 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 discard packages > wich are older than prebuffer*2. All clients I tried have recovered very...

> - 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...

...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 else for windows I think. Thanks in advance, Bryan --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to 'icecast-request@xiph.org' containing only the...

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 bette...

...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 altering one string in format.c, and another in fserve.c in the current CVS icecast sources, then rebuilding. WinAMP works just fine after you change them both to the old MIME type "application/x-ogg", and the other players I&...

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 it gets it from the source? I am not sure how it could send it out any faster. It is the client that figures out how much it wants to bu...

..."RecordDevice" = "default" [dsound] ;;HEL only: Number of waveOut fragments ahead to mix in new buffers. "HELmargin" = "50" ;; HEL only: Number of waveOut fragments ahead to queue to driver. "HELqueue" = "50" ;; Max number of fragments to prebuffer "SndQueueMax" = "28" ;; Min number of fragments to prebuffer "SndQueueMin" = "12" ;; Forces emulation mode (using wave api) "HardwareAcceleration" = "Emulation" ;; Sets default playback device (0 - number of devices - 1) "DefaultPla...