similar to: Version 1.0.4 is out

Le jeudi 13 janvier 2005 ? 10:59 -0500, Jared Whitby a ?crit : > Interestingly enough.. I started playing around with preprocessing > options in 1.1.6 and happened upon the denoise filter > (SPEEX_PREPROCESS_SET_DENOISE). When i run the test tone using that > option it is completely filtered out and I just get (complete) > silence. When the test tone is intermixed with regular voice

Hi, Just to let you know version 1.1.6 is out. Changes are: * Improved jitter buffer (now actually works!) * Denoiser tuning * Improved echo canceller (please send feedback) * Support for Symbian OS (thanks to Conrad Parker) * Gapless playback for speexenc/speexdec * Run-time identification of Speex version with a new speex_lib_ctl() call * Moved the includes to /usr/include/speex/ Jean-Marc

Interestingly enough.. I started playing around with preprocessing options in 1.1.6 and happened upon the denoise filter (SPEEX_PREPROCESS_SET_DENOISE). When i run the test tone using that option it is completely filtered out and I just get (complete) silence. When the test tone is intermixed with regular voice I only get the voice. So while i still don't quite understand why the test tone

Sorry, the 3rd party and (AFAIK) windows builds are still 1.1.6. I have no idea what compiler was used. Of course ARM4_ASM was not used, since the option didn't exist in 1.1.6. Jean-Marc Le samedi 05 mars 2005 ? 13:04 +0100, Fabio a ?crit : > Hi all, > Are the 3rd party builds in page > http://www.speex.org/download.html > based on Speex 1.1.7 ? > > Thanks > Fabio

Thank you very much for your immediate attention! I've exported my wav files to raw files before encoding and the click is now gone. That sure beats discarding the last packet, which was the workaround "solution" I was going to implement... Best regards, David Resnick -----Original Message----- From: Jean-Marc Valin [mailto:Jean-Marc.Valin@USherbrooke.ca] Sent: Thursday, July 07,

Hi, To increase the stack size, you can increase the value in the speex_alloc() call in nb_celp.c. Let me know if it solves the problem. Also, what compiler are you using. I think some older versions of gcc may have problems with SSE intrinsics. Jean-Marc Le mardi 19 avril 2005 ? 16:02 +0200, jehan Monnier a ?crit : > Hi, > > I also tried to build speex 1.1.7 with sse enabled on a

If you have a couple minutes, it probably wouldn't be hard to see where the bug is (I just don't have time this week). It would have to do with the granulepos computation in speexenc. Jean-Marc Le jeudi 07 juillet 2005 ? 08:41 +0300, David Resnick a ?crit : > Thank you very much for your immediate attention! > > I've exported my wav files to raw files before encoding and

Le jeudi 03 mars 2005 ? 20:29 +0100, Kleijn, J.S. de a ?crit : > I use the Intel compiler 7.0 , and it doesn't like it Does it have an option to enable C99? Otherwise, if you're aware of another mechanism that allows it to allocate temporary stuff on the stack, I might be able to support it. Jean-Marc > -----Oorspronkelijk bericht----- > Van: Jean-Marc Valin

Actually, I just tried again with the wav file directly and it just seems to be a bug with the way the speexenc utility handles wav files. The problem doesn't occur with the equivalent raw file (which I used for testing). It's probably really stupid anyway. Jean-Marc Le mercredi 06 juillet 2005 ? 14:00 +0300, David Resnick a ?crit : > I am using speexenc/speexdec to convert WAV files

Hi Maitland, I think your solution is probably the best implementation you could do for SPEEX_GET_QUALITY. However, unless you have something against that, I think removing it would be better, as in many cases (e.g. VBR), the call just doesn't make sense). This would of course break the API (not the ABI), but given the fact that it was never implemented, any app that uses it is already broken

Le lundi 21 f?vrier 2005 ? 15:08 +0100, Claude Brisson a ?crit : > Why not move the manual on the wiki ? Because it's already written in LyX and can be easily converted to pdf, html or anything. However, the trouble shooting part could be on a wiki. Do you volunteer? Jean-Marc > Claude > > On Mon, 2005-02-21 at 21:35 +0800, James Oh wrote: > > Hi, > > > >

Hi everyone, I've just released Speex 1.1.9. The main change in this release is the echo canceller work sponsored by Tipic Inc (http://www.tipic.com/). It is now possible to do acoustic echo cancellation and obtain good attenuation after a short adaptation time. This has been tested at 8 kHz, but it should also work at 16 khz and above, so give it a try. There were also some fixes to the

hi, jean-marc: i think we can remove spx_sig_t *orig. but am not sure about exc2Buf. is it for extension? rgds, tk On Mon, 28 Feb 2005 12:42:38 -0500, Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> wrote: > Hi, > > I looked at the code I think there are still places where you can reduce > memory. For example, I think bufSize can be reduced to around 400 > (instead of

Well, it's an ARM7TDMI core, so basically one register operation per clock, with memory accesses taking longer. Having the memory on-chip should make memory access much less of an impact. I was afraid that you would answer the way you did: I thought about my question after I sent it, and the "LP" in CELP is what makes it a sequential process; it can't do linear prediction on a

> 16 bit ints have a range of -32768 to 32767. If you divide > -32768 by 32767.0 you end up with -1.00003051850948 which > is a bad thing. > > Try normalizing with a value of 32768.0. No. Speex expects values in the +-32767 range, not +-1.0. Just converting from int16 to float *is* the right thing to do. Jean-Marc -- Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca>

Good question. I'm coming to the conclusion that using plain UDP and "home-grown" packet construction for transmitting the speex data (with timestamp/sequence counter) and implementing jitter control on the receiver end is an adequate implementation for a VoIP application. Assuming of course that I don't care about any interoperability issues with other applications etc. I was

Hi, I tried the same options suggested in your post and the problem continues. What do you suggest? 2005/9/15, Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca>: > Hi Eduardo, > > All I can say is that the timings you have are a bit odd. What > optimizations options are you using (I suggest -O3)? Also, perhaps you > can try --enable-arm4-asm just in case. I've had the

I strongly suggest you start by reading the Speex manual (you can skip the technical parts about CELP). If you still ask questions, then post them. Jean-Marc Le mardi 14 juin 2005 ? 22:30 -0700, David Barrett a ?crit : > Ok, this is a silly question, but what does the jitter buffer do? I'm > really new to audio, so please bear with me. > > From what I gather (primarily from

Hi, A couple things you may want to check: - set sampling rate to 8 kHz (at least for now) - make sure the far end signal in the playback signal is always a bit in advance (never late) compared to the mic signal. - Set the tail length to something around 100 ms. Also, if you're using two different soundcards (as I understand) for the playback and the capture, you're *never* going to get

Le mercredi 05 janvier 2005 ? 13:37 +0800, tunkeat a ?crit : > thanks Jean for the codec. My first name's Jean-Marc BTW :-) > I am new to speech coding. Am interested to implement > speex on a TI dsp, say C54. I think I've heard someone was working on that before... > just downloaded speex v 1.1.6. > when I removed optimisation flags for win32 in libspeex.dsp, >