2 Mar
2005
2 Mar
'05
5:26 p.m.
Hej!
how can I record the output of the RealPlayer?
alsa - ecasound?
(I did not get working realplayer with jack)
thanks for help
Jürgen
2 Mar
2 Mar
5:49 p.m.
Jürgen Schöneberg wrote:
Hej!
how can I record the output of the RealPlayer?
alsa - ecasound?
(I did not get working realplayer with jack)
thanks for help
Jürgen
I use mplayer
mplayer -dumpstream -dumpfile filename url
if you have the .ram url use
mplayer -playlist -dumpstream -dumpfile filename url
Bails
3 Mar
3 Mar
4:48 p.m.
Robert Bailey wrote:
Jürgen Schöneberg wrote:
Hi.
Thanks for discussing this problem; I've been struggling with it for
quite a while.
When I try the dumpstream approach mentioned above, using
mplayer -dumpstream -dumpfile filename -playlist url
(since -playlist (at least for me) seems to want an argument), I do get
a local file with the contents of the stream, but it is still in some
sort of Real-specific format, so that I need to use realplayer to listen
to it. And, so, the original problem recurs: how do I then convert this
cached stream to .wav format? I went to the mplayer site and downloaded
the Real-Video(/Audio?) related tgz, and tried to install it in all the
reasonable places I could think of, but got no happiness. I'm clearly
missing something, or there must be another approach to this problem.
It's almost as if there must be some way of interposing an ALSA-aware
(and re-directable) loopback device for /dev/dsp. I can almost taste
it, it's so close. But my poor brain sizzles before I get the taste of
the solution. (or something).
Any insights?
/ken
/ken
Hej!
how can I record the output of the RealPlayer?
alsa - ecasound?
(I did not get working realplayer with jack)
thanks for help
Jürgen
I use mplayer
mplayer -dumpstream -dumpfile filename url
if you have the .ram url use
mplayer -playlist -dumpstream -dumpfile filename url
Bails
7:21 p.m.
Ken Dawson wrote:
...
It's almost as if there must be some way of interposing an ALSA-aware
(and re-directable) loopback device for /dev/dsp. I can almost taste
it, it's so close. But my poor brain sizzles before I get the taste of
the solution. (or something).
Another similar issue is correctly capturing both the
incoming and outgoing streams of skype on the same machine.
Any insights?
http://www.zorg.org/vsound/
--markc
9:54 p.m.
New subject: [linux-audio-user]Skype and internet jamming [was Howto record PCM (from realplayer)?]
Mark Constable wrote:
Ken Dawson wrote:
just taking this a bit further - has anyone experimented with
live-jamming over skype?
i've done a couple of tests, but only one-way so far (i've got my mate's
bass at the moment, so we haven't tried live jamming). i'm pretty sure,
though, that latency would be the killer - i do some stuff with the
local community radio station, and me and my mate found a pretty solid
hardware codec that wasn't being used - it plugged into any phone line
and would send whatever you put into it (there were two line inputs and
two microphone inputs) through the phone lines to the decoder at the
station.
when we did live broadcasts from a local club, though, i would monitor
from the club side what was being fed back from the station's broadcast
(just to make sure it didn't drop out) - but with the headphones on,
listening to the band up on stage being broadcast back out to me, there
was at least a half-beat delay (mind you, i couldn't take the headphones
off, because this delay turned what was an average reggae band onstage
into a dubbed-out head-freak).
even still, perhaps in an electronic music environment, a delay like
this would be manageable via quantization or something?
anyway, just because of the audio quality of skype, and the ability to
access all areas of the globe, i thought this would be an interesting
area worthy of research and experimentation.
any thoughts?
shayne
...
It's almost as if there must be some way of interposing an ALSA-aware
(and re-directable) loopback device for /dev/dsp. I can almost taste
it, it's so close. But my poor brain sizzles before I get the taste
of the solution. (or something).
Another similar issue is correctly capturing both the
incoming and outgoing streams of skype on the same machine. 
11:09 p.m.
New subject: [linux-audio-user]Skype and internet jamming [was Howto record PCM (from realplayer)?]
--- Shayne O'Connor <forums(a)machinehasnoagenda.com>
wrote:
Mark Constable wrote:
the
No clue about skype. I gotta look that up. On a
related note I do have oddcastv3-jack-3.0.3 working on
my local lan so I can stream the master L,R channels
of my mixing consol over a network to xmms or whatever
client application. It's definitely not going to cut
it for live jams but it's great for letting clients
listen to what I'm producing or mixing for them.
