similar to: Microphone database

> > Different brand/model, but similar as they are both el cheapo, > entry-level headsets. I tried using them on a laptop, and I get > marginally better microphone output, even with its volume cranked all > the way up + automatic gain control enabled. > > I guess those on-board soundcards by Realtek aren't as good as a > quality microphones. I'll get a USB headset

> Actually, computing the "power spectrum" for each frame of W shows > how large an ammount of the original signal at time offset j the > echo canceller thinks should be removed from the current input frame. Careful when looking at W because of how the real and imaginary parts are packed in the array. > If you compute W*X for each j and ifft, you'll get the >

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,

Hi, I made a small error in the impulse reponse function; when doing the inverse fft in fixed point, it will overflow unless it's properly scaled. This patch uses the same scaling as that used when updating the filters, and the outputs now have the same shape and the same scale in fixed and floating point. Best regards, Thorvald -------------- next part -------------- diff -ubBwr

>> Actually, computing the "power spectrum" for each frame of W shows >> how large an ammount of the original signal at time offset j the >> echo canceller thinks should be removed from the current input frame. > > Careful when looking at W because of how the real and imaginary parts > are packed in the array. Err. Ok, as I got it, 'bin 0' has it's

Thorvald Natvig wrote: >> I'd like know about anyone using the current jitter buffer in Speex. I'm >> planning on changing it to make it more general and I'd like some >> feedback about how to make it better. Also, let me know if you're doing >> anything serious with it and want to make sure I don't break your stuff. >> >> Basically, I want

Hi there! Recently I bought a simple Headset so I don't have to use my webcam mic in Teamspeak anymore. Its a normal analog Headset, so no USB. I plugged it in my Audigy2 and it works fine. The thing is, when I'm in Teamspeak starting Wine (wincfg is enough) it seems to "lock" the Microphone input of my Audigy2. Then I only can use it inside Wine and only hear what the others

On 15 March 2008, Anne Wilson cannewilson at googlemail.com wrote: > It's not obvious from that. There is an alsa-users list at > alsa-user at lists.sourceforge.net. I had to go to them once with a problem and > found that they have detailed knowledge of a lot of sound cards. The > solution to my problem wasn't obvious either - and it was a matter of finding > the

Hi, so I'm rather new to Linux. With a little bit of fiddling I was able to install Rosetta Stone (version 3.3.5). I'm running the latest versions of Wine and Ubuntu. Everything in Rosetta Stone is working great, however, it can not detect my microphone (which is not a USB headset, it is built in). I can't think of what could be wrong, as the microphone itself works fine in Sound

Hi, A couple things you may want to check: - set sampling rate to 8 kHz (at least for now) - make sure the far end signal in the playback signal is always a bit in advance (never late) compared to the mic signal. - Set the tail length to something around 100 ms. Also, if you're using two different soundcards (as I understand) for the playback and the capture, you're *never* going to get

On Tuesday 10 May 2005 9:45 am, David Masure wrote: > Hi folks ! > > I bought two sipura 841 phones. I used to have GN Netcom headset which > I connect instead of the handset. The problem is that I don't have any > sound coming out the headset and I can't speak neither ! > ... > > Or....can anyone advise me on headset working with the sipura 841 ? I just use a

The original post in this thread was never answered, and I too was requesting help on the same subject -since as of yet, this issue remains unresolved. So, to recap, Rosetta Stone is installed and running, but can't detect my microphone (analogue, or the USB headset that comes with the software). Since Rosetta Stone is reported as fully functional under WINE, I am asking for advice, in terms

I'm looking for suggestions as to the best multimedia headsets for a call center environment. A few considerations: 1) USB headsets are preferable, because they don't require a soundcard. 2) Omnidirectional microphones are problematic, because they pick up too much background noise. Thanks, Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian

Hello, I'm having the following problem when using a headset on XP connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus motherboard: - Using any sound recorder (Windows', Audacity, XLite), the level is just too low when speaking at a conversational level, even with the microphone level pumped all the way up (line displayed totally flat in Recorder)

>> 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

I am trying to use the speex echo canceler and preprocessor with sipXtapi to develop a sip user agent. the sipXmedialib call flow graph uses 10ms frames and I am not sure what the implications are if I try to change this. The documentation seemed to indicate that a 20ms frame was recomended, but it didn't go into the consquences of using other frame sizes. We are going to use headsets

I'm having a problem getting a good clear output sidnal from Ekiga to a VOIP conference call using the Ekiga.net free conference call system. I'm told that each time I speak, my voice is clear & intelligible for about .5 - 2 seconds, but then it starts to be garbled, sounding like the sounds R2D2 makes. I've used 2 or three mic/headsets - two plug into my audio I/O sockets on my

To all: I am having an echo problem with X-Lite and iaxComm. I am using the monitor speakers and a desktop microphone. My problem is that the sound from the speakers is repeated by the microphone causing an echo that is annoying. Is this correctable? I have searched throught this archives and played with various settings but have not been able to fix this. I have purchased a couple if

On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > Alexander Chemeris wrote: > > On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > > > I've done listening tests when converting wb_male.wav to 44.1, 48 and 8khz, > > > and there aren't any obvious artifacts. I also did a 16=>16 test, and the > > > results are delayed

Hi, we are looking to roll out PBX IN A Flash at our office. The first group will be using Soft Phones (X-Lite appears to be the best and works in Windows, Apple & Linux). There are many types of USB Headsets to choose from and a fairly broad price range. Is there any USB headsets people would recommend? I'm specifically interested in acceptable audio (speaker and microphone) quality