similar to: Speex specific jitter buffer missing in source

Hi all I have seen that, for speex 1.1.6, there is a jitter buffer implementation. I´d like to know, please, what kind of jitter algorithm it´s been used to implement it. Is there anything like an algorithm overview or something else that someone could send me? Thanks in advance and best regards Guild __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new

Hi, I've a couple of questions about the jitterbuffer. First of all, I'm a bit confused about the code to use. Googling the subject I've find that there are, or there were, two jitter buffers, a generic one and a speex-specific one, is this still true or they've been joined? Downloading the 1.2rc1 source code there's no trace of the SpeexJitter structure and wrapper functions

(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

hi what is the status of thespeex jitter buffer? is it generic or dedicate to speex? can anyone let me know how it works? i am having abit problem understanding it from the code i saw the cost function but not sure i understand the basic assumptions? how does it handle VAD? ifanyone has used it i would appriciate some insight .. thanks -------------- next part -------------- An HTML attachment

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

> Hello, Hi :) First off, could you try to set your email client to break long lines before transmitting? In my (somewhat outdated) pine, the lines appear as VERY long lines when I try to reply, making it hard to read :) Minor detail though, I should probably fix pine. Some day. > The way you describe how the jitter buffer should be implemented makes me > wonder: How does the

Dear all. Questions regarding VoIP implementation and the use of the Speex jitter buffer, if I may: Am I right in my understanding that the Speex jitter buffer implementation is used only on the receiving end of a network VoIP stream? 1) The sender would sample+encode+timestamp packets/frames of speex data and send via UDP to receiver. UDP packet would be constructed as: [TIMESTAMP][Speex

It seems that speex jitter buffer is tightly coupled with SPEEX codec [we have to give a speex decoder instance to JB]. It would be better if we could use it with any codec, like speex preprocessor and AEC. What are the any paper/theory/algorithms behind current ADAPTIVENESS of speex JB? Links to those algo/papers would help to understand better. -- Shantanu --- Thorvald Natvig

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

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

> I believe it is adaptive, but no, I haven't used it, because it's > coupled only to the speex codec. We're working on a generic > application and codec-independent jitter buffer algorithm, for use in > asterisk and iaxclient (at least). Some information is available at > http://www.voip-info.org/tiki-index.php?page=Asterisk%20new% > 20jitterbuffer Yes, this

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

Right - and I deal with that on the receiver end based on an approximation of sender's and receiver's responsiveness - the minimum latency I've been able to get into the system is about 150 ms. Of that, jitter buffering is about 40-100ms. I'd love to figure out how to get that down without killing myself on thread switching or Win32 kernel calls, but ms has to actually implement

The audio frame speex generates sounds pretty terrible most of the time, and I don't use it for jitter correction instead I just use it for dropped packets - so I usually drop the late packet. It sounds acceptable as long as I drop less than 5% of traffic (dropping 2 in a row makes a bad robot noise, so I reset the stream in that case). The good news is that on an unsaturated DSL line jitter

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 Ouss, Can you test again? I think I found and fixed the problem. Jean-Marc Ouss a ?crit : > 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 > > >

Hi Farhan, My mistake, it is 9.6kps because is on gsm ! Regards, Danny -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Danny Chan Sent: Friday, November 19, 2004 2:08 AM To: Ashhar Farhan; speex Subject: RE: [Speex-dev] jitter buffer Hi Farhan, It is interesting that GPRS is 9.6 kps, as per GPRS standard it should be 115kps ? if

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

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

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