Having a streaming jack icecast/oddcast server is
awesome.
I just reconfigured my network to enable port
forwarding so it's time to test the oddcast stream
over the internet. For the first time in several years
I actually have a motivation to tolerate network
administration.
ron
Ken Dawson wrote:
> ...
> It's almost as if there must be some way of
interposing an ALSA-aware
> (and re-directable) loopback device for
/dev/dsp.
I can almost taste
> it, it's so close. But my poor brain
sizzles
before I get the taste
of the
solution. (or something).
Another similar issue is correctly capturing both incoming and outgoing streams of skype on the
same
machine.
just taking this a bit further - has anyone
experimented with
live-jamming over skype?
i've done a couple of tests, but only one-way so far
(i've got my mate's
bass at the moment, so we haven't tried live
jamming). i'm pretty sure,
though, that latency would be the killer - i do some
stuff with the
local community radio station, and me and my mate
found a pretty solid
hardware codec that wasn't being used - it plugged
into any phone line
and would send whatever you put into it (there were
two line inputs and
two microphone inputs) through the phone lines to
the decoder at the
station.
when we did live broadcasts from a local club,
though, i would monitor
from the club side what was being fed back from the
station's broadcast
(just to make sure it didn't drop out) - but with
the headphones on,
listening to the band up on stage being broadcast
back out to me, there
was at least a half-beat delay (mind you, i couldn't
take the headphones
off, because this delay turned what was an average
reggae band onstage
into a dubbed-out head-freak).
even still, perhaps in an electronic music
environment, a delay like
this would be manageable via quantization or
something?
anyway, just because of the audio quality of skype,
and the ability to
access all areas of the globe, i thought this would
be an interesting
area worthy of research and experimentation.
any thoughts? shayne
__________________________________
Celebrate Yahoo!'s 10th Birthday!
Yahoo! Netrospective: 100 Moments of the Web
http://birthday.yahoo.com/netrospective/
11:19 p.m.
New subject: [linux-audio-user]Skype and internet jamming [was Howto record PCM (from realplayer)?]
On Friday 04 March 2005 14:09, R Parker wrote:
over the internet. For the first time in several
years
I actually have a motivation to tolerate network
administration.
Nooooo, don't do it man, it's evil, EVIL I tells ya!
--
Dan Harper
http://danharper.org/linuxdesktopblog
11:26 p.m.
New subject: [linux-audio-user]Skype and internet jamming [was Howto record PCM (from realplayer)?]
--- Dan Harper <tech(a)danharper.org> wrote:
On Friday 04 March 2005 14:09, R Parker wrote:
You're right and I spoke to soon. The DSL router isn't
forwarding 8000. Do I fix it or work on what I'm
supposed to be doing....hmmm, must fix this network.
over the internet. For the first time in several
years
I actually have a motivation to tolerate network
administration.
Nooooo, don't do it man, it's evil, EVIL I tells ya! --
Dan Harper
http://danharper.org/linuxdesktopblog
__________________________________
Celebrate Yahoo!'s 10th Birthday!
Yahoo! Netrospective: 100 Moments of the Web
http://birthday.yahoo.com/netrospective/
4 Mar
4 Mar
5:32 a.m.
New subject: [linux-audio-user]Skype and internet jamming [was Howto record PCM (from realplayer)?]
Please use Skype to your hearts content!! :)
Why...the traffic created is "on demand" over our backbone (Level 3) so it
means more billing for our company! Jam on!!
R~
--- Shayne O'Connor <forums(a)machinehasnoagenda.com>
wrote:
Mark Constable wrote:
the
No clue about skype. I gotta look that up. On a
related note I do have oddcastv3-jack-3.0.3 working on
my local lan so I can stream the master L,R channels
of my mixing consol over a network to xmms or whatever
client application. It's definitely not going to cut
it for live jams but it's great for letting clients
listen to what I'm producing or mixing for them.
Having a streaming jack icecast/oddcast server is
awesome.
I just reconfigured my network to enable port
forwarding so it's time to test the oddcast stream
over the internet. For the first time in several years
I actually have a motivation to tolerate network
administration.
ron
Ken Dawson wrote:
> ...
