search for: hissing

Hi, I have got a hissing/humming noise on my handset that is connected to the S100U USB device.. I was running a CVS version of * that was about 2 to 3 weeks old and had perfect quality.. But that HDD died on me so I reinstalled and used the latest CVS, now I have this sound that is best describes as a similar sound to th...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi to everybody. First of all I hope this is the right place to discuss such an (nitpicky) issue. I've just been testing the current Opus release and for mere curiosity compared its performance to WMAPro with CD quality music at low bitrates (48 kbit/s). While Opus generally does a very good job, I found one particular example (a high pitched

I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an HP ML110 G6 using Ubuntu Linux 10.04 LTS. I have two Digium TE121 single T1 port cards and a Digium AEX800 8-port FXS card. All PCI Express cards. Co-workers are hearing hissing sounds on some calls, and I am getting IRQ errors when running "dahdi show status". I see that sharing IRQs for Digium cards isn't recommended, so I'm trying to set it so each card gets its own. From the few web sites I've read so far, including Digium's FAQ site, I'...

I'm running a X100P connected to a POTS line and a TDMP400P w/ two FXS daughter cards. Both calling out from one of the FXS phones (internally) or calling my home number (externally) the FXO card starts to freak out. By freak out I mean I can still hear but it sounds like you are underwater, there is an annoying hiss or buzz on the line as well. If I hang up and pick up another house phone

On 7/18/05, Christian Leitold <linux@leitold.info> wrote: > I've just found out the problem also occurs when using darkice instead > of ices2. Therefore, I guess that icecast2 is actually responsible for > this strange behaviour. This couldn't possibly be icecast's fault - icecast itself doesn't, and can't, do any processing of audio. Given that it happens with

Hello list! I've recently installed icecast2 and ices2 on my Debian Sarge box, to stream some musik for my local network. Basically, it just runs fine, using my soundcard / ALSA as audio source. Unfortunately, there is one BIG problem: after a certain amount of time (varies, can be a few minutes up to an hour or so), ices2 just "crashes", thus producing a terrible noise on the

Hi all, I started testing the RC1 encoder at ftp://sjeng.sourceforge.net/pub/sjeng/oggdrop.exe (based on branch_monty_20010708) On the songs I have tested so far (not much :) I did not hear any stereo issues, but there are some very noticeable problems with the produced files. Songs without much high-end will suddenly have one when encoded. (you'd expect it the other way around) It

2 things, first an idea... next a question. QUALITY MEASUREMENT IDEA: I find it difficult to hear 2 voice samples and tell which is nearer the original, especially if the background hiss is slightly different. So what if you actually subtract the post-compression sound from the original and then listen to the DIFFERENCE. If you can't hear any voice except background noise and some hiss from

I was doing some testing with RC2 and I noticed that RC2 doesn't encode past 19kHz with this clip (-b256 and -b350). There are no problems with this clip like it was before, but this clip contains signal past 19kHz which is audible as a faint high-frequency hiss - and that hiss is gone in the encoded file since RC2 cuts off at 19kHz. I think that -b256 and -b350 should encode at least up to

Using narrow, wideband, and ultra-wideband encoding on a short 16khz wav gave .spx's of 3,789 ... 2,935 ... and 1,875 bytes. Even after reading the manual, smaller files for the higher frequency encoding seems counter-intuitive. My mp3 at 32 kbps on the original 22khz wav is 3,866 with a quality comparable to speex wideband on the converted 16khz wav, so speex is a 24% improvement in size.

Is anyone using icecast and a client in any sort of reliable setting, such as a backup studio to transmitter link for a radio station? Something the requires reliability and good audio quality? What client are you using? I ask because I haven't found any good clients that meet my needs. Here are my requirements: 1. It should run in linux, for security and stability. 2. It should be

Hi, I have speex running as a part of a voice conferencing app. Well, one under development anyway. I'm running VBR at quality 3 and get a "hissy-squelchy" background noise. This is fine, kinda, because the internal microphone in the laptop picks up hiss, the sound of the (actually very quiet) hard drive and generally speaking is of less than exemplary quality. To help

I think you misunderstood my quality measurement idea. I mean if you subtract the original and the one after, the LESS voice that is less over or the LESS you can tell when someone is speaking, the better the compression. This is still subjective but I think its easier to tell this way because its easier to tell how much voice is remaining than to tell how much the compressed voice is missing from

Hi, Does any one have any pointers on improving moh quality?? Symptoms are crackling and hissing as the sound comes and goes.. I installed mpg123 this morning.. I have tried various MP3's sampled at 160k, 128k, 32k and 8k and they all sounded terrible... The PC is a P4 so its got plenty of processing power.. I have tried a few different types of classical music (Piano, Violin and full Or...

I'm trying to use Asterisk with a TDM400P and 2 analog lines, but I'm having a hard time getting the kind of audio quality that I'd like. I'm hoping to be able to use SIP phones to make calls through Asterisk and have the same quality as a regular analog phone connected to the PSTN. Are my expectations too high? Is it that once I'm using Asterisk as a gateway to the PSTN, I

Heya all, I'm experiencing some problems with a TDM40P and was wondering if anyone else on this list has similar experiences, or maybe even a possible solution. My setup is a dual PIII-750 with 1 gig of RAM, with an X100P, connected to an analog line to my telco, and a TDM40P with analog phones connected. The TDM-card is not sharing any interrupts, but the X100P is, with the 2 Adaptec SCSI

Hi, I have been trying to figure out why the S100U is not performing very well on RH9.. Here is my thinking..( may be totally wide of the mark but here goes anyway) I remember reading somwhere that the sound system used by RH has changed... Does the S100U not depend on the sound subsystem?? So what I think is that the sound subsystem in RH9 and the S100U are not happy working together.. Does

> QUALITY MEASUREMENT IDEA: > I find it difficult to hear 2 voice samples and tell > which is nearer the original, especially if the > background hiss is slightly different. So what if you > actually subtract the post-compression sound from the > original and then listen to the DIFFERENCE. If you > can't hear any voice except background noise and some > hiss from

Hi, After trying to use the prebuilt Windows 1.1.6 binaries I've founded that the DLL version doesn't export the mode variables properly, this could probably easily be fixed. Anyways, as for now I've downloaded the 1.1.6 source and built it myself and I'm linking speex statically to my application. I've written an experimental software which just packs 20 ms frames and sends

I tried running the festival app today with little success. I have a working festival installation that does TTS to the linux sound output perfectly. With * it just produces a sort of hissing sound. The length of hissing is proportional to the length of text string that it is given to speak. Since I'm running on a PPC system I fear the dreaded endian problem is to blame and that app_festival may need changing. Has anyone else experienced similar problems? Iain