search for: 1khz

Ok here is a real geek question, I building my own linux kernel for my asterisk system and came across the kernel setting for the timer frequency. I have one of 3 hardcode choices 100Hz, 250 Hz and 1000Hz. From what I understand the default Freq was changed from 100Hz in kernel 2.4 to 1000Hz (1KHz) in kernel 2.6. Timing is a BIG issue in asterisk with all the TDM and zap channel stuff. My guess is to go with the lower 100 or 250 Hz option but that is only a guess. The 1KHz sounds like it will conflict with the Zap 1khz timer (or am I wrong about that). Does anyone know what the pref...

Hallo I'm new in the mailing list :) I'm working with a program called mumble that uses speex and the problem is that, if I use a mumble client that tries to reproduce a wave file containing a 1Khz tone, I cannot hear that tone. I've tried to comment all the denoise functions but nothing could solve the problem. I've turned off the VBR, with the function speex_encoder_ctl and the SPEEX_SET_VBR define but the result is that I can here the tone for 1-2 seconds and then noise, terrible...

Jason Hennigar wrote: > Hi Jean-Mark and Marco, > > I encountered the same problem using SPeex on a blackfin where any tone > would pass correctly for between 1 and 2 seconds and then switch to > noise afterwards. Disabling the VAD solves this issue. > This is not a problem, its a feature. A tone is not voice. VAD should suppress it. Regards, Steve

...kbps (this is understandable because in this sample practically there are no sounds above 5kHz) but the artifacts can be easily heard. Mode -b350 gives ~310kbps but still there is a difference (subtle though). I also noticed that you get similar artifacts when you encode simple tones (like 440Hz or 1kHz), no matter what bitrate is used. - The most occurring artifact in -b128,160,192 modes is pre-echo (-b256+ seems to be free from this curse). The leader in eliminating pre-echo is AAC (Liquid Audio, Low Profile AAC), because even at ~128kbps stereo there is very little or no pre-echo (Temporal Noi...

I'm having trouble with ztdummy and I can't seem to figure it out. I am running Zaptel 1.4.12.1 under Debian 4.0 with latest updates applied and I have compiled Zaptel from source along with a new kernel from Debian sources to include 1khz timer support. The modules build fine, yet when I load them I get the following output from dmesg: rtc: lost some interrupts at 1024Hz And then Asterisk fails to load. I don't have any hardware in this box as it will be utilized for a pure VoIP solution so a hardware timing interface is not...

...n be done, but everything I read is rather vague. Check out section 3 at the URL below: http://wiki.centos.org/Manuals/ReleaseNotes/CentOS5.1/ I'm using a 32 bit server installation (no graphical stuff) on a Dell Poweredge 2650. I'd like to compare performance of a 100 Hz kernel against a 1KHz kernel. It reads like it's simple enough to set the divider if I just knew where to set it. -- Ben

Earl Chew wrote: > Erik, > > Ok. I've updated the patches, and made some additions to the test harness. > > For the test harness, I used sox to generate 1kHz wav files over a range of > sampling rates. The test harness runs the replaygain analysis, and compares > the resulting metadata. > > Are you ok with this approach ?? It means that the wav files need to be > checked into the repository. How many? One or two would be fine. The thin...

Earl Chew wrote: > That being said, I think my patch leaves us better off than before ! I agree. We will add it as it is and then tweak futher as needed. Earl, would you be able to update your patch so that it applies against the current git master? Cheers, Erik -- ---------------------------------------------------------------------- Erik de Castro Lopo http://www.mega-nerd.com/

Hello, I am working on a project analysing the performance of motor-vehicles through messages logged over a CAN bus. I am using R 2.12 on Windows XP and 7 I am currently plotting the data in R, overlaying 5 or more plots of data, logged at 1kHz, (using plot.ts() and par(new = TRUE)). The aim is to be able to pan, zoom in and out and get values from the plotted graph using a custom Qt interface that is used as a front end to R.exe (all this works). The plot is drawn by R directly to the windows graphic device. The data is imported from...