> It's almost as if there must be some way of
interposing an ALSA-aware
> (and re-directable) loopback device for
/dev/dsp.
I can almost taste
> it, it's so close. But my poor brain
sizzles
before I get the taste
of the
solution. (or something).
Another similar issue is correctly capturing both incoming and outgoing streams of skype on the
same
machine.
just taking this a bit further - has anyone
experimented with
live-jamming over skype?
i've done a couple of tests, but only one-way so far
(i've got my mate's
bass at the moment, so we haven't tried live
jamming). i'm pretty sure,
though, that latency would be the killer - i do some
stuff with the
local community radio station, and me and my mate
found a pretty solid
hardware codec that wasn't being used - it plugged
into any phone line
and would send whatever you put into it (there were
two line inputs and
two microphone inputs) through the phone lines to
the decoder at the
station.
when we did live broadcasts from a local club,
though, i would monitor
from the club side what was being fed back from the
station's broadcast
(just to make sure it didn't drop out) - but with
the headphones on,
listening to the band up on stage being broadcast
back out to me, there
was at least a half-beat delay (mind you, i couldn't
take the headphones
off, because this delay turned what was an average
reggae band onstage
into a dubbed-out head-freak).
even still, perhaps in an electronic music
environment, a delay like
this would be manageable via quantization or
something?
anyway, just because of the audio quality of skype,
and the ability to
access all areas of the globe, i thought this would
be an interesting
area worthy of research and experimentation.
any thoughts? shayne
__________________________________
Celebrate Yahoo!'s 10th Birthday!
Yahoo! Netrospective: 100 Moments of the Web
http://birthday.yahoo.com/netrospective/
6:11 a.m.
New subject: [linux-audio-user] Skype and internet jamming/podcasting
hanaghan(a)starband.net wrote:
Please use Skype to your hearts content!! :)
Why... the traffic created is "on demand" over
our backbone (Level 3) so it means more billing
for our company! Jam on!!
Using it for interviews for podcasting is also an
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.
Anyone ? Please make some notes here...
http://alsa.opensrc.org/?page=UsingSkype
Mark Constable wrote:
> Another similar issue is correctly capturing
> both the incoming and outgoing streams of skype
> on the same machine.
I'm happy to do some skype testing if someone is
needed on the other end. Just email me privately.
This could be of interest if anyone is considering
dabbling with skype and not seen this before..
http://internet.newsforge.com/article.pl?sid=05/02/25/2148229
--markc
2:39 p.m.
New subject: [linux-audio-user] Skype and internet jamming/podcasting
On Fri, Mar 04, 2005 at 08:10:51 +1000, Mark Constable wrote:
hanaghan(a)starband.net wrote:
You can do it with JACK, just mix in each stage of the jam, and forward on
the mix the next studio:
studio A ----> voip ---> mix+voip ---> studio C (endpoint + record)
(source) ^
|
studio B
- Steve
Please use Skype to your hearts content!! :)
Why... the traffic created is "on demand" over
our backbone (Level 3) so it means more billing
for our company! Jam on!!
Using it for interviews for podcasting is also an
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.
9:35 p.m.
New subject: [linux-audio-user] Skype and internet jamming/podcasting
Steve Harris wrote:
On Fri, Mar 04, 2005 at 08:10:51 +1000, Mark Constable
wrote:
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.
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.
As for real-time jamming, I can't imagine it being feasible
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.
Is their a kind and patient LAU user out there somewhere
who would be interested in trying some skype tests ?
--markc
Using it for interviews for podcasting is also an
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 forward on
the mix the next studio:
studio A ----> voip ---> mix+voip ---> studio C (endpoint + record)
(source) ^
|
studio B 
9:50 p.m.
New subject: [linux-audio-user] Skype and internet jamming/podcasting
On Sat, 2005-03-05 at 11:35 +1000, Mark Constable wrote:
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".
You aren't using ALSA CVS with the multichannel device to do this, are
you? If so you need to update to the latest CVS, there was a serious
bug that was just recently fixed.
Lee
10:35 p.m.
New subject: [linux-audio-user] Skype and internet jamming/podcasting
Lee Revell wrote:
On Sat, 2005-03-05 at 11:35 +1000, Mark Constable
wrote:
i know the "efx_voices_mask" patch has been applied to cvs (btw - this
cures one of the long-standing bugs i had with the patch: it was sending
the "mix" of all channels to the first and second "capture" channels
plus a few others ... now i get: livedrive on 11+12, sblive ac97 on 1+2
... nice and clean), but has the one to do with playback pops+clicks
been applied yet?
shayne
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".
