similar to: multiple sound cards - howto have IP phone on each ?

I'm developing an application that have a video conference component. For that I need echo cancellation, and is looking around for algorithms/implementations of that, and the one in speex is an alternative. In the documentation for speex I find the following sentence however. "Using a different soundcard to do the capture and plaback will *not* work, regardless of what you may

Hi i have an @home box with Uniden UIP200's and the speaker phone works great for recording my voicemail box and recording the digital receptionist menu but when i make calls out the PSTN the mic on the speaker phone acts like it only picks up when my voice is really high, make a noise and get closer to the mic all the sudden it starts working then if i back off it quits and visa-versa, but I

Are you sure you're not just inverting the two inputs? Jean-Marc On Wed, 2005-11-09 at 22:16 -0800, Jason Harper wrote: > I ran some further tests on mdf and here are the > results: > 1. reduced tail length to 100ms, aligned mic and > speaker signals to within 10ms - almost no echo > attenuation > 2. aligned mic and speaker signals to within 5 samples > - still almost

I'm pretty much sure of it. When I test inverting the inputs, my output is pretty much the same as my speaker signal. Whereas the way that I normally test the output is my mic signal with very little attenuation. If you are interested I can send my test files; they are about 94KB each. -Jason --- Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > Are you sure you're

I ran some further tests on mdf and here are the results: 1. reduced tail length to 100ms, aligned mic and speaker signals to within 10ms - almost no echo attenuation 2. aligned mic and speaker signals to within 5 samples - still almost no echo attenuation 3. ran testecho using the same file for mic and speaker - very good echo cancellation (of course this is expected, but I needed to do a sanity

Hello! I'm attempting to implement Speex AEC support in GStreamer, using work started by Olivier Cr?te. In the Speex manual, I see this text: "Using a different soundcard to do the capture and plaback will *not* work, regardless of what you may think. The only exception to that is if the two cards can be made to have their sampling clock ``locked'' on the same clock

I'm having a problem with sound output to the console. My basic dial plan is as follows: exten => _1NXXNXXXXXX,1,Dial(IAX2/####@voxee/${EXTEN},30,A(beep)) exten => _1NXXNXXXXXX,2,Playtones(info) exten => _1NXXNXXXXXX,3,Hangup I get the following output in the console: ___*CLI> dial 1#######@voxee -- Executing Dial("ALSA/default",

I've got a problem with audio output from the Asterisk console. I'd really appreciate any help. I'm simply trying to dial out to a phone on PSTN. My extensions.conf entry is as follows: exten => _1NXXNXXXXXX,1,Dial(IAX2/####@voxee/${EXTEN}) exten => _1NXXNXXXXXX,2,Hangup After starting asterisk and dialing, I hear a ringback tone through the console speaker, and the PSTN

This kind of behaviour is odd. One of the reason could be the fact that you're using a really long impulse response. Try syncing your signals and making the tail length more in the order of 100 ms to 300 ms. Jean-Marc Le dimanche 06 novembre 2005 ? 21:25 -0800, Jason Harper a ?crit : > Thanks for alerting me to the new changes. I just > tried the latest code from SVN, but

All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4

Thanks for taking a look. There was no motion; however you are right about sampling from a different card. The speaker is connected to the Sound Blaster card, while the microphone is part of a USB webcam. I don't think that this is likely to be too unusual a configuration among users. I can retry the test using a sound card microphone to see if there is a difference. If it turns out that

Anyone got any tips on improving sound quality on soft phones running under Window XP SP2? I have tried Xlite, SJPhone and Firefly. They all seem to have significant sound quality problems. We have a reasonable sized network of several hundred devices connected together using Layer 2 switches, i.e. pretty dumb switches with no QoS. I also have a Grandstream connected to the same switching gear.

Thanks for alerting me to the new changes. I just tried the latest code from SVN, but unfortunately I still have just about the same results. The estimated echo that gets subtracted from the actual echo is such a small signal that it doesn't really result in any noticeable echo attenuation. I currently have my filter size set to 2 seconds even though the echo in my microphone file is only

When I ran test 4 as originally described there is substantial echo cancellation (but not as good as when the files are perfectly aligned). When I invert the inputs, there is no noticeable cancellation. I'm using testecho with the preprocess line commented out. Preprocess seems to work very well at cleaning up the residual echo when mdf does its job, so I'm just focusing on testing mdf.

This is a very real problem though.. I've encountered many sound cards that use different clocks for input and output (even on the same card!) Also, if you open up a sound device on windows at 8kHz, the microphone is often around 8100Hz, while the output is 8000Hz.. I'm not sure if there's a bug somewhere in some of the OS resampling algorithms, but I've seen that on many machines.

Hi, I'd like to try using a good quality microphone and a set of PC speakers (in the first instance) to create a powerful speakerphone; if I get that working, I'll probably try more elaborate audio equipment. For this to work, I'll need software acoustic echo cancellation, or the caller at the other end will constantly hear his/her voice echoing back. I gather Asterisk can do

I wasn't implying that anyone do anything about it, just that's it a real problem. Unfortunately, most of the crappy sound cards are the ones that ship with your typical PC, so it's just something that people should be aware of. The solution is pretty straightforward -- just resample the audio data in real time using a reference clock. -----Original Message----- From: Jean-Marc

Thanks Geoff, it's becoming more clear to me now... > So, assuming it does, you could try: > aumix -w r [yann@raglou yann]$ aumix -w R [yann@raglou yann]$ aumix -q vol 100, 100 pcm 100, 100 speaker 0, 0 line 0, 0, P mic 4, 0, P cd 0, 0, P igain 0, 0, P line1 0, 0, R phin 0, 0, P phout 0, 0 video 0, 0, P No "R" on the pcm line, so that is probably the problem : my soundcard

I was wondering is it possible to use the ices client with more that one soundcard under Centos Linux? I only want to stream one mono audio feed to icecast, but am already using the Line input of my sound card for something else, and the mic input might be far to sensitive. TIA Keith ----------------------------------------------------------- Websites: http://www.karsites.net

Jean Marc, Ok so I tested with the new code, same result- frame size= 160, filter length=160 ms/1280 samples, 8000 Hz diverged after 9564 calls/packets or 191 seconds. My system stays relatively in synch but is not perfect- difficult to measure if there is any clock drift, but it probably is. In order to make it break faster I use an open air usb microphone that is part of a logitech notebook