similar to: feedback about speex 1.1.6 [echocancel]

Hi, I am currently working on a thin-client "low-cpu" usage telephone application. The current setup is that you can make phonecalls using an alsa supported usb headset and a smartlink based usb modem (a sweex usb modem, which -in large quantities- goes for 17.95 euro's) for analog lines. Unfortunately, it is really made as a modem and not for audio purposes, so, there is a pretty

>> 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

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

> Err. Ok, as I got it, 'bin 0' has it's amplitude in W[0], bin 1 to N-1 has > it's real part in W[i*2-1] and it's imag in W[i*2], and finally the > nyquist amplitude is in W[N-1] Not quite, it's packet "real, real, imag, real, imag, ...". > I took this from how power_spectrum() computes, so I might be off :) But power_spectrum() handles that

>> Actually, computing the "power spectrum" for each frame of W shows >> how large an ammount of the original signal at time offset j the >> echo canceller thinks should be removed from the current input frame. > > Careful when looking at W because of how the real and imaginary parts > are packed in the array. Err. Ok, as I got it, 'bin 0' has it's

Hello, I am testing speex 1.1.6's echo canceller. I am using testecho.c, with a few modifications to get it to run on Windows. My problem is that I am unable to get the echo cancellation to work correctly. I am working on an audio conferencing software, and one issue we have is sometimes the microphone picks up what is being played through the headset, resulting in an echo of the other

Thank you, Andreas Engel. I downloaded the white paper of the Fraunhofer Acoustic Echo Control. http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf It said > "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is > modified so that the undesired echo components are removed from the signal transmitted to > the

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

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

> As the thing is to encode for human ears (AFAIK), I'd say that 4kHz is already "quite high", > and I wonder who can actually hear pure 20kHz sine. If you read the beginning of RFC 6716, you learn that Opus never encodes any frequencies that are higher than 20 kHz. So at some medium or high bitrates, anything above 20 kHz is filtered out, not because of the bitrate but

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

Hallo, I'd like to get all channels of an ac3 5.1 stream coded into vorbis. With sox and oggenc it 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

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

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

My school district will be building a new elementary school in 2006. We were about to go to bid with a traditional intercom system for the campus but I would like implement Asterisk at the campus. My question is, do we build in a traditional intercom/paging system 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

A previous issue has popped up and once again I'm out of ideas. During the evenings it seems that the TDM channels will spike (dahdi_monitor) and will refuse to listen for audio of any type, this includes DTMF. The only resolution I know of is to stop Asterisk and restart the dahdi service, but that's not a solution. All channels look like this, even the FXS. [root at asterisk Hardware]#

An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20040715/b3253d77/attachment.htm -------------- next part -------------- ? Hi, I have some queries: I am using speex in Windows platform. 1. I am having some problem with the speex,for input 8KHz,8bits/sample mono stream i am getting output as 8KHz, 16bits/sample mono stream. I would like to know

The client is the adobe flash player. No install and on 98% of all desktops but we can't change it. It works ok if people use headphones but we need to stop the howl than can build up if more than one person in a conference has mic to close to speakers. Any ideas? Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > 1) Is it ok if the audio is encoded (using

> 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent > to the client and decoded when it is recevied so the AEC is always > performed on raw PCM16 8KHZ ? No. The entire path from AEC to loudspeaker and from mic back to AEC must be free of any non-linearity, codec, drift, ... > 2) The audio is moved in 32ms (512 byte) chunks and the reading and > writing to the

> I'm wondering if there's a way to speed up the initial convergence > time. Maybe I'm completely off base here, but specifically I was > wondering if the code starts the search assuming the echo is near to > zero, and works longer to find the adapted point. If so, is there a way > to have it reverse the search, that is, start with the assumption that >