You aren't using ALSA CVS with the multichannel device to do this, are
you? If so you need to update to the latest CVS, there was a serious
bug that was just recently fixed.
Lee
10:55 p.m.
New subject: [linux-audio-user] Skype and internet jamming/podcasting
On Sat, 2005-03-05 at 13:34 +1100, Shayne O'Connor wrote:
Lee Revell wrote:
Yup, sorry. This was really a stupid bug. I should have tested it with
an SBLive sooner.
On Sat, 2005-03-05 at 11:35 +1000, Mark Constable
wrote:
i know the "efx_voices_mask" patch has been applied to cvs (btw - this
cures one of the long-standing bugs i had with the patch: it was sending
the "mix" of all channels to the first and second "capture" channels
plus a few others ... now i get: livedrive on 11+12, sblive ac97 on 1+2
... nice and clean) 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".
You aren't using ALSA CVS with the multichannel device to do this, are
you? If so you need to update to the latest CVS, there was a serious
bug that was just recently fixed.
Lee
, but has the one to do with playback pops+clicks
been applied yet?
No, the original patch did not fix that problem 100%, but the author has
sent me an updated patch that claims to fix it. I have not tested it
yet.
Lee
10:45 p.m.
New subject: [linux-audio-user] Skype and internet jamming/podcasting
Mark Constable wrote:
Steve Harris wrote:
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
get echoes.
On Fri, Mar 04, 2005 at 08:10:51 +1000, Mark
Constable wrote:
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.
Using it for interviews for podcasting is also
an
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
forward on
the mix the next studio:
studio A ----> voip ---> mix+voip ---> studio C (endpoint + record)
(source) ^
|
studio B 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
feasible
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
somewhere
who would be interested in trying some skype tests ?
sure ... email me off-list, and we'll arrange it.
shayne
11:30 p.m.
New subject: [linux-audio-user] Skype and internet jamming/podcasting
Shayne O'Connor wrote:
Yes.
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.
this is with skype? then it's definitely yr alsamixer settings ...
Yes, I do supect so... my problem is "which 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
get echoes.
I'll come out and opening admit to being incredibly dumb
so it took me DAYS to figure how to simply record into
the mic input of an SBLive, then record mic and playing
wav file with aplay at the same time, and then record
mic and midi/wavetable at the same time. I still haven't
managed to do all 3 at the same time.
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 ...
I would respectfully disagree with your comments. 9/10s
of the computer users on this planet are probably not
even aware of what "open source" actually means... all
most of these people care about is a) is it free and b)
does it install without any problems. skype is already
the market leader in this field, they just need to stay
in front and whether they open source their crown jewels
to give us GNU zealots a warm and fuzzy feeling is...
debatable.
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).
Does that mean it is actually not at all possible to use
voice both ways AND record both "channels" at the same
time anyway ?
--markc
11 Apr
11 Apr
12:32 a.m.
New subject: [linux-audio-user] Skype and internet jamming/podcasting
just saw this on slashdot, thought it might be of interest to those
involved in this thread:
shashark writes "Technologically savvy users are merging these
technologies to "Skypecast", using Skype's service to distribute
recordings across the internet for free. This allows expert users to run
their own mini-radio stations, which can be accessed by any Skype user.
Skype does not actively support these uses, but encourages its users to
find new applications for their service. Other possibilities discussed
by Skypecasters at Unbound Spiral or Moodle are to turn an MP3 player
into a radio station for any of Skype's 29 million registered users to
dial up using their Skype line. Instructions also are available on how
to record a personal soap opera and use Skype to distribute it en masse.
Even more ominously, some Skypecasters record Skype calls and post them
on the Internet."
shayne
4 Mar
4 Mar
12:17 a.m.
New subject: [linux-audio-user]Skype and internet jamming [was Howto record PCM (from realplayer)?]
On Fri, Mar 04, 2005 at 12:54:23 +1100, Shayne O'Connor wrote:
Mark Constable wrote:
As long as your far enough apart that you cant hear each other (thats the
point, right ;) I dont think the delay is much of a problem, the latency
on skype is pretty good. I think youre more likly to hit quality
problems, its pretty good for voice, but probably not for music.
