similar to: using the interpolation feature

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 -

? 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

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

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

Steve, The main problem I am having with the system is clipping off the start of someone's speech when they first start talking- the ends of the sentences seem to be handled properly. I am wondering whether this is the fault of the audio playback system or whether this is a speex issue- I also get the musical artifacts problem with the denoiser. This seems to be more of a problem on open

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

Hi Jean Marc, I think there is just a confusion over terminology going on here- I agree that support for 3dnow base version may not necessarily be relevant; However, even though 3dNow extended is a bastardized version of SSE, it still supports the same instructions, and that is what is important- I don't think we intend to add any AMD specfic code. The real issue is cross CPU SSE support,

Jean Marc, I ran across the silence issue that results from the echo canceller producing "bad" floating point data. Unfortunately this hasn't been done in a way i can reproduce, but I found that calling reset didn't fix the silence, unless I also reset: st->PHI[i] st->Eh[i] st->Yh[i] So it may be good to add these to the reset function. I was using these in

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

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

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:

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: