similar to: speex_denoise on non-microphone noise (static ?)

Hi, I'm not sure how speex_denoise() works, but my solution to this problem is fairly straightforward and works well in practice (in a simple VoIP program.) Note that I am also an amateur and am making up some terminology here... but I've tested my approach on several noisy environments with low-quality mics and it does work well. Problem: There's usually annoying noise present

That's might be speex_denoise() is trying to do already, based on the fact that if noise is present in speech signal, the former gets suppressed to a certain extend. Also, I noticed after suppression, the short, noisy attack - speex_denoise() have to be doing some kind of profiling. I've looked at a few offline denoise() solutions. They all involve profile building. A real-time

Understand that it's a *guess*. If this fixes or at least betters the situation, you're going to need to find a legit way to insulate these cards, or to switch cards. If it's for customers, I should hope you wouldn't be using enamelled aluminum foil. :D -----Original Message----- From: Tongbiao Li [mailto:tli@viack.com] Sent: Thursday, September 18, 2003 11:20 AM To:

Take what I say with a grain of salt: I'm an amateur and haven't actually touched Speex in any way, yet. I'm just sort of passing on personal belief from personal experience. Also, check and make sure that the microphone line is insulated. There are a number of problems with sound cards picking up interference from the host machine. The wires that run between ICs on a card

Thanks for the speedy response and detailed, enlightening explanation. Now I understand where the problem is, and will try out your suggestions just to further confirm my conjecture. When I am done, I have to take the foil out, though. This is a product for our customers to use, and although we've got budget for mulffing every sound card we developers use, most likely the company

The problem started with speech detection. Speech sections are detected well. However, once in a while non-speech sections are also marked as speech. The root was finally traced down to microphone static noise. Then I pulled the microphone out. Our system still records noise. To isolate the problem, I wrote a small app just to open the device and record raw samples, calls speex_denoise()

> Then I pulled the microphone out. Our system still records noise. To > isolate the problem, I wrote a small app just to open the device and > record raw samples, calls speex_denoise() and outputs both sample > sets. > The noise is still there, with level fluctuating with gain level, > unless > "All mute" is chosen. > In the case when NO microphone is

Hi, There seems to be some confusion as to how speex_denoise (now called speex_preprocess) works. Unlike what some here assume, the noise estimation is not done offline or while prompting the user. The system automatically adapts the noise estimate when it detects that the user is not talking. This also means that it is able to adapt to changing noise environments - as long as the noise is not

This works really well for white noise reduction. However what I've noticed was the amplitudes of normal speech samples also get reduced. Is this something by design, or is there a way to automatically recover the original speech sample volumes ? <p>Thanks. <p>Tongbiao <p>-----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@hermes.usherb.ca] Sent:

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.

These are my results with the Speex AGC and denoiser: Over the past couple of months, I've spent probably 10-20 hours of conversation with Speex (wideband) doing AGC + denoiser on both ends of the call. I think it does an excellent job of removing most noise while keeping speech intact. Yes, it's possible to make some constant noise (like humming) and have that get removed, but

Le dim 31/08/2003 à 20:12, Daniel Vogel a écrit : > > This works really well for white noise reduction. However > > what I've noticed was the amplitudes of normal speech samples > > also get reduced. > > Noticed this as well recently. This is probably due to the AGC (Adaptive Gain Control) that's integrated with the denoiser. I'll try adding an option to

Hello everyone. I have a problem with a high level of noise after opus encoding/decoding procedure. If I stream a raw data from the microphone and send it back to the speakers I hear the clear sound (with a small delay). If I encode each frame, decode it and send it back to the speakers I hear a very noisy signal - it is so large, so it is almost impossible to distinguish words in my speech...

Hi all again, Thanks for the response to my previous question, Jean-Marc. It was of great help. Now another problem I encountered: How do I detect that comfort noise is being sent out (I have VAD, DTX on)? I could not find an API function for it, so tried frame rate. For nb, it gives 2950-3950 for silence, however it seems to also give low number sometimes even for speech periods. I

Hello, We currently have code for sampling at 8 KHz at device layer. In order to have minimal code change while still doing wideband I am thinking of packing two 20 ms frames of the low band data as input to Wideband mode for encoding. Is this a feasible idea? I actually tried it out. Speech came out at the decoding end, but I could occasionally hear low pitched, high frequency

>> Generate a test signal (10+x sine waves per frame), where x increases by >> one for each iteration, and wraps around at 100. > > Testing with sine waves is usually not a good idea. If you intend on > cancelling speech, then test with speech. Ok, I tested more extensively with both music and two-way speech. More on this below. >> However, when peeking at the

Actually, if you have a second microphone that captures just the noise (perhaps a directional mic pointed away from the announcer), and the sound path between that microphone and the primary microphone is linear, then JM has already done the work for you. Just use the AEC and plug the "noise" microphone into where you would normally plug in the speaker on the AEC (and delay the

Lis, I suggest you try tweaking Speex's VAD probabilities as Steve suggested. But consider a simple threshold-based approach as a backup option. Personally, I struggled with Speex's VAD algorithms (both encoder and preprocessor) for a long time, tweaked the probabilities, wrote special case code to work around the mistakes, and was still never satisfied with the results. In times

Hi, I've changed the m_waveFormatEx.wBitsPerSample for both record buffer and playSound buffer, changed my array from char to short and I'm still getting the same constant background noise. I notice that when I talk into the mike, I will get a slightly louder noise, can't hear any clear speech... Any idea what's wrong? Regards, Carine ----- Original Message ---- From: John Miles

I done it speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB_START, &g.s.VADstart); speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB_CONTINUE, &g.s.VADcontin); /*char *lisDebugCh = (char*) malloc(20); int decimalFcvt, signFcvt; static int firstDebug = 1; lisDebugCh = gcvt(g.s.VADstart, 20, lisDebugCh); if(firstDebug == 1) { ::MessageBoxA(NULL,