similar to: Using speex (newbie)

Hi, i'm using speex in a conferencing application and i have a little problem. I encode single buffer of 320 byte (160 short) and send it to destination where i play it. The problem is that between frame i hear a "tick" so, if i encode single frames, send it and play it, i heard like to be underwater (because there is a tick 8000/320 times/sec ), if i encode many packet all together

Thank you for the answer but i've already read manual and examples and i'm doing same thing of your code, so, maybe you can give me a hint about the problem i have, that it is for sure in my code. A question: is it possible that i hear a "tick" because between two frame i call speex_bits_reset? (for this reason if i encode more than one frame together there is no

Thank you to all, i've discovered the problem, i was using 1.0.5 version, now i'm using 1.1.11.1 and tick is disappeared.... speex_decode now return -1 if end of stream is reached, in previous version it was returning 1...maybe can help someone...thank you again Carlo -------------- next part -------------- An HTML attachment was scrubbed... URL:

(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

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

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

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

In my experience most of the jitter related issues are because people are using too small of audio buffer sizes that match the framing size of Speex - particularly in Windows. This isn't a problem with Speex, but as a programmer you should collect and append a few frames to match the size of your output audio frame buffer before attempting to play the sound. -----Original Message----- From:

Hi all, I have a set of files that are shared from a Windows 2000 Server that smbclient can't access but Windows 2000 clients and DAVE clients can. I am using samba 2.2.5. Here's an example from smbclient: [mark@protagonist ~]$ smbclient //medusa/share added interface ip=192.168.0.38 bcast=192.168.0.255 nmask=255.255.255.0 Password: Domain=[STAFF] OS=[Windows 5.0] Server=[Windows 2000

Thanks all for your prompt reply. I agree with Alex. I think jitter buffer will have to drop packets if its input bitrate is higher than it is configured for. I never tried speex jitter buffer. What does it do when input bitrate is higher or lower ? As an alternate solution, I was thinking to play with directsound playrate [using SetFrequency()]. Has anyone used it to solve such problem? Any

But that's what a jitter buffer will do, eventually - drop a packet. And it's better to drop a silence packet. Cheers, .a -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Wednesday, December 21, 2005 3:39 PM To: Alex Bakaev Cc: Shantanu Gramsci; speex-dev@xiph.org Subject: RE: [Speex-dev] Bitrate problem On Wed, 2005-12-21 at 15:29 -0800,

Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> wrote: > > See the SPEEX_SET_VBR and SPEEX_SET_VBR_QUALITY options. But be aware > that unless you're aggregating many VoIP calls on the same link, there's > little use for VBR in VoIP. I don't agree with this. I've used Speex's VBR mode exclusively in my P2P VoIP software for the past 4 years and never

Jean-Marc Valin wrote: > Thorvald Natvig a ?crit : > >> I see you're changing the jitter buffer around quite a bit. Could you >> let us know when it's ready for general testing? (At the moment it >> doesn't handle missing packets at all) >> > > While I'm not completely done yet, I thought the current version was > working. Can you tell

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

-----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Thorvald Natvig Sent: Monday, March 20, 2006 12:11 PM To: speex-dev@xiph.org Subject: Re: [Speex-dev] Who is using the jitter buffer? But I must say that it really does work well at the moment, at least from the "minimizing latency" point of view :) How do you timestamp the

Hi, Is there are document where the interaction between all the configuration options of speex is explained? Speex seems to have CBR, VBR and ABR. You can also use SPEEX_SET_QUALITY (SPEEX_SET_VBR_QUALITY) and SPEEX_SET_BITRATE which I suppose can't be set at the same time. Is there a list of possible combinations somewhere? I also have another question related to jitter. To minimize jitter,

Thorvald Natvig a ?crit : > I see you're changing the jitter buffer around quite a bit. Could you > let us know when it's ready for general testing? (At the moment it > doesn't handle missing packets at all) While I'm not completely done yet, I thought the current version was working. Can you tell me what happens exactly (without output if possible) so I can fix it? Also,

I'm using CWRSync 2.6.9 protocol 29 on a Windows 2003 Standard R2 server. I'm trying to copy files onto a local usb harddrive. Both filesystems are NTFS. Rsync is failing to copy some files due to the file name length. I kind of assumed that Rsync could handle the same file name length that Windows users are allowed to enter. :-) Is there *anything* I can do with Rsync to allow these

Hi, I have a few questions about the speex_jitter_buffer when used with DTX here goes: I want to use the speex_jitter_buffer with muliple streams (multi client) and with DTX so that clients will only transmit data when they are actually saying anything, but Im unsure of how the ticking is handled. I have 1 x SpeexJitter struct per client stream Im using portaudio and Im planning on