- Steve
Ken Dawson wrote:
just taking this a bit further - has anyone experimented with
live-jamming over skype?
i've done a couple of tests, but only one-way so far (i've got my mate's
bass at the moment, so we haven't tried live jamming). i'm pretty sure,
though, that latency would be the killer - i do some stuff with the
local community radio station, and me and my mate found a pretty solid
hardware codec that wasn't being used - it plugged into any phone line
and would send whatever you put into it (there were two line inputs and
two microphone inputs) through the phone lines to the decoder at the
station.
when we did live broadcasts from a local club, though, i would monitor
from the club side what was being fed back from the station's broadcast
(just to make sure it didn't drop out) - but with the headphones on,
listening to the band up on stage being broadcast back out to me, there
was at least a half-beat delay (mind you, i couldn't take the headphones
off, because this delay turned what was an average reggae band onstage
into a dubbed-out head-freak). ...
It's almost as if there must be some way of interposing an ALSA-aware
(and re-directable) loopback device for /dev/dsp. I can almost taste
it, it's so close. But my poor brain sizzles before I get the taste
of the solution. (or something).
Another similar issue is correctly capturing both the
incoming and outgoing streams of skype on the same machine.
2:28 a.m.
New subject: [linux-audio-user]Skype and internet jamming [was Howto record PCM (from realplayer)?]
Steve Harris wrote:
On Fri, Mar 04, 2005 at 12:54:23 +1100, Shayne
O'Connor wrote:
actually, the quality of music broadcast through skype (setting the
"mix" channel of a sblive to be captured) seems to be just as good as if
you were hearing it locally ...
i'm gonna try and have a jam with a buddy over the weekend and see how
it goes ...
shayne
Mark Constable wrote:
As long as your far enough apart that you cant hear each other (thats the
point, right ;) I dont think the delay is much of a problem, the latency
on skype is pretty good. I think youre more likly to hit quality
problems, its pretty good for voice, but probably not for music.
Ken Dawson wrote:
just taking this a bit further - has anyone experimented with
live-jamming over skype?
i've done a couple of tests, but only one-way so far (i've got my mate's
bass at the moment, so we haven't tried live jamming). i'm pretty sure,
though, that latency would be the killer - i do some stuff with the
local community radio station, and me and my mate found a pretty solid
hardware codec that wasn't being used - it plugged into any phone line
and would send whatever you put into it (there were two line inputs and
two microphone inputs) through the phone lines to the decoder at the
station.
when we did live broadcasts from a local club, though, i would monitor
from the club side what was being fed back from the station's broadcast
(just to make sure it didn't drop out) - but with the headphones on,
listening to the band up on stage being broadcast back out to me, there
was at least a half-beat delay (mind you, i couldn't take the headphones
off, because this delay turned what was an average reggae band onstage
into a dubbed-out head-freak).
...
It's almost as if there must be some way of interposing an ALSA-aware
(and re-directable) loopback device for /dev/dsp. I can almost taste
it, it's so close. But my poor brain sizzles before I get the taste
of the solution. (or something).
Another similar issue is correctly capturing both the
incoming and outgoing streams of skype on the same machine.
2:34 p.m.
New subject: [linux-audio-user]Skype and internet jamming [was Howto record PCM (from realplayer)?]
On Fri, Mar 04, 2005 at 05:28:25 +1100, Shayne O'Connor wrote:
actually, the quality of music broadcast through skype
(setting the
"mix" channel of a sblive to be captured) seems to be just as good as if
you were hearing it locally ...
Hmm.. cool :) I had no idea what codec they were using, but as it sounds
so good for voice, I assumed it was voice optimised (like ogg speex).
According to the website it uses up to 128 kbps if you have the bandwidth.
For those people who are shapening thier free-software knives :) you could
also use gnome-meeting, which lets you pick appropriate codecs, and is
free as in speech.
- Steve
2:47 p.m.
New subject: [linux-audio-user]Skype and internet jamming [was Howto record PCM (from realplayer)?]
On Fri, Mar 04, 2005 at 06:34:22PM +0000, Steve Harris wrote:
On Fri, Mar 04, 2005 at 05:28:25 +1100, Shayne
O'Connor wrote:
Skype scares me in the same way that mp3 should have scared me before I
knew fraunhofer was not my friend.
