similar to: About speex_jitterbuf_get

> (PS, if you do use threads, protect speex_jitter_put/get with a mutex > (CRITICAL_SECTION I believe they're called in Win32Speak) -- calling put > and get at the exact same time from different threads leads to "features") I've never tested this, but I designed the jitter buffer to work from two threads even without using a mutex. This would work as long as there is

> Err, unless I'm totally wrong, there are a few race conditions. > > Assume the buffer is full of packets newer than the current pointer, and > one that is at the current pointer. > > get and put start at the same time. > > get will find the correct buffer index. Now, just after it finds it's > index, assume we switch to the put thread. > > Put needs

Jean-Marc Valin wrote: >>OK, I'm actually about ready to start working on this now. >> >>If people in the speex community are interested in working with me on >>this, I can probably start with the speex buffer, but I imagine >>there's going to be a lot more work needed to get this where I'd like >>it to go. >> >> > >And where

Jean-Marc Valin wrote: >>Heh. I guess after playing with different jitter buffers long enough, >>I've realized that there's always situations that you haven't properly >>accounted for when designing one. >> >> > >For example? :-) > > I have a bunch of examples listed on the wiki page where I had written initial specifications:

Hello, I'm experiencing some problems with running IAX2 protocol on quite reliable link with G729A codec. My customer has 2mb FR link to the Internet used in about 20%. Ping statistics: 50 packets transmitted, 49 received, 2% packet loss, time 49496ms rtt min/avg/max/mdev = 9.308/13.126/33.307/4.851 ms Everything would be great, but the quality isn't good enough. I have 2mb/512kb DSL

Hi, I'm a developer currently using the speex codec in a VOIP application of ours.. It sounds amazing, especially at wideband.. my question is how do I force it to do a concealment? We have a low latency application, and based on the current API, I'm guessing concealment only kicks in when a packet is lost.. However, our jitter buffer knows when a packet is missing, and I'd like to

Jean-Marc Valin wrote: >>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

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, Ok, what you suggest sound logical to me. Currently, I have done a small trick to prevent this problem. What I did is that whenever windows request a voice packet from me and if I do not have the voice packet, I repeat the previous packet. Hence, all the breaking portion is filled with previous packet. This trick seems to work so far. I am not sure what is the side effect. I think jitter

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

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

Daniel Schmidt a ?crit : > I found the cause of the problem. The function shift_timings can > produce overflows in the timing array if the jitter is huge or the > time units are very short. After changing the timing values' type from > spx_int16_t to spx_int32_t it seems to work. Hmm, I always assumed there wouldn't be any overflows. What parameter range are you using that

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

Hi, We are planning to use Speex as the speech codec for a VoIP application. 1) May I know how Speex compared with GIPS codec? It seems that Google, Yahoo, and Skype are licensing from GIPS. Are there any good benchmarking or fair comparisons? 2) In particular, how is the jitter buffer control for Speex in response to intermitent poor connection hiccups? Is it robust enough to smooth out

[sorry for the loss of proper attributions, this is from two messages]: [Me] >This is something I've encountered in trying to make a particular > asterisk application handle properly IAX2 frames which contain either > 20ms of 40ms of speex data. For a CBR case, where the bitrate is > known, this is fairly easy to do, especially if the frames _do_ always > end on byte

Hello, I've only one question: how does a jitter buffer work? Regards Kapul

Hello, I understand that only the destination of a call should do jitter buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no transfers), PhoneA and PhoneB need to perform their own jitter buffering, and Asterisk will just forward the frames, correct? What happens if the peer does not support jitter buffering, but is close by so there's no need for jitter buffering? My

Danny Chan wrote: >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. > > 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

Hi everyone, Judging from the lack of complaints about the split, I think I can go safely forward with that. The only thing to decide is the name of the new library that will contain the preprocessor, the AEC, jitter buffer, and resampler. So far, I've got a few potential names: - libspeaks - libspeexproc - libspeexvoip - libspeech - libspoke Let me know (in private so as to not flood the

Hi, Is there an easy way to get the size in samples of a Speex packet, without decoding the packet? If I receive a "narrowband packet" with - Zero or more wideband frames (must be skipped apparently) - Zero or more Speex inband requests - Zero or more user inband requests - One or more narrowband frames I need to know how much samples the packet contains to calculate jitter buffer