similar to: Amplify Ogg files without decode/encode

Compiling libao on a system w/linux2.4.2 & glibc-2.0 (I think), I get an error. gcc -DPACKAGE=\"libao\" -DVERSION=\"0.6.0\" -DSIZEOF_SHORT=2 -DSIZEOF_INT=4 -DSIZEOF_LONG=4 -DHAVE_SYS_SOUNDCARD_H=1 -I. -I. -I../../.. -I../../../include -O20 -ffast-math -D_REENTRANT -fsigned-char -DAO_PLUGIN_PATH=\"/usr/local/lib/ao\" -c ao_alsa.c -fPIC -DPIC -o ao_alsa.lo In file

I've just started encoding some of my cd's to ogg with the new beta4-release, and noticed a few "bugs" .. [oggenc] (*) When using long filenames that make the "encoding.."-line not fit in one line, that line is repeated 1000 times on the screen (one for each .1 percent). That could be solved either by not including the filename on the updating line (but above or

Hi. I created a patch for Bug #4 (ogg123 not honoring /etc/ogg123rc) and uploaded it to Bugzilla, but I have the feeling that the Bugzilla is somewhat neglected... so I just send the patch here. The changes are very simple, one typo (I guess) was fixed (from "/etc/ogg123.rc" to "/etc/og123rc" as it's documented), and I changed an if to a while-loop to make it read the

> -----Ursprüngliche Nachricht----- > Von: David 'Superkind' Roden [mailto:superkind@uni-neukoelln.de] > Gesendet: Donnerstag, 5. April 2001 22:53 > An: vorbis@xiph.org > Betreff: Re: OT: long - Replacing CD's? was RE: [vorbis] New type of > copy-prot ected audio CDs are coming... > > Why? That 2nd part was perfectly demonstrated with the > MiniDisc. I

Hi I wonder if you can help... I have setup shorewall(2.2.3) under debian on a machine that has 4 network ports... the idea is that there is 1 WAN port, 1 DMZ port, and 2 LAN ports, 1 LAN port has static NAT setup for selected incoming connection from trusted sources, and the second LAN port I am trying to setup using masq NAT as it only requires outgoing connections, no incoming. the static NAT

Hi, folks. I have a problem using Asterisk 1.2. I create conferences using app_meetme and Zap channels, and for every participant I run the script defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF tones. As the docs tell me, when using the AGI background script one loses the ability to control the meetme conference via the command line so for muting conference participants I

Echo cancellation works like a charm, but it seems to confuse the preprocessor a bit. If listening to background music (properly fed through the echo cancellator), the music is removed but the result is still detected as speech even if almost silence remains in the signal. Also, the AGC keeps adjusting to the minute remains in the signal, meaning that sooner or later it will amplify the

My data is speech sampled as pcm mono 32khz. I want to use vad for regulating the agc gain mechanism. So... if voice is not detected I just reduce the agc value in order to not amplify the 'silence' otherwise we just increase the agc value if it is below the minimum. Thanks, Daniel ----- Original Message ----- De: Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> Para: danflu at

Hi there, i called one asterisk server from another asterisk server. The calling server played back a audio data und the answering server recorded the audio sample using record() function. I tried both ISDN, VoIP connections. Camparing with the original audio data, the recorded samples from both connections were amplified by asterisk, so that the recording were much louder. But I didn't

Hello there, recently installed CentOS6 on a (quite old) 64-bit system, and from the GNOME's sound preference app, I can see that the Output volume range is said "amplified" from 63% to 100%. Below 63% it's unamplified. No idea what's implied behind this amplification (I don't see this on other desktops), but I notice that when "amplified" the sound is pretty

Jean-Marc Valin wrote: >> I've had a small case with noise suppression and AGC. I have a fairly >> noisy environment here, and with the default parameters, noise >> suppression works fairly well while I talk. However, when I shut up, AGC >> starts slowly increasing the gain until it has amplified whatever noise >> is left to levels about equal to having no

On Thu, Dec 17, 2015 at 9:21 AM, Andy Ayers <andya at microsoft.com> wrote: > While your bb count distribution is extremely likely to be some kind of power-law like distribution, it's not guaranteed. > > Also you might think about operations that can amplify (rerolling) or appear to amplify (TRE) or diminish BB counts, and how you'd go about reclassifying block hotness. yes

-----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

Hello! I have a cabinet full of Wilcom Enhanced Line Powered Amplifiers with Manual Balance, model ELPA-421V. I *believe* these were used for a bank of analog modems back in the mid-90's. They were removed from a suite when the old company moved out. Here's a URL: http://www.wilcominc.com/elpa421v.htm Does anyone have any interest in these? If so, please reply off-list. Tim

I think you'll have to modify Speex to get the functionality you're looking for. I've made a few simple modifications to the AGC to prevent it from 1) exceeding a specified level of amplification and 2) enable and disable adaptation, so I can freeze it at a certain level while speech is not detected. It's mostly just a matter of doing this at the end of speex_compute_agc():

Hi, The leakage problem you describe is very, very common and you will need to do something to address it. I modified the version of Speex I use to implement an adjustable max gain. If you look at speex_compute_agc in preprocess.c, you will see: if (agc_gain>200) agc_gain = 200; This max of 200 is usually more than enough to amplify leakage which occurs either in the sound

> Feature Requests item #310492, was opened at 07/01/2008 09:57 > Status: Open > Priority: 3 > Submitted By: Arnaud Quette (aquette) > Assigned to: Nobody (None) > Summary: Allow to specify hostnames in ACL (upsd.conf) > Category: None > Group: None > > > Initial Comment: > allow a new ACL form: > ACL hostname/mask > > example: > ACL localhost

agc_gain seemed to fit with the idea of what I wanted to do, it was easy to understand its units and behavior, and freezing it produced the desired results. Also I wanted to cap it, so that's done at the same place, and that definitely works. All I want to do is be able to freeze AGC adaptation and put an upper bound on the AGC (for example, 2x amplification). Both of these things seem

Hi, On Wed, Nov 1, 2017 at 5:32 PM, Amye Scavarda <amye at redhat.com> wrote: > Welcome to an exceptionally busy month for Gluster! > > Gluster Summit > Thanks to all who attended this year's Gluster Summit. As you can see, > conversations are continuing to happen, with notes from the Birds of a > Feather sessions starting to come into the mailing lists. If something

I've done gain tuning as per the info I've found online. I've got my RXGain set so my volumes list as about 14,800 (using a milliwatt test number and ztmonitor -vv). However listening to the line now, this sounds too loud to me. The person speaking sounds fine, but I've now got a large amount of background hiss coming thru. In order to get the "recommended"