similar to: Preprocessor denoise. Does it work?

There are many ways to implement a VAD. What you described is actually perfectly equivalent to the most trivial (and least robust) VAD algorithm. Jean-Marc Andy Ross wrote: > Ton Grandgent wrote: >> Andy Ross wrotte: >>> I wrote a trivial squelch feature* in 10 minutes that works >>> basically 100% of the time. >> Could you please explain how this differs from

Jean-Marc Valin wrote: > Andy Ross wrote: > > Not knowing how VAD works, I can't say for sure. > > There are many ways to implement a VAD. I meant "not knowing how speex's VAD works", of course, not VAD in general. If you would stop interpreting everything I say in the least charitable manner, this might be going more smoothly than it is. (Tom was right, by the

Jean-Marc Valin wrote: > Andy Ross wrote: > > Uh, production applications almost always require squelch, no? > > Some do, some don't. In general, distinguishing between a keyboard > and a speech transient is next to impossible based only on a few ms > of speech. That is true for distinguishing it by waveform, but not by amplitude. As I mentioned, these transients are

Andy Ross <andy@plausible.org> wrote: > > I wrote a trivial squelch feature* in 10 minutes that works > basically 100% of the time. > > * Zero the sample data if the maximum sample in a frame is less than > 4% of saturation or 20% of the maximum sample yet seen. It's about > 8 lines of code. Could you please explain how this differs from VAD? Tom

I'm having trouble with the preprocessor's noise reduction feature. The basic issue is that it simply doesn't work very well. With my laptop (whose microphone is otherwise quite capable) I routinely hear transient background noise, typing, and other "quiet" sounds leaking through to the speex stream. Even worse, the AGC feature is blowing these things up into just awful

> That is true for distinguishing it by waveform, but not by amplitude. > As I mentioned, these transients are objectively tiny. *Your* transients may be "tiny" and in any case, it doesn't help if you don't know the level you're recording at. I guess I'd be > curious as to which voice codec applications require no squelch (other > than trivial examples

Andy Ross a ?crit : > I'm having trouble with the preprocessor's noise reduction feature. > The basic issue is that it simply doesn't work very well. > > With my laptop (whose microphone is otherwise quite capable) I > routinely hear transient background noise, typing, and other "quiet" > sounds leaking through to the speex stream. Even worse, the AGC >

Jean-Marc Valin wrote: > The noise suppressor will only attempt to remove stationary noise, > such as thermal noise, fans, ... The AGC can indeed do strange > things in these cases, but it's been improved in svn (compared to > 1.2beta1). OK, then the problem is that I misunderstood the feature. I assumed that dynamic squelch was part of it, but it's really something more

> OK, then the problem is that I misunderstood the feature. I assumed > that dynamic squelch was part of it, but it's really something more > along the lines of active noise cancellation. That's fine, I'll work > on improving my own squelch code. No. Active noise cancellation is yet another thing, where you cancel the noise in the "acoustic world" by

I implemented the calcPower(). It works perfectly. The example is given you in just about 6 hours. Cant paste the whole source here and need to meet someone now. Thanks all (particulary tom). I try to figure out whitch problem exists with the #define SPEEX_PREPROCESS_SET_PROB_START 14 theese days ----- Original Message ----- From: "¼Õ½Â¿ø" <ssw0725@ncsoft.net> To: "Tom

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

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

Jesus, Unfortunately, FFT and magic algorithms don't work (yet?). You might want to try this if you're not satisfied with Speex VAD: http://lists.xiph.org/pipermail/speex-dev/2008-August/006860.html It won't perform any miracles, but I think it works pretty well and is easy to tweak. Tom >---- Original Message ---- >From: jmorion at toomeeting.com >To: speex-dev at

Hello, in my project im using speex 1.2rc1 and the preprocessor VAD seems to only separate complete silence from not complete silence frames. The Speex Manual, you can read "The voice activity detector (VAD) provided by the preprocessor is more advanced than the one directly provided in the codec." but if you go to the source code in preprocess.c line 995 "/* FIXME: This VAD

agc_gain seemed to fit with the idea of what I wanted to do, it was easy to understand its units and behavior, and freezing it produced the desired results. Also I wanted to cap it, so that's done at the same place, and that definitely works. All I want to do is be able to freeze AGC adaptation and put an upper bound on the AGC (for example, 2x amplification). Both of these things seem

As I understand it, there are two separate ways to get VAD information from Speex: 1) Using the encoder. 2) Using speex_preprocess(). I present the following observations from an application developer's perspective. They may be wrong, in which case I would appreciate corrections. - The two VAD systems are implemented differently. - speex_preprocess()'s VAD provides more accurate

hi there, I am new to mailing list so excuse me if I don't obey to the 'netiquette'. i am writing voice chat and speex is in the root of it. i write it in Java and use JNI to link with 'C'-based Speex 1.2beta. [I know of JSpeex but there are not implemented some features] recently i decided to use DTX feature of speex as well. the code follows. The problem is that no matter

I have been tyring to use the preprocessor standalone by feeding it pcm data from a headerless pcm file (mono, 22050hz) and then write the result to another file. Unfortunately, all I get back is noise (first frame is set to 0, rest is noise). Frame size is set to 440. I just modified testdenoise.c slightly to read/write from a file (see below) but I can't get useful results. What am I

Jean-Marc, Ok, I understand. It's floating point but I don't have a mechanism in place yet to capture the source data. I'll do that, reproduce the problem, and put together a minimal sample program that can reproduce it given a data file. Thanks, Tom Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> wrote: > > Hi Tom, > > In order to trace the problem,

Hi...Tom, How to use the code you written? Can you show me some example? Thanks, -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On = Behalf Of Tom Grandgent Sent: Friday, March 03, 2006 12:57 AM To: Steve Kann; Lis Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Voice Activation Level (speex 1.1.11.1) Lis, I suggest you try tweaking Speex's