similar to: How to reduce jitter buffer size?

Hi Jean-Marc, Thanks for your response. Given a worst case scenario, what is the "worst case" latency (in terms of Speex frames) that the jitter buffer algorithm will incur? We're trying to determine the worst case hard number. Sorry for unclear question below; what I was trying to ask is that given a worst case latency (which I'm asking in the first question) inherent in

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

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

Is is possible to give a short hint about how the jitter buffer would "catch up" when network condition have been bad and then get better? I'm using the jitter buffer with success now, but sometimes I have a long delay that's caused by bad network conditions and then later when the conditions get better, I would think we would want the audio to gradually catch up with real-time

Thank you for a very good explanation which shed light on some of the questions that I had after reading the source code. Reading your text however, I wonder if I'm perhaps missing an important point on the proper use of the jitter buffer: ... > Now, clearly, if early_ratio is high and late_ratio is very > low, the buffer is buffering more than it needs to; it will > skip a frame

> FYI: The below is just my interpretation of the code, I might be wrong. Most of it is right. Actually, would you mind if I use part of your email for documenting the jitter buffer in the manual? > Each time a new packet arrives, the jitter buffer calculates how far ahead > or behind the "current" timestamp it is; this is called arrival_margin. > The "current"

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, 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, I am using the JitterBuffer. Since there is not so much documentation I think I dont use it in a correct way. All the packets are recieved (I control the sequence numbers) but the JitterBuffer often tells me he has no packet. I am using it in the following way: I am not sure if I use the ticks correctly but I think it can be set to 20(msec). It is set as a Member in my class and i

> > Is is possible to give a short hint about how the jitter buffer would > "catch up" when network condition have been bad and then get better? > > I'm using the jitter buffer with success now, but sometimes I have a > long delay that's caused by bad network conditions and then later when > the conditions get better, I would think we would want the audio to

Thanks for your reply Jean-Marc! this was what I had before. But I decided to restructure it since the thread that plays the sound is a callback from the sound hardware, more or less an interrupt handler. For me it seems more reasonable to waste some memory for to save the decompressed Packet. While I write this I begin to think that it is possible I decompress Packets that are never used

>> FYI: The below is just my interpretation of the code, I might be wrong. > > Most of it is right. Actually, would you mind if I use part of your > email for documenting the jitter buffer in the manual? It would be my pleasure :) >> early_ratio_XX is the sum of all the positive bins. >> late_ratio_XX is the sum of all the negative bins. > > Right. And only the

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:

(Sorry about the delay -- currently attending ICASSP) Hi, Haven't looked at all the details, but what's clearly wrong is that you need to put the *compressed* packets in the jitter buffer and decode them only when you _get() them. Jean-Marc David Feurle wrote: > Hi, > > I am using the JitterBuffer. Since there is not so much documentation I > think I dont use it in a

> Thank you for a very good explanation which shed light on some of the > questions that I had after reading the source code. > > Reading your text however, I wonder if I'm perhaps missing an important > point on the proper use of the jitter buffer: > > ... >> Now, clearly, if early_ratio is high and late_ratio is very >> low, the buffer is buffering more than

David Feurle wrote: > Thanks for your reply Jean-Marc! > > this was what I had before. > But I decided to restructure it since the thread that plays the sound is > a callback from the sound hardware, more or less an interrupt handler. > For me it seems more reasonable to waste some memory for to save the > decompressed Packet. While I write this I begin to think that it is

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, I thought I'd repost this to the -users list for some background on the jitter buffer and its workings and remaining issue.s I'll also pu a little executive summary here at the top: Where a channel is native bridged to another iax2 channel: 1) Lag is not measured and will usually show 0ms. Any other number is an old measurement from the start of the call 2) The jitter

Hello, The way you describe how the jitter buffer should be implemented makes me wonder: How does the jitter buffer works when there is no transmission? Let's say my "output" thread gets a speex frame from the jitter buffer every 20ms. What happen when there is no frame that arrived on the socket? No frames at all for a pretty long time (ie many seconds). This is my case because I

Hi there, I am reading the 2004 paper "Smoothing with mixed model software" in Journal of Statistical Software, by Ngo and Wand. I tried to run their first example in Section 2.1 using R but I had some problems. Here is the code: library(nlme) fossil <- read.table("fossil.dat",header=T) x <- fossil$age y <- 100000*fossil$strontium.ratio knots <-