search for: amplified

...lled 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 adjust any rxgain or txgain. I am wondering, how could it be!? Is this case normal? Can anyboby tell me, how many dB will the transmitted and recieved signals be amplified by asterisk in case of default rxgain and txgain (with an...

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 bad (at 100% and above, it's ridiculously bad). Below 63% it's simply very...

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 have been looking for analog phones for my * system that work with our plantronic amplifiers and headsets. The problem I am having with the Aastra phones that I have purchased (PT-390, 9116, 9120, 8009 ), is that they don't seem to stay hung up unless you physically hang up the handset everytime you finish a call. I have even purchased the Aastra 9120 which sais it has on-hook

> Just curious, is it possible to amplify Ogg frames without a decode/encode > cycle? I usually normalise before encoding all recordings for my radio > station, but I obtain some music from 3rd parties and it often need > normalising. I find this ability very useful for MP3s and it would be a big > handicap to me if Ogg can not. Add a comment into the ogg file :

I noticed that my previous message is not very complete so I send here an "enhanced version". Please disregard the old one an reply to this one only. ( you can delete the ATTENTION word from subject ) Wilson (defiler@null.net) wrote : > There are two ways to decode multi-channel audio. In hardware, or in > software. > Hardware: A receiver or processor takes a Dolby Digital

...81/headsetlowvolumeecho.jpg - In addition, when making a call with XLite and Asterisk, I get a bit of echo - Same issues when trying with a different headset - Enabling "Auto gain control AGC" in XLite makes no difference. Any idea what can be done? Should I use a different soundcard? Amplified headset? Can something be done in Asterisk about the echo? Thank you.

What headsets have people found work well with the Cisco 7940 and 7960 phones? To date, I have tried a couple of the headsets within the Plantronics H series (H41-N), and noticed that the volume of my speaking is lower over the headset than on the regular handset. I am currently looking for headsets that are known to work well. I do know that Cisco lists the H-91 and H-101 as certified to

I see a lot of chatter in the archives about intercom and paging, but has anyone addressed zone paging? Each of the 50 telephones in a large clinic would be members of one or more paging zones. Someone could then page Dr. X in zone #1. Would this be possible with analog phones? SIP? Thanks, Mike

..., 2x amplification). Both of these things seem necessary in a real-world app because: 1) AGC gain should not increase when speech is not detected. If it does, then it will inevitably rise during periods of inactivity on the part of the speaker, and then background sounds will be end up being amplified too much and detected as speech. This is a problem regardless of echo. 2) The upper bound is necessary in some situations when VAD is not sufficient to distinguish between desired and undesired sounds. For example, consider a person using a headset and communicating infrequently while consta...

Hello all. We have an asterisk working perfectly but we need a sollution for the PA system. Before Asterisk PBX we had an expensive analog PBX which plugged an extension into an audio amplifier, and that was the PA system. Now, the Asterisk server is quite far from the audio amplifier and it has no audio card. So my idea is to plug the audio card of another linux server, which is over the

Hi everyone, I have a user that needs a little extra volume on his Polycom IP 601 phone set for all calls (beyond what the volume control currently offers). Is there a provisioning setting for this anywhere? (I'd like to avoid a separate amplifier between the phone and handset if possible.) Alternatively, is there a way to have Asterisk 1.4.x boost the volume to a particular SIP device

...background music or only the other end talking while I shut up): - Zlast (which looks like a SNR variable) is at 0.05-0.2, but jumps up above 1.0 if I actually say something. - loudness2 keeps decreasing from the "normal" of ~6000 to 1000 or so, at which point the residual echo is amplified enough that it's clearly audible at the other end. If I say something, it adjusts. - speech_prob is at 0.999 or 1.000 as long as the other end talks. This is all with up-to-date SVN version of speex, and in a fairly noisy environment (it's hot, so I have the window open, so passing car...

Hi, 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 filtering at all. As soon as I talk, AGC backs down fairly quick and the noise is again gone. Decreasing the AGC max gain and increasing the ammount of noise suppression fixed things, but it might be an idea to change the defaults sligh...

I quote from "Principles of Digital Audio" by Ken C. Pohlmann: "Because perceptual coders tailor the coded signal to the ear's acuity, they similarly tailor the required response of the playback system itself. Live music does not pass through amplifiers and loudspeakers, it goes directly to the ear. But recorded music must pass through the playback signal chain. Much of the

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

What rigs do you folks use to listen to your music? I have a P-III 500 with Altec Lansing speakers in the dining room and a P-II 350 with Labtec speakers in the Guestroom/office. Sorry, I can't remember what model the Lansings are off the top of my head. The Labtec speakers are fairly cheap. I have a PCI ensonique sound card in the P-III system. I not sure what kind of sound card is in the

> Then I pulled the microphone out. Our system still records noise. To > isolate the problem, I wrote a small app just to open the device and > record raw samples, calls speex_denoise() and outputs both sample > sets. > The noise is still there, with level fluctuating with gain level, > unless > "All mute" is chosen. > In the case when NO microphone is

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

On 6/7/14, 1:55 AM, Jean-Marc Valin wrote: > Actually... no! 24-bit can indeed be useful as extra margin and Opus > can actually represent even more dynamic range than 24-bit PCM. That's > not the case for 192 kHz. There's no "margin" that 192 kHz buys you > over 48 kHz. You can do as much linear filtering as you like, the > stuff above 20 kHz isn't going to