Hello all!
I'm sorry to bother you with that, but I don't find a feasable solution for
this. It's all about patterns. Imagine the following text examples:
<A HREF="url" TITLE="MY_TITLE">
<a NAME="x" id="x">
<P align="center">
What I need to do is make sure, that every TAG (a and p in these cases) and
every attribute (href, title, name, id and center in these examples) is lower
case. I've tried a lot, but I'm not the crack at patterns and so I got no
completely satisfying result, if I got any at all. :-(
Could someone please help me here?
Kindest regards
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de
I'm wondering if it is possible to force a jack client into an error condition
without actually changing the client. I'm wanting to see what actually happens
when my client's error + shutdown handlers are called. It seems as though my
'shutdown handler' isn't being called when jack shuts down.
Thanks,
-Alex
Hello all,
Today I had a strange problem when using the
WFS system here.
The 'master' PC sends a multicast message every
1024 samples (21.333 ms) to all 'render' PCs.
This has to arrive on time, and when it's late
the renderers will mute their output and report
the error in their status messages.
Today I used some ssh -X logins from my laptop
in the WFS room to the WFS master to run Ardour
and some other apps for a demo. All this worked
well all the time, as it has done before.
I left everything running when going for lunch
with our visitors, and when I returned restarted
Ardour to listen again. I got a lots 'of 'message
too late' errors from the rendering machines, and
interrupted sound. Strange enough this seemed to
be related to the _volume_ of the sounds...
In other words to Ardour's level meters.
Restarting Ardour in a new ssh login did not
help, but running it directly on the WFS master
solved the problem. So apparently the network
traffic required to update Ardour's meters was
causing the delays. And clearly the whole remote
X session was slower than normal. CPU loads
looked normal.
Now all this should be peanuts for a Gbit
network that has no other traffic at all,
and it worked perfectly before. I just never
left it running for such a long time.
Anyone an idea as to what is happening here,
and how it could be cured ?
TIA,
--
FA
Laboratorio di Acustica ed Elettroacustica
Parma, Italia
Be quiet, Master Land; and you, Professor,
will you be so good as to listen to me ?
Howdy!
It's been quite a while since last time (Fluffy Doll on Xmas:). However,
Qtractor is back again on track and this time with great news. And the
big news are that this pet is leaving its rusty four-on-the-floor cage
and spreading it's musical genre targets. Still a bedroom/home-studio
sequencer though, but not for the techno-boy/girl only anymore--if one
may trump about it, it's getting a general-purpose sticker now.
And what makes it like just that? One long due feature, now stroked by
implementation lightning: Tempo/Time-signature Map. Or in other words:
project sessions may now have multiple parts with different tempo (BPM)
and/or time-signatures.
Tempo/time-signature map is/was a very pervasive feature change.
Although deeply internal, as it's only evidently visible from the new
View/Tempo Map... interactive yet primitive dialog, it is also
accessible by double-clicking on the time rulers (main track-view and
MIDI clip editors) and by left-clicking over the main tempo spin-box,
which also introduces the time-signature figures as seen from the
current play-head position.
One should probably say that this is the major change in Qtractor
internals since its primordial inception. No doubt, we're still in
/alpha/ status. It surely will take a (another) while, a year or so, for
a similar payload take a lift, ever again. No sweat. Watch for next
favorite feature requests, like MIDI controller map/learn/feedback and
automation. Coming next, soon, hopefully :)
A few words of caution must be said now. It is of paramount interest for
all of you who already use Qtractor for any (small) project or
prototype, to be prepared for less amenable surprises. Being
optimistically wise, there's a very good chance that all new workings
will bring a better experience overall. But given the whole nature and
depth of what got through, it is wise enough to have your backups at
hand and up to date. Don't hesitate asking for help, in any case.
Caveat emptor!
So, with no further ado, there it is:
Qtractor 0.4.0 (foxy dryad) is now released!
Good old intro/description:
Qtractor is an audio/MIDI multi-track sequencer application, written in
C++ on top of Qt Software's Qt4 framework, having JACK and ALSA as its
main infrastructures and Linux as native and exclusive platform.
Specially suited to the lone-wolf composer, arranger and (re)creative
music-maker personal home-studio, it still hopes to evolve as a fairly
featured desktop audio/MIDI workstation or at least, a prototypical part
of it ;)
Release highlights:
* Tempo-map/Time-signature support. (NEW)
* MIDI Song Position cueueing support. (NEW)
* MIDI Clip Quantize command. (NEW)
* Zoom direction mode option (NEW)
* MP3 audio format support (FIXED)
* and many other assorted fixes and brand new bugs ;)
Website:
http://qtractor.sourceforge.net
Project page:
http://sourceforge.net/projects/qtractor
Downloads:
- source tarball
http://downloads.sourceforge.net/qtractor/qtractor-0.4.0.tar.gz
- user manual
http://downloads.sourceforge.net/qtractor/qtractor-0.3.0-user-manual.pdf
(nb. yes, it's the same old, dusty and outdated manual, sorry)
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.
Features:
- Multi-track audio and MIDI sequencing and recording.
- Developed on pure Qt4 C++ application framework (no Qt3 nor KDE
dependencies).
