similar to: reduction of noise due to high microphone gain

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.

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

I'm not sure you'd have much success in trying to make a generic profile to fit the "typical noisy PC" since PC configurations vary so much. Consider an integrated AC97 codec vs. the mic input on an SB Live!/Audigy Live!Drive, or whether or not the input from the CD-ROM audio is muted, or what noisy devices might be situated near the mic. It's nice to be able to get

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:

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

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

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

> 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

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

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

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

I tried feeding in the 3 encoded frame in ONE BLOCK, and calling speex_decode() 3 times in a roll. Only the 1st frames came out perfectly. For the other 2, I got "corrupt" frame warning. I was supposed to get 38 bytes consumed each frame (narrow-band, VBR off). I tried speex_bits_remaining() to peek on the # of bits consumed, and got variable (clearly wrong)#s returned. But if I

Hi Richard, Richard McGravie a ?crit : > We are looking for a consultant to assistance in the following areas. > 1. Shim in the pre filter from speex to all trunk interfaces in > asterisk. (E1, ISDN and FXO) > > 2. Optimize and improve the pre-filter for airplane noise floor. > Is this something you would be willing to assist us with or do you have > any contacts that could

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()

I just finished testing and integrating Speex into our test project. Speex works excellent and sounds great for the bitrate. I have noticed, however, on cheaper microphones that Speex will produce loud static noise at high harmonics/amplitude in the sound. Is there some setting to prevent this static? I don't have this problem with various other manufacturer voice coders that I am using

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

Here's the situation: 3 frames are being transferred back to back over RTP. The total payload is known (thus the TOTAL frame size). We'd like to decode all 3. According to IETF RTP Profile (appendix C, of Jean-Marc's Speex Codec Manual, verison 1.0), "it's only necessary to consider the sequence of bits as output by the Speex encoder, and present the same sequence to

Hi, I am considering to use Speex in a particular environment and I want to ask if you thought it realistic. I am going to use Wireless SIP (G.711) handsets in an environment that is typcially noisey (background), connecting to an Asterisk PBX. I am considering converting the G.711 to Speex so I can "massage" out the background noise. Is this realistic? Any input would be great.

> I would like to test the echo cancel algorithm available in unstable version > 1.1.4. This echo canceller can be used with other codecs like G711? If by "used with other codecs", you mean that you can use the result as input to any codec you like, the answer is yes. > Somebody could send me some documentation or sample, or explain the next > functions: > >