--
Eric Dantan Rzewnicki | Systems Administrator
Technical Operations Division | Radio Free Asia
2025 M Street, NW | Washington, DC 20036 | 202-530-4900
CONFIDENTIAL COMMUNICATION
This e-mail message is intended only for the use of the addressee and
may contain information that is privileged and confidential. Any
unauthorized dissemination, distribution, or copying is strictly
prohibited. If you receive this transmission in error, please contact
network(a)rfa.org.
actually, the quality of music broadcast through
skype (setting the
"mix" channel of a sblive to be captured) seems to be just as good as if
you were hearing it locally ...
Hmm.. cool :) I had no idea what codec they were
using, but as it sounds
so good for voice, I assumed it was voice optimised (like ogg speex).
According to the website it uses up to 128 kbps if you have the bandwidth.
For those people who are shapening thier free-software knives :) you could
also use gnome-meeting, which lets you pick appropriate codecs, and is
free as in speech. 
2:59 p.m.
New subject: [linux-audio-user]Skype and internet jamming
Eric Dantan Rzewnicki <rzewnickie(a)rfa.org> writes:
On Fri, Mar 04, 2005 at 06:34:22PM +0000, Steve Harris
wrote:
And its inaccessible on Linux. Another lock-out thing.
--
CYa,
Mario
On Fri, Mar 04, 2005 at 05:28:25 +1100, Shayne
O'Connor wrote:
Skype scares me in the same way that mp3 should have scared me before I
knew fraunhofer was not my friend. actually, the quality of music broadcast through
skype (setting the
"mix" channel of a sblive to be captured) seems to be just as good as if
you were hearing it locally ...
Hmm.. cool :) I had no idea what codec they were
using, but as it sounds
so good for voice, I assumed it was voice optimised (like ogg speex).
According to the website it uses up to 128 kbps if you have the bandwidth.
For those people who are shapening thier free-software knives :) you could
also use gnome-meeting, which lets you pick appropriate codecs, and is
free as in speech.
3:47 p.m.
New subject: [linux-audio-user]Skype and internet jamming
On Fri, Mar 04, 2005 at 07:59:05PM +0100, Mario Lang wrote:
Eric Dantan Rzewnicki <rzewnickie(a)rfa.org>
writes:
No its not, I use skype on linux. The open SIP stuff is probably better
though, and thier are free providers now, I'm just looking at
http://www.stanaphone.com/, which flogs hardware, but lets you use any
compliant SIP implementation IIUC. Not tried it, but a collegue uses it.
- Steve
On Fri, Mar 04, 2005 at 06:34:22PM +0000, Steve
Harris wrote:
And its inaccessible on Linux. Another lock-out thing. On Fri, Mar 04, 2005 at 05:28:25 +1100, Shayne
O'Connor wrote:
Skype scares me in the same way that mp3 should have scared me before I
knew fraunhofer was not my friend. actually, the quality of music broadcast through
skype (setting the
"mix" channel of a sblive to be captured) seems to be just as good as if
you were hearing it locally ...
Hmm.. cool :) I had no idea what codec they were
using, but as it sounds
so good for voice, I assumed it was voice optimised (like ogg speex).
According to the website it uses up to 128 kbps if you have the bandwidth.
For those people who are shapening thier free-software knives :) you could
also use gnome-meeting, which lets you pick appropriate codecs, and is
free as in speech.
5 Mar
5 Mar
10:31 a.m.
New subject: [linux-audio-user]Skype and internet jamming
Steve Harris <S.W.Harris(a)ecs.soton.ac.uk> writes:
On Fri, Mar 04, 2005 at 07:59:05PM +0100, Mario Lang
wrote:
I should have been more clear, I ment, the version of Skype for Linux
is not accessible in the sense of computer access for the blind.
And since its closed-source, we can not fix this ourselves.
--
CYa,
Mario
Eric Dantan Rzewnicki <rzewnickie(a)rfa.org>
writes:
No its not, I use skype on linux. On Fri, Mar 04, 2005 at 06:34:22PM +0000, Steve
Harris wrote:
And its inaccessible on Linux. Another lock-out thing. On Fri, Mar 04, 2005 at 05:28:25 +1100, Shayne
O'Connor wrote:
> actually, the quality of music broadcast through skype (setting the
> "mix" channel of a sblive to be captured) seems to be just as good as if
> you were hearing it locally ...
