similar to: Broken denoiser in SVN (?)

You have the point. "I get very distorted speech when using libspeex in my application. What¡¯s wrong? .... If the input speech has an amplitude close to ¡¾2^15, it is possible that at decoding, the amplitude be a bit higher than that, causing clipping when saving as 16-bit PCM." This is one of FAQs in 'speex-manual' you wrote. I guess amplitudes close to 2^15 bother me. Do

Hi, It is better to replace this line in function filterbank_new: max_mel = toBARK(EXTRACT16(MULT16_16_Q15(QCONST16(.5f,15),sampling))); to max_mel = toBARK(EXTRACT16(sampling/2)); It gives the same but it seems to be faster and avoids overflow on 44100 kHz that prevents denoiser to process 44100 streams. (Yes I know that Speex should not pack 44100 streams but it does now and I use it). Best

https://bugs.freedesktop.org/show_bug.cgi?id=105097 Bug ID: 105097 Summary: Computer hangs only mouse moves Product: xorg Version: unspecified Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau Assignee: nouveau at

Hello, I have been working with the denoiser in the Speex library and have a few questions. Some information before I get started: Speex version: 1.1.9 (as near as I could tell nothing new in the denoiser in cvs) Audio input: 32bit float - 48000 sample rate (the input starts from jackd) The first thing I am finding is that I can understand the audio in

Hi, I had the same experience. The VAD done in the encoder does not seem to work very well at all. However, the VAD done by speex_preprocess() works much better. You will also get better results if you enable the denoiser, I think. peex_preprocess_ctl(preprocess, SPEEX_PREPROCESS_SET_DENOISE, &denoise_enabled); speex_preprocess_ctl(preprocess, SPEEX_PREPROCESS_SET_VAD,

Hi all, I'm developing a VoIP application for my IPAQ 5450 and from some moths i'm using speex as codec. Now is the time to obtain the best possible audio quality. As my PocketPC is not a smartphone my main problems are the echo cancellation and the noise. As the first one can be solved by using headphones how can i solve the second? Do someone have some hint for me regarding the

Hello, The denoiser was good with a car noise or a raod noise before the r11739. But for some situation, the actual version is perfect. That's why I think that if we can adjust the agressivity of the denoiser, we can respond to all of the situations. Maybe two parameters should be enough : high or low. For the AEC, I have to test with older versions because it's another person that

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

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 the way. I'm trying to make the denoiser work with my application and has got reasonable noise reduction after applying the denoiser. I, however, haven't been able to find any information of what the purpose of

Jean-Marc Valin wrote: >If you set the denoiser to "on" and the VAD to "off", what difference >does it make in CPU time? > <p>Same program, running on Athlon XP 1700+: Test 1, using VAD, but AGC, denoise off: tevek@canarsie:~/work/hms/app_conference $ time ./vad_test /tmp/demo-instruct.sw 5 reading from /tmp/demo-instruct.sw, repeating 5 times read 537760

Hello, Is it possible to adjust the level of the denoiser ? In an old beta, before you change the aggressiveness of the denoiser, it works very well with the noise of a car, a road etc... but not now. The AEC too works very well in a old beta, but now, I think there is a problem... I have tested it with the same sound card (WB 16000). In any case, your project is the best VOIP library in this

Hello, With high microphone gain, I seem to have problem making the silence detection work. The speech detection works well for the rare dish sample, which has very low noise amplitudes in silence regions. However, if the microphone gain is set to really high, noise samples are taken as speech, as indicated by the non-zero return value from speex_encode() calls. I had VAD turned on.

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

Hi Tom, Of course the current tuning of the denoise is approximative and could probably be improved... That being said, the modulo 100 isn't the adaptation time itself, but rather the window over which to look for minima. The main idea is this: if the power in a certain bin isn't too much higher than the minimal value for a certain window, then we can adapt the noise estimate. The

OK, so the problem doesn't seem to be the VAD specifically. Can you tell me how much audio you had in the test? It may be that nothing's wrong and the code just isn't so fast that you can do 100 channels. Or maybe it just needs a bit of optimization... Jean-Marc Le mer 31/03/2004 à 10:03, Steve Kann a écrit : > Jean-Marc Valin wrote: > > >If you set the denoiser

Ouss a ?crit : > Hello, > > Is it possible to adjust the level of the denoiser ? Not for now. > In an old beta, before you change the aggressiveness of the denoiser, it > works very well with the noise of a car, a road etc... but not now. When did that change? > The AEC too works very well in a old beta, but now, I think there is a > problem... When did that change?

> 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 the way. Oops... Thanks for letting me know. I'll change that for the next release (in the mean time, the fix is obvious). In case you're interested, I'm currently working on a reverberation suppression

> Reverberation suppression? Basically, it means that if you are in a room with lots of echo (long decay), I can reduce it a bit. > I guess this would help reduce local source echoes? I've never > _noticed_ that to be a problem in my use, but I would imagine that > using a notebook's built-in microphone, you'd get some echo off of the > screen and stuff [also from

> Hmm, How can I find that out? How much CPU would you expect it to > take? I don't know. It's been a while since I last played with that code, but I'd expect it to take less time. > I've been playing with oprofile, but I don't see it getting that > finely grained.. Can you make sure the time is spent in the VAD and not in the encoder or decoder (at the other

> > These numbers sound like a problem I has a while ago with the decoder. > > The VAD shouldn't take much CPU so I suspect there might be floating > > point underflows in some part, slowing down the Intel CPUs a lot (for > > some reason, the AMD CPUs seem to handle underflows faster). > > > > Hmm, How can I find that out? How much CPU would you expect