similar to: High Frequency Hiss with Opus at 48 kbit/s

Hi all, I''ve got a BOX running CentOS 4.5. It acts as a firewall + router. I have installed both iptables and iproute2. I has 3 network cards. eth0 is connected to Internet (is has an internet ip. pls assume its ip is 1.2.3.4/29). it is a 256 Kbit link. eth1 is DMZ. its ip is 192.168.100.254 eth2 is LAN. Its ip is 192.168.101.254 I have alreday shaped traffic to 64 Kbit on eth1 for

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 the noise a cpu fan makes.. Like a high piched hissing noise.. Any one

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

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

Hi Jean-Marc, thank you for looking into this. I'm using the current release version of libopus (1.2.1) and version 0.1.10 of opus-tools, both built from source. Libopusenc is the current git master. Indeed I cannot hear any glitches in continuous.opus, continuous.wav, and chained.wav. chained.opus decodes just fine with opusdec as well, but doesn't play properly in any standard audio

Hi Jean-Marc (and everyone else who replied), > Considering you're switching to Ogg, I think you should give libopusenc> a try. It does a really good job at getting rid of *all* discontinuities> -- to the point where you can chop a song into files less than one> millisecond each and it still sounds good. It's also pretty simple to> use. You just feed it audio and tell it

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

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

Hi Andreas, So I encoded your file in chunks with a slightly modified version of opusenc_example and I can't hear anything wrong. Maybe there's a problem in the tools you used? I uploaded the files at: https://jmvalin.ca/misc_stuff/continuous.opus (one file) https://jmvalin.ca/misc_stuff/continuous.wav (one file, decoded) https://jmvalin.ca/misc_stuff/chained.opus (many small files)

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.

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

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

Why does opus scale down frequency from 22k to 20k ? 64kbps<https://lh5.googleusercontent.com/-DGsEd1ijTBs/UDE3OFVrE5I/AAAAAAAADBo/bFcWQ6I4-Zk/s588/opus_64_test.png> 128kbps<https://lh3.googleusercontent.com/-3Wfvq6OPGRw/UDE3OClxEdI/AAAAAAAADBk/vsBJ9Gcd_pU/s589/opus_test.png>

Hi, I have just built my home Asterisk box into a better PC that became available (still only a P2 350 but it only has to manage 1 analog line).. Anyway I have built it on Fedora Core 1.. I have an X100P and a TDM400P (1 module installed).. These cards were working fine in my older PC that was running my Asterisk at home.. The inbound calls via the X100P to my sip phones are working great..

I'm wondering if anyone has seen (heard!) this before. I have a site which has Grandstream Budgetone 100 phones (don't laugh, they weren't my choice and I was quite angry when I heard they'd been installed )-: They have an asterisk box with a TDM400 card in it with 4 FXO ports and 4 lines to the telco (BT, in the UK) It basically works and does what it says it's supposed