search for: hiss

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...

...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 chorale) where Opus produces some strange high-frequency hiss/resonance, whereas WMAPro does not seem to have this issue (although it audibly seems to change the bitrate after a few seconds). The hiss is (at some extend) already present in the original file, yet Opus seems to amplify it. You can find the example uploaded here (please ignore the certificate w...

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...

...100P 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 the hiss and buzz is still on the other house lines. The only thing that fixes it is rebooting the asterisk machine / or unplugging the FXO card from the wall. Any ideas on what is causing this to happen or what might be don...

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

...e 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 nearly sounds as if the encoder adds 'hiss'. Beta4 at 128Kbps does not have the same artifacts. Monty noted on IRC: >>> The biggest to expect is that very high frequency diffuse noise will tend to shrink toward a point. >>> However, this sounds like it's something else than what I am experiencing. -- GCP --...

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 "s"'s and "d"'s and such, that means most of the actual voic...

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 19.5kHz. -- Vorbis Xtreme | http://solair.eunet.yu/~aldov/ Ogg Vorbis is the free, open source alternative to MP3 --- >8 ---- List archives: http://www.xiph.org...

...ding 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. The mp3 had a little more (but acceptable) hiss, but it had better sharpness in the higher frequencies (hence the hiss). Overall i rank the quality even. I tried LAME mpeg-2 layer 3 on the converted 16khz wav, but i still had to go up to 32 kbps (same file size) to get the same quality as speex wideband. So all in all, i estimate speex to have...

...the sake of stability or audio quality, I will. I haven't found anything yet that meets all these. Here's what I've tried and the results. ogg123: The obvious choice. But it can't seem to do it for long. After several hours it begins to get a very repeatable pattern of quick hisses. Usually pairs of two, the first one long, the second short. Perhaps every 2 to 3 seconds. Also, I notice that the buffer keeps decreasing as it plays. I've tried with multiple sound cards just in case it was a sound card clock inconsistency, but it does the same thing in both cases....

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 disguise this I've been implementing various 'amplifier...

...do you prefer: a codec that sounds good or one > for which > subtracting the original gives a better signal? > > 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 "s"'s and "d&q...

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...

...the PSTN, I have to expect a certain amount of voice quality degradation? I've never used Asterisk before so I really don't know what is truly possible and what isn't. I'm getting a slight echo...sometimes...it varies from call-to-call, but the biggest problem I have is a constant hiss in the background. Again, this varies from call-to-call. I know my SIP phones are fine as SIP-to-SIP calls on my LAN work perfectly. I only have problems going out to the PSTN. So for those of you isng a TDM400P, would you say your voice quality is as good as a pure analog/PSTN call? I have don...

...;s mostly idle, as I only use it during backups. Not loading the SCSI-drivers at all doesn't make any difference in the problems below. All of the disks in the machine are IDE, and those controllers have their own interrupts. My problems with this setup are as follows: - I hear a soft fading hiss in the background on the TDM40P-connected phones. It's also present when I call something like voicemail or a conference on my local box, so it doesn't seem to come from the X100P. If I uncomment the "NO_CALIBRATION"-option in the Zaptel Makefile, the hiss decreases in volume, b...

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 "s"'s and "d"'s and such, that means mos...

...Wide band, Quality=3, Complexity=3, SampleRate=16000, BitRate=38400, AvgBitRate=38400, ENH=false, VAD=true, DTX=false, DENOISE=true, VBR=false, AGC=true, AGCLevel=16000. Using a peer to peer setup the sound from my peer get corrupted in a very strange way, he sound (my debug friend) sounds like a hiss, it's just like if you would cut out everything that makes sound to speak and keep the rest. This doesn't happen with my sound sent to him. Looking at a 44100/16 sample of his sound I've found that he has a pretty strong noise at 50 Hz. What can I do about this? //Regard Jonas T?rns...

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