From: Atte:
>
> Hi
>
> Trying my luck again with recordmydesktop. Found that the following
> eventually starts record my desktop:
>
> https://dl.dropboxusercontent.com/u/4343030/recordmydesktop
>
> However it seems to only record in mono. How do I capture two audio
> channels?
>
> Even better: is there an alternative out there that actually works?
>
>
I have been using ffmpeg for my recordings.
It's a bit tricky to get sound from jack though, but if you start
pulseaudio using
the jack sinks, you get the sound if you connect the output
of what you want to record to the pulseaudio jack input. I do
that connection manually.
So:
1. Start pulseaudio (make sure it is configured to use jack):
$ pulseaudio
2. Connect the output jack ports of what you want to record into pulseaudio.
3. Run something like this:
$ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 720x646 -i :0.0 -acodec
pcm_s16le -vcodec libx264 -preset ultrafast -crf 0 -y output.mkv
(this works for me)
Here are a couple of links with info I used to configure ffmpeg:
http://www.commandlinefu.com/commands/view/148/capture-video-of-a-linux-des…http://www.commandlinefu.com/commands/view/7109/capture-video-of-a-linux-de…
And here is my version of ffmpeg, which works:
ffmpeg version 0.10.7 Copyright (c) 2000-2013 the FFmpeg developers
The ffmpeg version might be relevant since I think there are two different
versions of ffmpeg,
one hijacked version made by the libav folks, and one real version.
Hello,
I have a situation in mind : in a LAN a central computer running jack server with a lot of audio I/O, and lightweight clients with stereo I/O.
I'd like those clients to be able to send a receive 2 channels of audio to the central jack server, lowest latency possible.
With the zita-njbridge and jack server running on the lightweight clients, it works, but considering i don't need the jack server on the lightweight clients (no effect plugins, no routing, no synths...) could a zita-nabridge exist to capture/override the streams from the client's embedded alsa device ?
Raphaël
http://www.jerashmusic.fr
Hey hey everyone,
I hreard, that the Bash (Bourne Again shell) had a vital security issue, that
was only fixed very recently. So if you rely on Bash better update. I _THINK_
the problem was only fixed last week or so. Let your friends know! :)
Don't ask me about specifics, I just got the info and passed it along, since
it sounded like good advice.
Thanks and sorry for the OT.
Ta-ta
----
Ffanci
* Internet: http://freeshell.de/~silvain
Hi all,
I'm the team leader of Voice Communication Systems development in ARTISYS company. We try to utilize RME RayDat sound device on Linux system with ALSA sound interface. After a lot of experiments we are not able to set up proper sound output from this device, so we kindly ask you for a help to solve our problem. Here is a brief description:
General task of our troubleshooting is that our audio signal output is not continuous. The output signal is rather interrupted. Here is our configuration:
PC with Linux (kernel v. 3.17.0) with ALSA sound libraries compiled in kernel. We have also additional packages installed: alsa-tools, alsa-utils and alsa-lib (all version 1.0.28).
Inside the PC there is PCI Express card RME Raydat, whose drivers are also compiled in the Linux kernel. This interface is connected by optical fibres (ADAT) to the Ferrofish A16 MK-II (2 pairs of TOSLINK cables).
Playing test files using commands
aplay -D pcm.out_test -r 48000 -f S32_LE /usr/share/sounds/alsa/Front_Left.wav -vv
and
aplay -D pcm.out_test2 -r 48000 -f S32_LE /usr/share/sounds/alsa/Front_Right.wav -vv
causes audible interruptions. The configuration of the ALSA device is following:
pcm.out_dmix {
type dmix
ipc_key 56874
ipc_key_add_uid false
ipc_perm 0666
slave {
pcm "hw:2,0"
period_size 2048
channels 36
rate 48000
}
bindings {
0 0 # from 0 => to 0
1 1 # from 1 => to 1
}
}
pcm.out_test {
type plug
slave.pcm "out_dmix"
ttable.0.0 1
}
pcm.out_test2 {
type route
slave.pcm "out_dmix"
ttable.0.1 1
}
Let me describe other two example experiments:
1. If I try to record sound using the same parameters as out_dmix device but of type "dsnoop" (equivalent of dmix for capturing), there are no interruptions and the recorded sound is perfect.
