search for: loudspeaker

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! I want to use speex for echo cancellation in my program, but I have bad results. I will explain what my program does. it is a client-server application. I run a server in room A and a client in room B. the client sends some voice to the server and the server plays it on loudspeakers. I run another server in room B and connects to it from room A using the same application that runs server in room A. the client from room A records from microphone and sends the recorded voice to server in room B. the server in room B plays it on loudspeakers. in room A there is also anot...

Hi, I plan to use AEC for a live performance, storytelling for very young children (and their parents!) in a mongolian yourte . Actually the storyteller can make vocal loops, there is an omnidirectional microphone in the center of the yourte, 5 loudspeakers in a circle along the yourte's wall and Pure Data in a linux box. And now she wants to make vocal loops over music and loops over loops... Maybe aec will help her? I did some testing on a small setup, 2 desktop loudspeakers and have very good results (20 to 30dB rejection) with mono music and...

...re removed from the signal transmitted to > the far-end. The general approach is illustrated in figure 2." This modification is a gain multiplication in each frequency instead of frequency domain subtraction in common frequency domain echo cancellation. See ... > As can be seen, both the loudspeaker and microphone signals are first transformed into the > frequency domain by a spectral transform (ST). Based on these input signals, the control unit of > the Acoustic Echo Control determines an optimum gain factor for each individual frequency > band separately. These frequency dependent...

Im looking for an Intercom solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a "loudspeaker" ? Thank you, Steve Maroney

I'm looking into getting a new phone and wondering what the difference in functionality is between a single line phone with call waiting and a real 2 line phone (either a real SIP phone or an analog 2 line phone and a 2 port ATA) is. Why would I want the real 2 lines vs. just being able to take an incoming call via call-waiting? Cheers, b. -------------- next part -------------- A non-text

Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I have to open 50 connections and send 50 times the same packets or if can use in some way the multicast. Is there anybody that may give me some idea. Thanks in advance,...

...out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to Linksys PAP2T ATAs as part of a call group to get a loud (advertised as 95dB) ring out there, but it still could be louder. Maybe a light-up option would be better. The old phone system here had some huge loudspeakers that someone had wired right into the speakers of the old digital phones. I haven't figured out yet if they need a different voltage, or even if they still work; they were not responding when I replaced the attached phones. Justin C. Sherrill - American Rock Salt p: 585-991-6825 f: 585-991-6...

I can only evaluate this with my subjective point of view. I had a special test scenario doing chat with cheap webcam microphones and loudspeakers. Fraunhofers solution was the only one that could eliminate the echo. In double talk the quality gets lower but is still very good. You might want to ask Fraunhofer for a demo version to test for yourself. I have no details on the algorithms being used, I only know that it is patented. Since it...

Hallo, an alle Wine-Freaks, das Programm - bis jetzt Freeware-, ist High-End unter den Hornsimus: http://www.users.bigpond.com/dmcbean/Setup.exe , 400 kB Ich habe wine in etlichen Versionen bis zur 0.9.26 probiert, auf Debian und Mepis (ubuntu), auch Crossover von 4.2 bis zum Beta 6.3a. Es geht nicht, alle Dinge wie native dll,s andere Winvers bringen keine Besserung, IE, Dcom--hilft nicht. Alle

Not sure yet what or where the problems are but having just done a jum update on my HP laptop (nw8240) and my IBM desktop from 2.6.9.42.0.2.EL the laptop is locking up during boot sequence and the desktop when running VMware Workstation seems to take all CPU and makes strange noises from the loudspeaker. Using grub to fall back to last kernel all is ok again !!. Ian -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20061017/b02ac026/attachment-0002.html>

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 original signal present in a live recording merely degrades the playback system's ability to reproduce the audible signal. Because a perceptual coder removes inaudible signal content,...

Hi everyone, I've just released Speex 1.2rc1. This adds support for acoustic echo cancellation with multiple microphones and multiple loudspeakers. It also adds an API to decorrelate loudspeaker signals to improve multi-channel performance. In the bugfix department, there are fixes for a few bugs in the echo canceller, jitter buffer and preprocessor. At this point, the API for 1.2 should be stable and only a few very minor additions are pla...

...t doesn't seem to work. Does anyone know a tool that can do this under linux? The ac3 stream is also avialabe as wav6 file converted by the cvs version of a52dec. This wav6 file contains one stereo stream and 3 mono streams. Is there a convention which stream number in an ogg file means which loudspeaker position in the wide spread 5.1 sourround systems? (Or in the better 7.1 systems) Thanks alot and bet regards Lars

...ftware that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV format Email messages Web based config utility Compatibility with most phone systems (Panasonic, Avaya, Vodavi, Nortel...) Web based User control panel Class of service groups Pager-email-cellphone-loudspeaker notify Message waiting lamp notify Is there anything in the open source arena that you all can think of that might work?? Thank You, John Berry, Owner Lantex Voice & Data Systems 312 Billow Dr San Diego, CA 92114 619-690-4428 fax619-741-6821 john@lantex.com lic#805538 -------------- nex...

...m and tie that into the Asterisk PBX, the way such intercoms have been connected to other PBX's in our district in the past, or do we put IP phones in the classrooms and tie that into a PA system for paging? Are there IP based paging systems that could be used instead of the traditional PA/loudspeaker systems in most schools? We will be writing a spec shortly (and I will be seeking a consultant on asterisk-biz soon to assist us), but I need to know whether I need to get our engineering consultant to redraw the cabling to reflect a data jack at the location of the classroom phone instead of t...

...eex-dev@xiph.org Oggetto: Re: [Speex-dev] echo cancellation > echo cancellation: > speex_echo_cancel (st, ptr, echo1[0], e_buf, noise); speex_preprocess > (den, e_buf, noise); > > where ptr is a buffer that was recorded from microphone, echo1[0] is a > buffer that was played on loudspeakers. the delay between ptr and > echo1 is 2 buffers (512 samples). > with this configuration I can still hear the echo signal (althought > the signal is a little more silent. when using the delay 3 buffers the > echo is still not cancelled and after few seconds I don't hear anything...

...t;> music that has a few "long" sounds, and saying "aaaaanyway" is enough to >> trigger this. > > Can you sent a pair of files so I can run testecho on? I'll need to add support for saving audio to my program, so I can give you the "actual" sampled loudspeaker and mic files, and I'll also need to get hold of a test person again. (I had a friend with a friend who has an exceptionally clear voice. My own "aaaaaa" is far too muddy to cause this). I'll try to get this done this week, but it might be delayed 'till after christmas. &...

I'm trying to convert my current headphones and microphone chat application to support loudspeakers and microphone, and so I thought I'd give speex_echo_cancel() a try. Since my users quite frequently have other sound-producing applications running on their computer (such as winamp), I sample 'wave' recording device of the soundcard in addition to the microphone. I then call spe...

...> Also if you look at the samples for (e.g.) a 20kHz sine samples at 44kHz, > the samples hardly resemble a sine wave very mch, and seen reversely: > It's not obvious that it once was a pure sine wave. Please remember that this is *not* how the sound is sent to your amplifier or loudspeaker when you press Play. If the DAC on your soundcard works properly, the signal is first lowpass-filtered so that it only contains frequencies that are lower than 1/2 of your sampling rate. If this is done, then the discontinuous steps are not present in the signal that comes out of your amplifier...