Am 19.08.2015 um 19:07 schrieb Steve Edwards: > Please don't top post. > > On Wed, 19 Aug 2015, James Cass wrote: > >> Steve, would you be willing to share that "quick bash script"? > > There's no magic in the script, but here it is, embarrassing myself: > > cp sample-call-file /tmp/ > chmod +x /tmp/sample-call-file >

...on. As pointed out Philipp yesterday to me, it's approximately a 5% drift, apparently caused by the client audio card clock. But, then since there are no cuts in the sound as I am listening, it means the audio rate is lower than the 22kHz of the files, right ? And a 5% drift at 22kHz is around 1kHz difference ! And even if I did not properly measured this behavior, I am not subjectively able to hear a difference in the output. So, what are the solutions for locking clocks so that the latency is constant ? Sorry if I don't understand properly. Best. Sylvain

Hi, I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple "hello world" dial plan. I have tried the release of 1.4 and also 1.4 svn and both display this issue. I have also tried it on a dedicated linux box and on a linux install running under

...=> 1234,n,Playback(beep) ; Beep to let them know the test has started exten => 1234,n,Echo ; Do the echo test exten => 1234,n,Playback(demo-echodone) ; Let them know it's over exten => 1234,n,Goto(s,6) ; Start over ; send a 1KHz test tone exten => 1235,1,MilliWatt ; play the hold music exten => 1236,1,Answer exten => 1236,n,MusicOnHold(default) exten => _X.,1,Dial(DAHDI/R0/${EXTEN:4}) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pi...

Copy additional filter values from Foobar2000 as found in <http://code.google.com/p/sirens2/source/browse/trunk/libwavpack-4.32.psp/wvgain.c?r=32> to allow metaflac to perform gain analysis on high sample rate audio. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac-dev/attachments/20120107/018c821b/attachment-0001.htm

Hello Earl, I used your patches while they cleanly applied to the Git tree. There is one question though, what do you think of?. http://www.hydrogenaudio.org/forums/index.php?showtopic=84086&st=0&p=782407&#entry782407 which apparently states (if I understand correctly) that your approach is not completely correct. Kind regards, Michael -------------- next part -------------- An

...not an expert at DSP work, and I'd gladly defer to any one with any reasonable amount of DSP experience. I followed the link, but I'm not sure how to interpret the graph exactly. I believe it's showing filter behaviour. The two curves look similar, but are clearly not the same. At the 1kHz mark, the blue shows -23, and the green shows perhaps 17.? Are they supposed to be the same ? I believe the approach I took is reasonable based on my limited DSP knowledge and some reading, but is probably not even close to being optimal. That being said, I think my patch leaves us better off tha...

Erik, Ok. I've updated the patches, and made some additions to the test harness. For the test harness, I used sox to generate 1kHz wav files over a range of sampling rates. The test harness runs the replaygain analysis, and compares the resulting metadata. Are you ok with this approach ?? It means that the wav files need to be checked into the repository. (There are already some small image files in the repository as a preced...

...flac-dev at xiph.org Cc: Sent: Tuesday, February 14, 2012 12:41:51 AM Subject: Re: [flac-dev] Regain play analysis patches Earl Chew wrote: > Erik, > > Ok. I've updated the patches, and made some additions to the test harness. > > For the test harness, I used sox to generate 1kHz wav files over a range of > sampling rates. The test harness runs the replaygain analysis, and compares > the resulting metadata. > > Are you ok with this approach ?? It means that the wav files need to be > checked into the repository. How many? One or two would be fine. The thin...

Is anybody using XEN in conjunction with Asterisk and/or Openser? I would like to get some info about such an environment and experience reports. bye Ronald Wiplinger

I'm using ices2's metadata facility to update the name of a track in a vorbis stream: <input> <module>alsa</module> <param name="rate">44100</param> <param name="channels">2</param> <param name="device">hw:1,0</param> <param name="metadata">1</param>