Mark Constable wrote:
Steve Harris wrote:
On Fri, Mar 04, 2005 at 08:10:51 +1000, Mark
Using it for interviews for podcasting is also
almost mandatory requirement but until someone works
out how to record both ends of the conversation at
the same time it's hardly worth the effort. It's the
same issue for doing live jams... or at least would
be all much easier to manage if using AND recording
both sides of the discussion/jam, all at the same
time, was feasible.
You can do it with JACK, just mix in each stage of the jam, and
the mix the next studio:
studio A ----> voip ---> mix+voip ---> studio C (endpoint + record)
Thanks but in the case of a simple two ended voice comms,
where one end is me with an SBlive, the other end only gets
an echo of their own voice and cannot hear me. I can hear
them, and their echo, just fine.
The "other end" in my case are professional people who do
not use linux and the perception on follow-up landline calls
is that when they know I am using linux on my end the response
is "hey pal, get a real computer and don't waste my time".
Which leaves me in an embarrasing position and also with no
one on the other end patient enough to give me feedback when
I try to tweak alsamixer to find the right comdination.
Sure I could use 2 computers, 2 headsets and 4 ears to try
and work out the right alsamixer combo but I still have to
deal with the same clueless person to try and work thru all
the combinations on both ends.
this is with skype? then it's definitely yr alsamixer settings ... when
i've been recording and forget to change my mixer settings back to
"normal", i sometimes get problems like this on skype ... also - you
must make sure skype is the *only* app using bandwidth, otherwise you do
To Mario, yes I know skype is "evil" like
mp3 but there are
too many "buts" already. iPods won't play ogg content and
most VoIP users do not run linux. It's a sad thing on both
counts but that's life so if I want to interact with "most
people" I have to bend my own rules... or not play at all.
yeah ... in life, there are no absolutes :) i'm sure, though, that
skype has the potential - in time - to be open-sourced. think about it -
the *only* thing that matters to something like VoIP is market-share ...
the more people that use *your* application, the more likely *other*
people are to use it - there's no use joining a network that has no one
to talk to ;) i reckon (maybe with some pressure, heh heh) that once
skype has cemented its market share, then it can only benefit, and
benefit *big*, by going open source ...
that is why it's a bit silly to avoid discussion of it.
As for real-time jamming, I can't imagine it being
in any useful way other than proofing tracks... as in side
A plays composition, side B records it then adds extra
content while side A then records that combination, then
after that is recorded, said A adds their bit to the mix,
which side B then records, and on it goes ping-pong fashion.
yeah, this would at least be something .... i was checking out the
forums, and it doesn't look like full-duplex recording will be an option
until they release an alsa-enabled version (which could be soon).
Is their a kind and patient LAU user out there
who would be interested in trying some skype tests ?
sure ... email me off-list, and we'll arrange it.