Hmm.. cool :) I had no idea what codec they were using, but as it sounds
so good for voice, I assumed it was voice optimised (like ogg speex).
According to the website it uses up to 128 kbps if you have the bandwidth.
For those people who are shapening thier free-software knives :) you could
also use gnome-meeting, which lets you pick appropriate codecs, and is
free as in speech.
Skype scares me in the same way that mp3 should have scared me before I
knew fraunhofer was not my friend. 
12 Apr
12 Apr
4:49 a.m.
New subject: [linux-audio-user]Skype and internet jamming
"Shayne O'Connor" <forums(a)machinehasnoagenda.com> writes:
has anyone experimented with live-jamming
Yes, I do it several hours every night over the iax protocol. The
remote party is using windows with the libre iaxcomm client and I'm
using asterisk.
The voice round trip delay is constant 600ms while the ping is 38ms
(measured using a recording of a session). The other party is located
20km from me. The connection is rock stable. The longest I've tested
was 6h.
I'm struggling to find a sip/iax voip application that lets me hook
into jack, using either oss2jack, jackplug or native, but still no
luck.
Proprietary software is not an option.
If anyone wants to jam with me or test their voip connection to
Norway, please call me using the urls in my signature.
--
Esben Stien is b0ef(a)esben-stien.name
http://www.
irc://irc. /%23contact
[sip|iax]:
jid:b0ef@
3 Mar
3 Mar
8:49 p.m.
When I try the dumpstream approach mentioned above,
using
mplayer -dumpstream -dumpfile filename -playlist url
(since -playlist (at least for me) seems to want an argument), I do
get a local file with the contents of the stream, but it is still in
some sort of Real-specific format, so that I need to use realplayer
to listen to it. And, so, the original problem recurs: how do I then
convert this cached stream to .wav format? I went to the mplayer
If you have mplayer, jack and bio2jack you can say
$ mplayer -ao jack localfile or url
and capture with some jack recorder like timemachine.
Wolfgang
4 Mar
4 Mar
10:38 a.m.
On Fri, 4 Mar 2005 01:48:39 +0100, Wolfgang Woehl <tito(a)rumford.de> wrote:
"mplayer -ao pcm -aofile foo.wav foo.rm" should also work
cheers,
Christian
When I try the
dumpstream approach mentioned above, using
mplayer -dumpstream -dumpfile filename -playlist url
(since -playlist (at least for me) seems to want an argument), I do
get a local file with the contents of the stream, but it is still in
some sort of Real-specific format, so that I need to use realplayer
to listen to it. And, so, the original problem recurs: how do I then
convert this cached stream to .wav format? I went to the mplayer
If you have mplayer, jack and bio2jack you can say
$ mplayer -ao jack localfile or url
and capture with some jack recorder like timemachine.
5 Mar
5 Mar
10:33 a.m.
Christian Henz <christian.henz(a)gmail.com> writes:
On Fri, 4 Mar 2005 01:48:39 +0100, Wolfgang Woehl
<tito(a)rumford.de> wrote:
"mplayer -ao pcm -aofile foo.wav foo.rm" should also work
Yes, I used this to convert problematic formats to ogg :-)
Very helpful. If you add "-vc dummy -vo null" you can even extract audio
streams from videos pretty easily and fast.
--
CYa,
Mario
When I
try the dumpstream approach mentioned above, using
mplayer -dumpstream -dumpfile filename -playlist url
(since -playlist (at least for me) seems to want an argument), I do
get a local file with the contents of the stream, but it is still in
some sort of Real-specific format, so that I need to use realplayer
to listen to it. And, so, the original problem recurs: how do I then
convert this cached stream to .wav format? I went to the mplayer
If you have mplayer, jack and bio2jack you can say
$ mplayer -ao jack localfile or url
and capture with some jack recorder like timemachine.
7115
days inactive
7156
days old
linux-audio-user@lists.linuxaudio.org
29 comments
16 participants
participants (16)
-
Bengt Gördén -
Christian Henz -
Dan Harper -
Eric Dantan Rzewnicki -
Esben Stien -
hanaghan＠starband.net -
Jürgen Schöneberg -
Ken Dawson -
Lee Revell -
Mario Lang -
Mark Constable -
R Parker -
Robert Bailey -
Shayne O'Connor -
Steve Harris -
Wolfgang Woehl