hey Dann
yup its a stuggle
my romote hosts are on skype
and it seems to do a pretty good job of holding the levels good
i have run skype and Team speak at the same time on seperate channels
and that worked very well also
its funny i do things a little different than most
all my audio sources are fed to the mixer seperatly except for the skype line
i run 3 xmms at the same time
1. for music we play lots of that
2. for promos and sound bites and impromptu smart ass sound bits
3. this one is for quick stream checks and such
all on seperate cards
all those lines run to the mixer
and another card for skype or team speak which ever we are using
which also goes to the mixer
the fith card was just a dream that works fine now
but i found i neeed it in the 64bit sled10 box
as the 5th card really didnt have alot of use in the primary machine
all of the mixed audio goes out the main bus to the Dynebolic box
for streaming and recording
so i dont use audacity for recording just for post production
i run 2 audio streams out of here full time using Muse on the dyne box with a icecast server and darkice as a source to the same server from another box
thats why i like those dumpster dive computers put DSL on them and let them play music
but i can see how the asterisk box/skype can cause problems with levels
i still dont think that this voip stuff is ready for primtime yet
have a Woody's Wheat and relax ahhhhh
For the first time the quality was just too bad to listen to
Moderators: snarkout, Patrick, dann
Update
I finally got down to playing with ardour and qjackctl. Wow is this program a monster especially since I don't have much recording experience. Anyway I was able to record a bass track going thru a direct box. Once I played around with qjackctl a bit I started understanding how to route audio thru the M-Audio card into Ardour. I eventually got it working. Today I'm going to pick up some audio cables for my mixer and start recording multiple tracks at once. More than likely it will be just a conversation between myself and someone else. I'm pretty psyched! I just printed out the Ardour manual and will read it thoroughly.
Ego contemno licentia
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elgordo123
- Posts: 47
- Joined: Wed Nov 02, 2005 5:46 pm
I miss the old days when there wasn't interviews every single show. I find it very informative when you guys talk about your experiences with different programs, etc. Now the only that talks is Pat. No one else really even needs to be there anymore because they never say anything.
Maybe when you are having problems like this you can actually talk amongst yourselves like the good ol' days and reschedule the guest?
Just a thought!
Maybe when you are having problems like this you can actually talk amongst yourselves like the good ol' days and reschedule the guest?
Just a thought!
- Wally Balljacker
- Posts: 1227
- Joined: Fri Jul 29, 2005 3:32 am
- Location: University of Massachusetts - Lowell
- Contact:
I agree, the old days where all four of you were in the same room just blabbering on about Linux, and open source was much more enjoyable than listening to Pat, or Dann talk to a guest for an hour, while everyone else is silent.elgordo123 wrote:I miss the old days when there wasn't interviews every single show. I find it very informative when you guys talk about your experiences with different programs, etc. Now the only that talks is Pat. No one else really even needs to be there anymore because they never say anything.
Maybe when you are having problems like this you can actually talk amongst yourselves like the good ol' days and reschedule the guest?
Just a thought!
You guys post so much interesting stuff on your blogs, and it seems like you never get to talk about it on the show, due to long guest interviews, and technical issues.
Are you running all channels through the mixer, then mixing down to 2 channels and recording that?dann wrote:Invariably there seems to be one person who's audio is lower than everyone else. In the post production trying to get this person's levels up to without distorting the rest of the audio is painful.
Any mixer you should have at least 2 stages of gain per channel - an input trim which matches the mic to the board, then a channel level that sets volume. Usually that gives you at least 100 dB of gain, which should be enough to boost anything up to a decent level.
Of course this assumes you're able to get the trim and volume level right at the beginning of the recording. If you're trying to correct an imbalance in post-production, good luck. The best you can do is compress it a lot and hope it works.