Hello all,
I wonder if any other users have experienced this problem and
how they handled it.
This has occured three times when doing an fresh Archlinux install
on a system using the RME MADI cards.
There seems to be something in the combination of recent versions
of the driver and alsactl that leads to alsactl freezing when the
configured (external) clock source for the card is not available.
The 'freeze' seems to be quite deep: it's impossible to kill the
process (even while that process is still a child of e.g. the
xterm from which it was launched, and not of PID 1). Any other
process trying to access the sound card (e.g. jackd) hangs in
the same way. This also means that when doing a poweroff or reboot
systemd will hang on the 'alsactl store' service, and the only
option is a power cycle.
An added difficulty when trying to resolve this (things will be
OK once you have the correct /var/lib/alsa/asound.state) is that
recent systemd doesn't allow to disable or enable the alsa store/
restore services easily (why not ?), you have to manually edit
some symlinks in order to do that.
Note: if this happens to be a driver problem, please do NOT revert
to the ancient behaviour of silently changing the clock source to
'internal' when the external clock is not available. I DO still
expect to see opening the device fail if the external clock isn't
present, as has been the case for some time. The thing that shouldn't
happen is that alsactl chokes on this condition - it didn't before
so it shouldn't have to.
Ciao,
--
FA
A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)
Hi,
I started working with Linux audio very recently. Primarily, I wanted to
understand the Linux audio stack and audio processing flow. As a simple
exercise,
- I write a simple capture/playback application for PCM interface and store
the captured audio data in a .wav or .raw file
- The device parameters I played with are: hw:2,0/plughw:2,0 (for USB
headset) , 1/2 channel, 44100 sample rate, SND_PCM_ACCESS_RW_INTERLEAVED
mode, period size 32, S16_LE format.
- Use Ubuntu, kernel 3.9.4 and Logitech USB headset for development/testing
To understand the control flow, I inspected the ALSA driver source code and
got an understanding of the flow in kernel space. However, I am kind of lost
in the user-space ALSA library.
- What happens after ALSA handovers the data i.e. the audio data from mic is
copied from USB transfer buffer to userspace buffer? What are the data
processing steps?
- Where is Pulseaudio working here?
- Is there any software mixing happening?
- In the USB sound driver, I changed the data in urb->transferbuffer to a
pattern like 'abcdefghijklmnop', but in the capture application when I store
the audio data in a .raw file, I don't completely get the data back.
If I set 1 channel in the hw params, I get almost 'abcdefghijklmnop' pattern
but sometimes get 'abcdefghijklm' i.e. the pattern incomplete and another
pattern starts over. For 2 channels, the data interleaves in a pattern like
'ababcdcd...' but there are also some incomplete patterns and also I see
some unexpected characters.
I know it's a long post, but even if you can help with a part of it, I'll be
greatly benefited. Thanks!
--
View this message in context: http://linux-audio.4202.n7.nabble.com/Linux-Audio-Architecture-tp85571.html
Sent from the linux-audio-dev mailing list archive at Nabble.com.
Hello all,
Returning home late, on the way from the car parking to my
door I was greeted by a nebula of hundreds of fireflies doing
their social thing.
A lovely thing to see, but it also reminded me that I should
really post the following:
The last months I'm receiving lots of invitations to join
Circles, Friends, Contacts etc. etc. on Google+, Facebook,
LinkedIn etc. etc, many of which from members of this list.
While I do appreciate the motivation behind such requests,
I will never accept them, and from now on I will also stop
responding to any such invitations. If you want to discuss
anything (Linux) audio you're welcome to get in contact via
private email or the LAU or LAD mailing lists.
Ciao,
--
FA
A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)
Hi!
Just a very brief update for those who care about AVB on Linux:
We've recently discussed the subject at LAC 2013 and agreed to use the
open-avb-devel mailing list for "our" internal communication.
Last Friday, we came closer to a working solution. I've already shared
the status quo in a private e-mail with involved parties, one of them
being Intel.
Long story short: the follow-up to this e-mail will be crossposted to
open-avb-devel, which means discussion is open to all. So if you care
about AVB on Linux, you are more than welcome to subscribe to
https://lists.sourceforge.net/lists/listinfo/open-avb-devel
and share your views.
Note that all this is very early development and highly technical,
especially since there is no definite kernel<->userspace API, yet.
Cheers
Dear Free Audio Tool Lovers,
I am very pleased to announce the first official release of FLAC, the Free
Lossless Audio Codec, in over 6 years. FLAC is not dead! It is however a
mature software product that is now being maintained by a team working
under the auscpices of the Xiph.Org Foundation.
The executive summary of changes in this new version:
* Nothing major.
* Source tree is now hosted in Xiph.org git: git clone git://git.xiph.org/flac.git
* Read and write appropriate channel masks for 6.1 and 7.1 surround input WAV files.
* Added support for encoding from and decoding to the RF64 format.
* Lots of build system fixes for your building enjoyment.
The full changelog is here: https://www.xiph.org/flac/changelog.html
Happy lossless encoding and decoding.
Cheers,
The FLAC project contribitors
Lacking access to the full midi specs document, I don't know
if this question is addressed. I've looked at manuals for products
which support them and searched the web but I don't see a clear
answer to my question:
Is it safe to assume that a product or app which allows
binding a *single* HW or GUI control to either 14-bit CC
or 14-bit (N)RPN, would *always* send the value LSB, even if
the LSB did not change but the MSB did, when the control moves?
