similar to: Jitter buffer

Hello Jean-Marc, thank you for your reply. > > 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.

Hello all, I am currently investigating the JitterBuffer struct provided in the Speex library, and I am actually thinking about using it with two different codecs: namely, Speex-NB and AMR-NB. From looking at the code, it seems that JitterBuffer is capable of working for any codec. Both Speex-NB and AMR-NB (and probably also other narrowband codecs) produce 20 ms frames and the sampling frequency

Hi Zachary, Zachary Schneirov a ?crit : > The speex_jitter_buffer.c wrapper seems to maintain (buffer?) one > packet-frame ("current_packet") in addition to the packets already > tracked by the JitterBuffer itself. Why is this necessary? That's in case there's more than one frame per packet. That way, it finds the packet, decodes the first frame it contains, and

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. > >

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

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:

Dear speex developers and users, I'm considering adopting the speex jitter buffer for use with a different codec in a voice conferencing system and would be very grateful if those more acquainted with it could help me with some questions. The speex_jitter_buffer.c wrapper seems to maintain (buffer?) one packet-frame ("current_packet") in addition to the packets already

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 > > >

I am trying to using Speex for some Wireless voice transfers. I am using the 8K narrow band but am feeding it 11025 sample rate. Each Speex frame is about 14.5ms instead of 20ms. In order to minimize WiFi packet traffic I am sending 7 Speex frames at a time for a packet rate of about 10 per second so each WiFi packet has about 101ms of voice data. I have three modes 1. Simplex - voice messaging

(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

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

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

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:

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

Sorry it's asterisk-users@lists.digium.com --- harry gaillac <gaillacharry@yahoo.fr> a ?crit : > Luca, > > you may find help here: > > http://www.cs.colostate.edu/~somlo/CSU-SIP-notes/ > http://www.asteriskdocs.org/ http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large > > ask for help to asterisk-users@lists.digium.org > > Regards >

>> 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

Hi, I'd like to report a bug in speexclient/speex_jitter_buffer.c. In the function speex_jitter_get(), after the line: packet.data = data; add the line packet.len = 2048; If this variable is uninitialized, then in jitter_buffer_get() (in jitter.c), the predicate "jitter->packets[i].len > packet->len" is undetermined. When this happens, the wrong amount of

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

I've been trying to resolve some quality issues and I was hoping someone might be able to provide some insight. To give you an idea the calls are coming in via a SIP DID and sent out via an IAX2 connection. Latency to both the SIP equipment and IAX equipment are around 80ms with 0 packet loss accoridng to ping tests. The server is located in a data centre so bandwidth is not an issue. Most

> I'd like know about anyone using the current jitter buffer in Speex. I'm > planning on changing it to make it more general and I'd like some > feedback about how to make it better. Also, let me know if you're doing > anything serious with it and want to make sure I don't break your stuff. > > Basically, I want to make the jitter buffer easier to use with