similar to: Jitter buffer latency

Hi, Our project is using the jitter buffer feature built in Speex. We noticed there are some latency when using the jitter buffer. Does anyone know what is the "worst case" latency inherent in the jitter buffer algorithm? I believe someone already mentioned that it's adaptive but is there a worst case hard number (in terms of 20ms Speex frames)? I'm not familiar with the

Hi, sorry for the repost again, but does anyone have a code snippet example of how to use the jitter buffer? Regards, Andy ----- Original Message ---- From: Andy Ngo <ndno72-speex@yahoo.com> To: speex-dev@xiph.org Sent: Wednesday, January 10, 2007 8:09:10 PM Subject: [Speex-dev] Sample snippet of how to use jitter buffer Hi, I searched around in the Speex manual and API but

Thanks Jean-Marc. Is there a Speex trunk for the C64xx I can use? Or do I have to figure what options to turn on (in configure) when building the library? Thanks. Regards, Andy Ngo ----- Original Message ---- From: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> To: ndno72-speex@yahoo.com Cc: jtrantow@ieee.org; speex-dev@xiph.org Sent: Tuesday, December 5, 2006 2:52:02 PM Subject:

Tried your fixed_generic.h change but that didn't help. Andy ----- Original Message ---- From: Andy Ngo <ndno72-speex@yahoo.com> To: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> Cc: speex-dev@xiph.org Sent: Wednesday, December 12, 2007 4:13:35 PM Subject: Re: [Speex-dev] 4kbps sounds robotic on TMS320C64 Jean-Marc, Yes, fixed-point is enabled (#define FIXED_POINT in

Hi, Is there a white paper describing how the jitter buffer in Speex works and what "side effects" (such as introducing delay) can happen? In the past, I have implemented a similar "dual buffer" algorithm to try to smooth out lost or delayed packets with a fixed know delay (no larger than 2 times the Speex frame). Thanks. Regards, Andy -------------- next part

Hi, I looked at the jitter buffer code and it seems like the maximum number of frames that Speex can hold up to is 200 (SPEEX_JITTER_MAX_BUFFER_SIZE). This is equivalent to 4 seconds (20msec/frame * 200 frames). Can I just reduce this constant to limit the size? I know I'll reduce the smoothness of delayed frames but I want to reduce the delay in case my audio application isn't

Andy Ngo wrote: > Thanks Jean-Marc. Is there a Speex trunk for the C64xx I can use? Or > do I have to figure what options > to turn on (in configure) when building the library? Thanks. Support is in the main version. Just look at the TI/ directory for extra TI-related stuff. Jean-Marc > Regards, > Andy Ngo > > ----- Original Message ---- > From: Jean-Marc Valin

Hi, I'm trying to use the jitter buffer feature that comes with Speex but I'm getting unexpected latency. I wrote a client application that does VOIP-like functions and without using jitter buffer, the end-to-end latency is around 250 ms (I'm using lowband 5.97 kpbs). However, when I tried to incorporate the jitter buffer feature, the latency would grow as time elapsed (up to a few

Hi Jean-Marc, I have tested Speex in fixed-point mode on my PC without clicking noise. As I mentioned below, I tested Speex in fixed-point mode running natively on the ARM side of the DaVinci without click noise. I only get the clicking noise when running Speex on the DSP side. For the TI C64+ DSP on the DaVinci processor, the TI-specific switches doesn't do much other than defining the

Jerry, Thanks for the response; basically, I need to get just the narrow band working now (maybe use advanced features such as jitter buffer, echo cancellation, VAD, etc in the future); the DSP will do the encode and decode simultaneously as data are available. My main objective is just do a voice application that runs on the ARM side of the DaVinci with the codec running on the DSP side; the

Hmm, I tried... I completly understand an idea of jitter buffer and I know there is a lot of kinds of this solution (eg. AJB - Adaptive Jitter Buffer). I simply want to know what type is used in speex codec and how could I use that. What is the reason for using jitter buffer implemented in speex against to my own (implemented at lower layer - transmission layer - eg. rtp). Kapul On Tue, Sep

Hi, I have a SIP client which is registered to asterisk. Asterisk is registered to a SIP trunk and also handles the media. Now since my client has some issues in its RTP Tx, which seems to have some amount of jitter (mean jitter as per ethereal trace is about 17ms, max jitter is 20 ms and max delta is 85 ms), to over come that I have enabled jitter buffer in the SIP channel by setting sip.conf

Good question. I'm coming to the conclusion that using plain UDP and "home-grown" packet construction for transmitting the speex data (with timestamp/sequence counter) and implementing jitter control on the receiver end is an adequate implementation for a VoIP application. Assuming of course that I don't care about any interoperability issues with other applications etc. I was

Hi All, I have a question regarding the nominal jitter buffer configuration: The call was setup as G.729A (annexb=no), ptime=20ms and nominal jitter buffer size = 50ms, and round trip delay is 200ms, the TDM side will experience intermittent one way voice during the call, but IP side can always heard the voice from TDM side. My question is, should this possible caused by the nominal jitter

I use the speex version of your jitter, and in speex_jitter_get, you always call the jitter_buffer_update_delay. -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: dimanche 18 mars 2007 13:06 To: Ouss Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Problem with the svn jitter buffer > I think that the new Jitter Buffer have a problem. > >

Greetings list, Some time ago (probably about a year ago now) we disabled IAX jitter buffering on all our boxes because it was causing issues in a mixed 1.0 and 1.2 environment. One thing I've noticed over the last few months as more and more clients have moved from the 512k/1mb/2mb ADSL connections they were using onto "up to 8mb" connections is that whilst overall throughput is a

Hi, I know that there was a jitter buffer patch (for sip) for the 1.0.9 branch some time agin. At this time, we can not upgrade to 1.4.x. Is there a useable, fairly stable INCOMING sip jitter buffer patch? That is.. I want Asterisk to jitter buffer incoming SIP packets. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Jean and Steve, Can you tell me whether the jitter filter / buffer is adaptive type, I saw the description of speex_jitter.h say it is "adaptive", anyone of the group has implemented it and confirm it. Thank you all. Regards, Danny Chan -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Jean-Marc Valin Sent: Tuesday,

Hi List, I have a question regarding jitterbuffer in Asterisk 1.4.24. I see that jitterbuffer is only effective on the receiving channels. My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch office. Questions: 1. To enable jitter buffer on SIP channels it seems I have to enable and force it, right? 2. If I enable and force jitter buffer, Asterisk would always have to stay

Since r12660, the speex_jitter_get with high latency doesn?t works, I have no sound. Before this release, the speex_jitter_get works in all conditions. speex_jitter_get return void, then I cannot know the reason of this problem. Regards Ouss -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: dimanche 18 mars 2007 23:07 To: Ouss Cc: