similar to: mdf no sound issue

Jean Marc, Ok so I tested with the new code, same result- frame size= 160, filter length=160 ms/1280 samples, 8000 Hz diverged after 9564 calls/packets or 191 seconds. My system stays relatively in synch but is not perfect- difficult to measure if there is any clock drift, but it probably is. In order to make it break faster I use an open air usb microphone that is part of a logitech notebook

Jean Marc, I have been fiddling around with preprocess to try to get it sounding less metallic/underwater- so based on some things said recently, I have been first fiddling with mean_post and mean_prior to try to get them to do change the update rate, so that update_noise gets called at the right times. Haven't made it too deep into the code, but I have a few questions in speex_preprocess:

Not a big deal, but the "<http://xiph.org/svn.html>Subversion Access" link on http://www.speex.org/download.html page should probably point to: http://www.xiph.org/svn/ rather than: http://xiph.org/svn.html Tom ______________________________________________ Tom Harper Lead Software Engineer SightSpeed - <http://www.sightspeed.com/>http://www.sightspeed.com/ 918 Parker

Congrats Jean Marc, Just heard you got a new google assistant for the Ghost project! Tom ______________________________________________ Tom Harper Lead Software Engineer SightSpeed - <http://www.sightspeed.com/>http://www.sightspeed.com/ 918 Parker St, Suite A14 Berkeley, CA 94710 Email: tharper@sightspeed.com Phone: 510-665-2920 Fax: 510-649-9569 My SightSpeed Video Link:

Jean-Marc, So I saw in the latest code that the vad in the preprocessor is gone/going to be re-written. Is there a plan as far as this goes? Just wondering as the current one seems to work pretty well. Thanks! Tom ______________________________________________ Tom Harper Lead Software Engineer SightSpeed - <http://www.sightspeed.com/>http://www.sightspeed.com/ 918 Parker St, Suite A14

re: At 03:22 PM 10/25/2005, Jean-Marc Valin wrote: >Are you really sure you didn't have some corruption elsewhere? Totally possible- this is the first time this has happened that I know of in many many hours of usage- On the other hand, this null check isn't in my code base and it was in nb_decode_lost, and nb_encode- so I figured it was just an oversight- Tom

Hi, So I got a crash on the following code: k1=SUBMODE(lpc_enh_k1); k2=SUBMODE(lpc_enh_k2); which in the newer codebase is: bw_lpc(SUBMODE(lpc_enh_k1), st->interp_qlpc, awk1, st->lpcSize); bw_lpc(SUBMODE(lpc_enh_k2), st->interp_qlpc, awk2, st->lpcSize); I am not sure if the newer code will have the same issue but the following check is

That's definitely strange and I've never encountered that. Normally, the only way for _mm_load_ups to generate a segfault is for the input to be invalid memory, in which case the C version should crash too. I suspect the compiler (or something else) may be hiding the real problem. Can you get a debugger and see exactly what assembly statement is causing the crash and what the operands are?

Jean Marc, So the C99 array syntax is breaking some lesser compilers- is there a known max size for bark[]? Fiddling with latest svn again... Thanks! Tom ______________________________________________ Tom Harper Lead Software Engineer SightSpeed - <http://www.sightspeed.com/>http://www.sightspeed.com/ 918 Parker St, Suite A14 Berkeley, CA 94710 Email: tharper@sightspeed.com Phone:

There is a port to mingw, which is what many folks use now: http://repo.or.cz/w/git/mingw.git I haven't had a chance to fiddle with it yet so ymmv. Tom At 06:00 AM 6/9/2007, Jean-Marc Valin wrote: >Hi everyone, > >I'm currently attempting to move Speex development over to Git. Because >Git is distributed, I means that anyone would be able to do development >and commits

Hi Jean Marc and List, So I have started finally fiddling around with Ultra-wideband mode. It appears to be very similar in operation to Wide mode, except that when peering into the packet structure it looks like (and these are kind of questions as much as statements here): 1. update rate 0 is not used in UWB- only 1-4? 2. The total bits used for each UWB update rate seem to be as follows:

Bjoern, As far as I understand it, AGC keeps the volume level of the speech elements (i.e. gain) stable no matter how quiet or loud the input becomes- Tom At 10:53 AM 3/28/2004, Bjoern Rasmussen wrote: >Hello > >The st->zeta pointer isn't freed in the speex_preprocess_state_destroy() >function of the preprocess.c file (alloced in line 167). It's in Speex >1.1.4 by

At 12:35 PM 10/6/2004, Matthias Granberry wrote: >There is some SSE assembly >language, but it's in GCC/AT&T syntax rather than the windows-standard >Intel syntax, so you might have to do some of your own translation to >something your compiler understands. We submitted an intel patch a ways back- maybe it is in the archives somewhere... Tom -- Tom Harper -

Jean Marc & list, Following up on this- there seems to be an issue with this calculation in mdf.c (Smooth echo energy estimate over time) Pey += Eh*Yh; Pyy += Yh*Yh; it goes too large for floating point, which seems to be caused by extremely large values in st->Rf[j], st->Yf[j] and smaller values in st->Eh[j], st->Yh[j]- things were relatively quiet

? is there any speex code in place that may help enable variable speed playback ? or perhaps another open source project has some code for this ? variable speed playback for voice messaging is an attractive feature, that would be well recieved for sure ! thx for any reply, Mark Tom Harper <tharper@sightspeed.com> wrote: At 12:35 PM 10/6/2004, Matthias Granberry wrote: >There is

Jean Marc, Thanks for the advice. The estimate update may come in handy. I have had some time to play with the adaptation time. I tried making the adaptation time 80, 160 and 320. It seemed like the smaller value was actually the best in my initial testing, but I need to test this against a more noisy setup than I have here. I am guessing that for vad a shorter adaptation time is probably

Hi Jean Marc & List, So I have been fiddling with the denoiser (again). While poking around I noticed that nb_preprocess is basically a counter that is mod-ded with 100 (the default), which causes Smin[] to be re- seeded with the value in Stmp[] (min of the previous adaptation period). Smin[] is then used to update the noise probability, which is (probably) less likely when adaptation

Hi Folks, Attached is a patch against the latest svn, plus new source files. This patch allows the specification of c or assembler versions of various functions at run time if _USE_SSE or _USE_ALTIVEC is specified at compile time. The basic concept is to use function pointers and preprocessor trickery to allow for run-time without changing how the other platforms work, esp. the platform function

Jean Marc & list, I was going to start fiddling with using the ability to pass NULL to speex decode and using that to return an interpolated frame. I am curious what the intended use of this feature was- I.e. if there is a packet loss situation is it preferable to keep interpolating until new data shows up, or to cut your losses after a certain number of frames? I suppose some of this

Jean Marc, So assuming the quality setting was set as high as possible, and the audio stream was 8000 hz, 16 bit mono, I have two follow up questions: 1. Does the complexity setting override the quality setting, or does increasing the complexity only lower the rate at a given quality? Or would setting both higher increase the quality more? 2. What sort of impact would setting quality as high as