2. If I play single sound file to the output of the RME card while the ALSA device is of type "route", the sound output is perfect. Playing second file on different output channel this way I'm not able to open
other output channel simultaneously by another program because the device is busy. There is this experiment's ALSA device configuration:
pcm.out_test {
type route
slave.pcm "hw:2,0"
slave.format "S32_LE"
slave.channels 36
ttable.0.0 1
}
pcm.out_test2 {
type route
slave.pcm "hw:2,0"
slave.format "S32_LE"
slave.channels 36
ttable.0.1 1
}
This should be solved by "dmix" type ALSA device, which currently causes interruptions in our configuration.
The question is: how can we play multiple audio streams to multiple output channels using our equipment?
Thank you.
Dear Mr. Knoth,
I'm the team leader of Voice Communication Systems development in ARTISYS company. We try to utilize RME RayDat sound device on Linux system with ALSA sound interface. After a lot of experiments we are not able to set up proper sound output from this device, so we kindly ask you for a help to solve our problem. Here is a brief description:
General task of our troubleshooting is that our audio signal output is not continuous. The output signal is rather interrupted. Here is our configuration:
PC with Linux (kernel v. 3.17.0) with ALSA sound libraries compiled in kernel. We have also additional packages installed: alsa-tools, alsa-utils and alsa-lib (all version 1.0.28).
Inside the PC there is PCI Express card RME Raydat, whose drivers are also compiled in the Linux kernel. This interface is connected by optical fibres (ADAT) to the Ferrofish A16 MK-II (2 pairs of TOSLINK cables).
Playing test files using commands
aplay -D pcm.out_test -r 48000 -f S32_LE /usr/share/sounds/alsa/Front_Left.wav -vv
and
aplay -D pcm.out_test2 -r 48000 -f S32_LE /usr/share/sounds/alsa/Front_Right.wav -vv
causes audible interruptions. The configuration of the ALSA device is following:
pcm.out_dmix {
type dmix
ipc_key 56874
ipc_key_add_uid false
ipc_perm 0666
slave {
pcm "hw:2,0"
period_size 2048
channels 36
rate 48000
}
bindings {
0 0 # from 0 => to 0
1 1 # from 1 => to 1
}
}
pcm.out_test {
type plug
slave.pcm "out_dmix"
ttable.0.0 1
}
pcm.out_test2 {
type route
slave.pcm "out_dmix"
ttable.0.1 1
}
Let me describe other two example experiments:
1. If I try to record sound using the same parameters as out_dmix device but of type "dsnoop" (equivalent of dmix for capturing), there are no interruptions and the recorded sound is perfect.
2. If I play single sound file to the output of the RME card while the ALSA device is of type "route", the sound output is perfect. Playing second file on different output channel this way I'm not able to open
other output channel simultaneously by another program because the device is busy. There is this experiment's ALSA device configuration:
pcm.out_test {
type route
slave.pcm "hw:2,0"
slave.format "S32_LE"
slave.channels 36
ttable.0.0 1
}
pcm.out_test2 {
type route
slave.pcm "hw:2,0"
slave.format "S32_LE"
slave.channels 36
ttable.0.1 1
}
This should be solved by "dmix" type ALSA device, which currently causes interruptions in our configuration.
The question is: how can we play multiple audio streams to multiple output channels using our equipment?
Thank you.
Yours sincerely
Ing. Vaclav Mach
Voice Communications System team leader
ARTISYS
www: http://www.artisys.aero
On 22/10/2014 11:45, linux-audio-user-request(a)lists.linuxaudio.org wrote:
> In my experience there's a greater risk of overheating without a fan and
> the ARM (allwinner) chipsets are prone to that. My bet is a low power x86
> processor/unit with a (quiet) fan will out perform and outlast an ARM
> chipset without.
