similar to: Speex and bandwidth usage on Asterisk's IAX

Hi, I've just joined the list. Thanks Jean-Marc and other contributors for your efforts with Speex. Anyway - I've been hired by a local firm to investigate, write-code-as-required and generally help them get the maximum bang-for-the-byte for VOIP traffic on the hyperexpensive Internet bandwidth going out of South Africa. I started the process by doing some methodical measurements and

Alex, It's purely empirical, but I've found that values in the range of 0.001 to 0.002 tend to offer a nice tradeoff between rejecting unwanted sounds and picking up all legitimate speech. I use a default of 0.0015 in my program and it seems to work well. For bounds of the GUI slider, I use 0.00005 and 0.01. Sorry, but I don't know how these values translate to dB or any other

I wish Speex could come with instructions on how to build libspeex_armce.lib. ________________________________ From: speex-dev-bounces@xiph.org on behalf of speex-dev-request@xiph.org Sent: Mon 11/13/2006 3:01 PM To: speex-dev@xiph.org Subject: Speex-dev Digest, Vol 30, Issue 15 Send Speex-dev mailing list submissions to speex-dev@xiph.org To subscribe or unsubscribe via the World

>That's a totally different topic. I do intend to reduce the wb memory >usage, just like I did with the narrowband for 1.2beta1. Still, don't >know where you take this 160k Word32 number (640 kB). I don't think >wideband requires anywhere near that amount of memory. Sorry, it's 160kB. What do you think? and any suggestions for memory reduction? Lianghu On 11/16/06,

> Another issue is the memory allocations distributed so many places that > it's hard to provide a single memory initial function interface. > > In a VoIP case on ARM, the total memory size for speex codec should be > known at the inital stage since all the memories are allocated > at the initial stage. If you want everything in the same big block, all you need to do is

It would be nice to merge the Run-Time SSE/Altivec patch that we have been maintaining. Aron Rosenberg SightSpeed Inc. -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Jean-Marc Valin Sent: Monday, November 13, 2006 1:32 AM To: speex-dev Subject: [Speex-dev] Quick survey for Speex 1.2 Hi everyone, As you may have guess, Speex 1.2 is

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,

Hi All, Another issue is the memory allocations distributed so many places that it's hard to provide a single memory initial function interface. In a VoIP case on ARM, the total memory size for speex codec should be known at the inital stage since all the memories are allocated at the initial stage. In my current implementation, all the memory allocations are collected together to form

Jean-Marc Valin wrote: > Hi everyone, > > As you may have guess, Speex 1.2 is slowly approaching, though there's > still a lot left to do so I can't say how long it'll take. I thought > this was the right time to ask if there's anything missing or that can > be improved to make 1.2 better. At this point, it can't be anything > major, but there are still

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

Hi Jean-Marc, I finally make it work! Thanks for telling me that the algo will not work for 8-bit PCM. In fact I've tried using 16 bit PCM last time, but apparently there's a bug. The bug lies with the RecBuffer. The RecBuffer must be set to 320bytes instead of 160 bytes since every sample is now 16 bit. The compression is amazing, from 320 bytes to 38bytes! Thanks a million, Carine

> I've been watching the speex development from its inception because I > and several Tech Startup Connection members have a very important > application for this voice encoding/decoding. Further, we are quite > familiar with ACELP as implemented in G.729. As far as I know Speex > is also ACELP ... yes/no? No. Speex is CELP (Code-excited linear prediction), but not ACELP

Hi, I try to use Speex codec into Win32 platforms. However, I find the CPU bandwidth usage is very heavy on a Pentium 3 machine. Compare to Microsoft's G723.1 codec, speex 8k is using more than 20% cpu bandwidth. Does anyone know what is the best version of Speex to "beat" the Microsoft's G723.1's on CPU bandwidth usage? Does Speex have MMX-enabled codes? thanks very

B. Mitchell Loebel a ?crit : > Thank you. You're right ... my error ... I meant to say 12 bytes > (including the 2 bytes for VAD). And it is 10ms/frame. No matter ... > thank you for the SPEEX specs. In terms of quality, what SPEEX bit rate > compares with G.729 at 8kbps data rate please? Haven't done formal testing and it depends on whether it's G.729 or G.729A. I'd

Thanks a lot for your reply, Jean. >I'm not aware of anyone packaging Speex especially for a Palm device, >but I have been able to use it as a command-line executable (you need >version 1.1.x to do that). Your reply essentially means that there aren't any commercial speex plug-ins for Palm available, so the only option I do have is to write one. Actually I was thinking of that

Fabio Pietrosanti (naif) a ?crit : > However, AMR it's not royalty free and if i need to use it on Windows > Mobile i will need to pay a quite expensive royalty for each peace of > software distributed (not sold, distributed!!!). This is exactly why I wrote Speex! > So, i am evaluating other codes and the only real alternative capable of > running on a 9.6kb/s link seems to

Hi all, I'm new to the speex development mailing list, so I don't have much experience/idea about Speex. I came to know about speex when I was working on a R&D for my application on Palm devices. I've created a Pocket PC application, which is a talking phrasebook using .NET Compact Framework. In this application I'm using a 3rd party MP3 plug-in for playing my MP3 files,

Hi! I?ve just discovered Speex, and sorry if this isn?t the right place to ask this, but I found no other mailing list (some projects have a "users" and a "developers" list separately, I guess this isn?t the case with Speex. If it is, please let me know). First, how I got here: 1. I?m looking for some low-bitrate codec to reduce the size of some very long (think 4-5+ hrs of

Thank you Jean-Marc. My understanding is that G.729 is a telephone codec, so there must have been some reason why its developers went to 10ms/frame. Do you know why that might be? From a recent post on this list I saw somebody talking about your decoded sample rate being 8KHZ/sec. and then he mentioned that being 160 bytes at 20ms/frame. That said, I take it that your decoded samples are

Kemal, It sounds like you are doing something wrong. I strongly recommend that you profile your application to see exactly where the CPU time is being spent. AMD happens to have a nice profiler called CodeAnalyst that they give away for free. And it's plenty usable on Intel CPUs as well. http://www.amd.com/us-en/Processors/DevelopWithAMD/0,,30_2252_3604,00.html Make sure you test a