Do the midi specs address this?
Or do you know of examples of such LSB optimizing-out?
Thanks.
Tim.
Laborejo 0.8 is released
We are still in Beta.
Streamlining, fixing and making things more robust is the main feature.
However, there are a few new features:
- Swing for Playback mode. In the Performance Signature.
- More preview and export options, especially for Lilypond
- Jack Transport start and stop sync. (Update calfbox as well!)
- "Did you know" help screen and splash dialog (optional)
There is a 0.85 release planned which will deal with making containers, selections and undo work together properly and implementing the last small features on my todo-list.
After 0.85 the plan is to work on speeding up the performance and writing documentation and translations. And the usual bugfixes of course.
This will transition smoothly into a stable 1.0 release
Since Laborejo now becomes more useful and usable with each release it became fair to accept donations.
There is now an extra donation page on the website. Please have a look. Your support would be much appreciated:
http://laborejo.org/Donation
Latest Screenshot:
http://www.laborejo.org/latestscreenshot.png
Download Page:
http://laborejo.org/Download
Greetings,
Nils Gey
Contact: http://laborejo.org/Support_and_Community
As an anticipated summer delight...
Qtractor 0.5.9 (juliet quebec) is now released!
Release highlights:
* NSM support (NEW)
* Audio buses plugin state restoration (FIX)
* MIDI channel file merge/export (FIX)
* MIDI editor rescale mode (NEW)
* German translation (NEW)
Website:
http://qtractor.sourceforge.net
Project page:
http://sourceforge.net/projects/qtractor
Downloads:
http://sourceforge.net/projects/qtractor/files
- source tarball:
http://downloads.sourceforge.net/qtractor/qtractor-0.5.9.tar.gz
- source package (openSUSE 12.3):
http://downloads.sourceforge.net/qtractor/qtractor-0.5.9-6.rncbc.suse123.sr…
- binary packages (openSUSE 12.3):
http://downloads.sourceforge.net/qtractor/qtractor-0.5.9-6.rncbc.suse123.i5…http://downloads.sourceforge.net/qtractor/qtractor-0.5.9-6.rncbc.suse123.x8…
- quick start guide & user manual:
http://downloads.sourceforge.net/qtractor/qtractor-0.5.x-user-manual.pdf
Weblog (upstream support):
http://www.rncbc.org
License:
Qtractor is free, open-source software, distributed under the terms
of the GNU General Public License (GPL) version 2 or later.
Change-log:
- Auto-incremental version numbering of backup session files; brand new
versioning mode option added.
- Fixed one long as ever MIDI file export bug, which was about washing
all self-induced MIDI bank-select (CC#0, CC#32) and program-change (PC#)
events into MIDI channel 1, regardless of the source track/channel.
- Fixed initial sample-rate settlement on plugin chains for all tracks
and buses whenever a session file original sample-rate differs from the
current audio device engine (ie. JACK).
- New German (de) translation added (by Guido Scholz, thanks).
- Now accepting JACK transport tempo and time-tignature changes while in
slave transport mode.
- Fix ref-count and auto-removal of created files that result from clip
merge/export when session is closed but not saved.
- MIDI track instrument bank/program selection, from plugin UIs that
support it, is now effectively and complete.
- Custom connections for dedicated audio outputs from all plugin chains
in MIDI tracks or buses are back in business, hopefully restored gain on
session load (ticket by yuba, thanks).
- Track Properties dialog now dirty after any plugin related change
(another head-up by yuba, thanks again).
- Tempo Map/Markers dialog is not set initially dirty anymore.
- Audio buses plugin state's persistence were found AWOL: while being
properly saved as usual, were being restored to default on every session
reload (as noticed by yuba, thanks).
- NSM (Non Session Management) support is being introduced.
- The MIDI clip editor (aka. piano-roll) gets a brand new rescale mode:
ctrl+dragging the right edge of a note now rescales all times and
duration of all the subsequent selected notes (after an interesting
idea/ticket by Joel Leclerc, thanks).
- The new MIDI clip editor rescale mode also applicable to event values
(eg. note velocities, pitch-bend), while ctrl+dragging the edges of an
event vertical bar.
- VST plugin (GUI) idle timer makes a comeback.
- Hard-wired LV2 UI selection order for plugins which supply more than
one option (lv2_external_ui gets always honored first then, as provided
via libsuil, X11, GTK and last but not least, Qt4).
See also:
http://www.rncbc.org/drupal/node/664
Have (lots of) fun.
--
rncbc aka Rui Nuno Capela
rncbc(a)rncbc.org
Hi LAD!
We, from MOD <http://portalmod.com/en> team, released now a new Jack / LV2
Host in github: the mod-host. mod-host is a command line application like
jalv, but that accept multiple instances and can be controlled via socket
or shell.
LV2 features currently supported:
* lv2core
* atom
* event
* buf-size
* midi
* options
* uri-map
* urid
* worker
More information can be found on README file.
repo: https://github.com/portalmod/mod-host
We hope that mod-host can be useful for others projects!
Best Regards,
Ricardo Crudo.
