similar to: speex voice seems to be bit breaking over long distance.

Hi. After changing my code to construct a JitterBufferPacket and passing that to speex_jitter_put, my program works with the new jitter buffer using the wrapper functions (speex_jitter_* instead of the new jitter_buffer_*). However, a lot of warning: did you forget to call jitter_buffer_tick() by any chance? is displayed. Looking at speex_jitter_get, the logic seems to be as follows: if

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:

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. > >

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

> 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

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

Hello, I think that the new Jitter Buffer have a problem. It works perfectly when I call the speex_jitter_put every 20ms (on my lan) but in other case (with big latency on Internet connexion), it doesn't works. The old version is OK in all cases. Hope it will helps. Thanks Ouss -------------- next part -------------- An HTML attachment was scrubbed... URL:

> 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

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, 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

Le mardi 02 mai 2006 ? 18:15 +0200, Thorvald Natvig a ?crit : > Hi. > > After changing my code to construct a JitterBufferPacket and passing that > to speex_jitter_put, my program works with the new jitter buffer using the > wrapper functions (speex_jitter_* instead of the new jitter_buffer_*). Oops, I forgot to make sure I keep the API stable for the old buffer. Any thoughts on

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 > > >

>> After changing my code to construct a JitterBufferPacket and passing that >> to speex_jitter_put, my program works with the new jitter buffer using the >> wrapper functions (speex_jitter_* instead of the new jitter_buffer_*). > > Oops, I forgot to make sure I keep the API stable for the old buffer. > Any thoughts on the change (revert or continue as is)? Well, I

This patch will export new speex functions in the generated library, such as speex_encode_int as well as the preprocessor, echo-cancellation and jitterbuffers. The ordinals used matches the 1.1.6 release from the speex.org website, so any new library created with this def file should be binary compatible with that one. -------------- next part -------------- Index: speex.def

Dear Jean, > Actually, 38 is normal for the 15 kbps mode. For the rest, could you > give more details. Also, what platform is the problematic machine? (What > CPU, what OS, what compiler). Thanks for your reply. The platform is WindowsXP Professional with SP1. CPU is Celeron CPU 1.70GHz. Compiler is VC++ 6.0. I have enabled VAD and DTX . In this computer, speex_bits_write() still

('binary' encoding is not supported, stored as-is) I'm working on a voip project with Speex and the only major design obstacle left (that I'm aware of) is the jitter buffer. Before I attempt to write something of my own I'd like to try out the jitter buffer functions in Speex's unstable release. Can anyone give me a basic run down of how the experimental jitter buffer is

[The following is perhaps a long question and even off-speex topic,] [but if anyone can at least point me in the right direction for ] [alternate sources of information, I'd really appreciate it. ] When speex_jitter_get(...) is called and there is no buffer/data to return, would I not want to know that there is no true data to play? If I turn around and queue 20ms of silence to play

Dear all, I have implemented speex. Under LAN environment, everything is working fine. However, when the source and destination is about 20 hrs away, with ping response time of about 800ms, the voice is breaking. Breaking means you can not hear a smooth voice. Like the voice is being "chopped" into many pieces. The amount of packet lost is less than 10%. I have tried 8KHz, 16KHz, 32KHz.

Hi Jean, > > > Well, just trace it, how many times are you > calling > > > speex_decode_int()? > > > > Maximum is 51 times per second. Will this cause > any > > CPU high utilization? > > That's normal... What CPU are you using? If it's a > fixed-point CPU, then > the reason is probably just the fact that the packet > loss

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