search for: amplifies

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

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

Hello, I'm having the following problem when using a headset on XP connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus motherboard: - Using any sound recorder (Windows', Audacity, XLite), the level is just too low when speaking at a conversational level, even with the microphone level pumped all the way up (line displayed totally flat in Recorder)

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

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

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

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

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

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

...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 plugged in, speex_denoise() smoothes > the signal and produces smoother (and even amplifies the signal) > speech > like signals. It seems that speex_denoise( ) is very sensitive to > static noise. What happens (denoiser amplifying the signal) is that there is also an AGC (automatic gain control) that thinks that the noise is speech (because it's the loudest signal is hear...

...f all users of my software. > (All of this is Win32) > > What happens is that when no one is talking I think Speex pre-processor > starts to think that the very quiet leak from my playback path is > actually someone talking. > I am using AGC and VAD and my guess would be that AGC amplifies the leak > to such a level that VAD starts triggering. > This is rather annoying for the user if the software is used while > playing MP3s (loud) or playing computer games. > > Would it be possible to implement some sort of a threshold to limit what > Speex classifies as "us...

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