similar to: Version 1.2beta2 is out!

Hi all, Sorry for the somewhat off-topic post, but I haven't come up with a good answer to this question yet. MDS did a port of Vorbis to the Philips TriMedia processors. Others have taken that port and extended/fixed it a bit. We provide the code to anyone who asks, but at this point we think it would be better off in some sort of public repository so that patches can be maintained,

> -----Message d'origine----- > De : Jean-Marc Valin [mailto:jean-marc.valin at usherbrooke.ca] > Envoy? : dimanche, 14. juin 2009 20:46 > ? : Stephane Lesage > Cc : speex-dev at xiph.org > Objet : Re: [Speex-dev] Resampler saturation > > Just to make sure I understand, the two patches you sent are > two different ways to fix the problem, with the only >

> 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

On 2/6/07, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > I actually recommend using the development branch (current is 1.2beta1) > instead of the (old) stable branch. The newer code is much better in > pretty much all aspects. Maybe it's time to label 1.2 as stable? Or do you still require more feedback and testing before that? -Ivo

hi, haven't got the chance to sit down and look into it. from what I read in the list, double talk detection is in the pipeline... :D. Just 2 quick questions: i) does this canceller handle wideband for speex wideband? ii) does it handle feedback or howling? Lastly, a newbie question for echo cancellation: is there a convenient test to ensure that the input and output buffers obtained from

Hi, I'm using the Speex codec on my Blackfin-based board, and plain-C performance is pretty poor. Decoder is OK (something 25 MIPS for wide-band). But Encoder is not (wide-band quality 8, complexity 1): - 162 MIPS with Analog-Devices lib - 128 MIPS with 1.2beta2 (faster but not working) I don't worry that much, as it's not the optimized version, and everything is running in SDRAM.

Hi Aymeric, > 2 questions: I count 4 :-) > * Do you "feel" that the current svn is enough stable Yes. I'm planning on releasing it as 1.2beta3 this week. Even in normal time, I try to keep svn in good shape so it is almost always better than the last release. And even when it breaks, Speex bugs tend to be very obvious and easy to reproduce (no segfaults that happen once a

I have tested the AEC by file model,but found it doesn't work!The output file I got still had echo. And I have found several version of mdf.c,could you tell me which version I should use? Now I'm using version 1.11.1.1 and under windows there have some bugs. --- Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>??: > > > haven't got the chance to sit down and look into

> -----Original Message----- > From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] > Sent: Wednesday, October 17, 2007 1:56 AM > To: St?phane Lesage > Cc: speex-dev@xiph.org > Subject: Re: [Speex-dev] Blackfin port on Visual DSP, Michael Shatz ? Salut Jean-Marc, After 1 month busy on other projects, I can finally answer you: > Some things to check. Do you

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Hello, To start with, thanks a lot for making such a great voice codec available! Having recently upgrading to speex rc1, It occurred to us that there seems to have been a regression in the quality of encoding since version beta1. We are compressing some 22khz wave files in wb mode with maximum quality / complexity in VBR, and the result was really great with speex beta1. With rc1 (or beta3),

Hi All, Many Thanks to Underwood for her excellent review of our big trouble which prevent LMS-based AEC algorithms to be used in most computer. Maybe it can be summaried as follows: 1. Different sample rate of sampling and rendering does exists in most low-cost soundcards (In my experiments over more than 20 soundcards, the differences range from 0.5Hz to more than 50Hz when sample rate is set

Dear experts, I am having now speex latest version speex-1.2 rc1. How i can compile this for trimedia support. Please help. regards vizzimani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20090113/19a33ea9/attachment.htm

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 some changes that are possible, e.g: - Improving some

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re:

Hi Chris, Thanks a lot for your help. I've been able to resolve all the errors except one. I don't have ogg.h header file, so when I try to build Speexdec.exe, I get an error d:\speex\speex-1.1.5\src\speexdec.c(44) : fatal error C1083: Cannot open include file: 'ogg/ogg.h': No such file or directory I don't have this file, but as I wrote in my previous mail, an ogg_static

What would be a good parameter to return that would better represent the loudness of the signal? As for the your other comments, I'll make the necessary changes and resubmit. Thanks, Mihai On Oct 25, 2007, at 7:45 PM, Jean-Marc Valin wrote: > I'm in favor of the idea, but not of the current implementation. There > are two problems: > 1) The st->loudness parameter

I'm developing an application that have a video conference component. For that I need echo cancellation, and is looking around for algorithms/implementations of that, and the one in speex is an alternative. In the documentation for speex I find the following sentence however. "Using a different soundcard to do the capture and plaback will *not* work, regardless of what you may

Jean-Marc, Frank, I have stumbled across a similar situation regarding optimization. I seem to have a similar setup as Frank does with a fixed 48khz in and out. The wideband mode and ultra-wideband modes are really what I?m looking for. I have a test application that reads audio, downsample to 16kHz (or 32kHz), speex encode, speex decode, upsample back to 48kHz, and playback. If I remove