I did some simple benchmarks on a Allwinner A20 board (cubieboard)
recently. The benchmark consist of computing a bunch of sine oscillators
(second order resonator filter), generally used for modal synthesis and
other types of sound synthesis. The results I got from the A20 when
clocked at 1GHz are suprisingly good: 1000 theoretical oscillator can be
computed in a 128 samples period, while on my quad core-i5 I get 1500.
On a 7-years onld Centrino Duo I get about 850. While this don't stand
as a real-world benchmark (buffer transfers are not taken into account)
and I haven't optimized for the architectures (but just let g++ go with
-O2) you get the idea.
I didn't experience overheating on the A20 but the tests are not
continuous as you would during a performance, so I won't bet it will
last long. :)
I have a sensation that generally the kernel is also quite unstable on
most platforms unless a silicon manufacturer is there to help (as it
happens with some TI chips) and in general I would prefer Intel for
reliable live performance. However as a researcher I am trying to
squeeze ARMs to perform as musical instruments and I think they can work
well if the industry supports kernel development. But I'm wondering if
this will continue to happen, since the eastern mobile market is
crushing the sales of the reliable manufacturers.
Going slightly OT: I really hate how the market is pushing on short
products lifecycles, following the trend of the mobile industry. On one
side the audio and music market is similar to the consumer market as
users want to have ever new and fancy products with appeal. On another
side it is similar to the industry/autmotive market as you need reliable
products that last for years. What you should find inside is sturdy
electronics with a support of >10 years and the possibility to find new
pin-compatible ICs after those 10 years from the manufacturer. There's
too much consumerism in the silicon industry following the mobile
"revolution" meaning that everything contains electronics is destined to
last less and die shortly. Or enter the market in pre-beta stage (which
nowadays is considered a "feature"). The only way to get long term
support is sticking to the old good silicon manufacturers, hoping they
won't discontinue your MCU/CPU/DSP soon (as they are doing to cut costs).
I hope someday people will realize that not all electronic products are
like smartphones.
Hi,
Something for Halloween.
Raul is a Vampire who is driven by his unquenchable passion in the dead of
night.
https://www.youtube.com/watch?v=2KJTGA4S0zQ
This is a re-edit of an earlier private work titled "Rauls Redeption" by
Volker Allert, http://volkerallert.eu, a renown physical effects creator
for the Hollywood film industry.
All audio recorded with Linux tools and edited in Blender on Linux of course.
--
Patrick Shirkey
Boost Hardware Ltd
Hi Linux Audio Users and Developers
Last week the MOD Duo kickstarter campaign was successfully funded and we’re now heading towards turning the Duo into a reality!
As most, if not all, of our software is related to Linux Audio we would like to have the community involved in the development as much as we can.
To do so we created a mailing list and registered an IRC channel.
The mailing list located at http://portalmod.com/cgi-bin/mailman/listinfo/developers
and the IRC channel is #portalmod at freenode.
We’re soon to start discussions regarding the ControlChain (our LV2 hardware communication protocol) and also methods of optimizing the plugin graph cpu usage and we’d like to have as many people involved as we can.
Of course the discussion is not limited to those topics and any thing regarding the MOD can be discussed at such places.
Hope to see you guys on our lists.
Best wishes to everyone
Gianfranco
The MOD Team
Hello,
A long time i didn't post here, and now for something a bit OT:
I just finished to write a tiny Qt based software for DIYers on
loudspeaker enclosure design, under Linux. It is inspired by the
discontinued gspeakers Gtk+ based software. For the moment it has
basic only features, but it is usable, though there is a long TODO.
I you are interested, (and if some debian packager is around here,
[whisling] ;-) you can checkout the svn tree and try it yourself:
$ svn co https://dbx.gtmp.org/svn/qspeakers/ [*]
$ cd qspeakers; qmake -config release && make && sudo make install
(it will install in /usr/local/..)
There is a 4 or 5 speakers DB for a start, but it won't be visible
if qspeaker is not _installed_. After installation, first run will
copy the DB into ~/.local/share/data/Herewe/QSpeakers so that you
can edit it with your own speaker's Thiele-Small parameters.
Thank you for trying !
- Ben
[*] thanks to Mid', a friend of mine, who gave me some space on his server.