- Uses JACK for audio and ALSA sequencer for MIDI as multimedia
infrastructures.
- Traditional multi-track tape recorder control paradigm.
- Audio file formats support: OGG (via libvorbis), MP3 (via libmad,
playback only), WAV, FLAC, AIFF and many, many more (via linsndfile).
- Standard MIDI files support (SMF format 0 and 1).
- Non-destructive, non-linear editing.
- Unlimited number of tracks per session/project.
- Unlimited number of overlapping clips per track.
- XML encoded session/project description file.
- Point-and-click, multi-select, drag-and-drop interaction (drag, move,
drop, cut, copy, paste, delete, split)
- Unlimited undo/redo.
- Built-in mixer and monitor controls.
- Built-in connection patchbay control and persistence (a-la QjackCtl).
- LADSPA, DSSI and native VST plug-ins support.
- Unlimited number of plug-ins per track or bus.
- Plug-in presets, programs and chunk/configurations support.
- Audio/MIDI clip fade-in/out (linear, quadratic, cubic).
- Audio/MIDI clip gain/volume, normalize and export.
- Audio clip time-stretching (WSOLA-like or via librubberband),
pitch-shifting (also via librubberband) and seamless sample-rate
conversion (via libsamplerate).
- Audio/MIDI track export (mix-down, merge).
- Audio/MIDI metronome bar/beat clicks.
- Unlimited tempo/time-signature map.
- MIDI clip editor (matrix/piano roll).
- MIDI instrument definitions (a-la Cakewalk(tm))
- JACK transport sync master.
- MMC control surface enabled.
- MIDI Song Position cueuing support.
- Configurable keyboard shortcuts.
Change-log:
- MIDI (re)connections fix; now caring for the ALSA client and port
textual names only, avoiding as much as possible, any reliance on those
volatile client and port numbers.
- Transport/Backward and Forward commands may now reset to immediate
full start or end of session locations, by just pressing the Shift or
Ctrl keyboard modifiers and clicking their respective toolbar buttons.
- Default session/MIDI resolution has been set to 960 ticks per beat
(960 TPQN, where a beat equals a quarter-note); it is worth of note that
the previous default resolution was set to one order of maginute lower,
ie. 96 TPQN ;).
- Making (dis)connections now also flags session as dirty.
- Internal Audio/MIDI engines queue/time drift correction takes a brand
new approach, specially adapted to rolling tempo/time-siganture changes.
- MIDI monitor refresh-cycle slight internal optimization.
- Converted obsolete QMessageBox forms to standard buttons.
- Transport/Rewind and Fast-forward commands may now be set to
double-speed, by pressing the Shift or Ctrl modifiers while clicking
their respective tool buttons.
- MIDI clip editor zoom ratios are now saved and preserved across sessions.
- Time-signature is now directly accessible from the main tempo spin-box
which also reflects current tempo status.
- Time/frame spin-boxes now allow to step change each field
individually, depending on the cursor beam position.
- Make sure that Transport/Follow playhead option is only effective when
playback is actually rolling.
- Primordial attempt to include MIDI Song Position Pointer (SPP), Song
Start, Stop and Continue sequencing support.
- A completely new time-scale infrastructure is now in place, with full
support for session tempo and time-signature map; this long due feature
is primarily accessible through the main menu, View/Tempo Map...; also
by double-clicking on the the main window and MIDI clip editor time
rulers and left-clicking on the main toolbar tempo/signature spin-box.
- Moving and resizing individual clips now cares for track proper
ordering and overlapping changes, avoiding nasty out of sequence clips
and other unpredictable effects.
- An expedite MIDI clip quantize command is now available from the main
track view menu (Edit/Clip/Quantize), which simply applies the current
snap-to-beat setting to a MIDI clip range selection.
- Fixed that hideous bug affecting overlapped audio clips when playhead
gets moved backward, causing the playback of those audio clips in
particular, go out of sync.
- Tracks are now limited to their minimum height, specially effective in
face of vertical zooming.
- Zoom mode option introduced (on menu View/Zoom/Horizontal, Vertical, All).
- Tempo beat type is a new session property; however it is not yet user
modifiable and currently disabled to default MIDI quarter note (1/4).
- All open MIDI clip editor time-scales are now updated and corrected
when the main session time base changes (tempo, time-signature,
resolution, etc.)
- MIDI metronome fixed, preventing duplicate click events.
- MP3 audio file decoding was broken for way too long and falling short
for every file with custom frames, ID3 tags and comments. Got shamefully
fixed.
- Time signature denominator (ie. beat divisor) is now an accessible and
effective session property.
- Attempt to retain original size (clip length) of all audio clips when
changing the global session tempo and automatic time-stretching is not
an option.
Cheers && Enjoy!
--
rncbc aka Rui Nuno Capela
rncbc(a)rncbc.org
Hi all !
I'm new in this group, and also a newbie in audio dev on linux...
I'm looking for a cross platform library con which I could read wav and
mp3 files, and pass the steams that I have read to portaudio.
Does anyone know which library could I use, under the LGPL licence ?
I found some libraries to read wav files (like libsndfile) but none to
read both mp3 and wav
Thanks in advance
Regards
Vincent
Hello all,
Jconv-0.8.0 is now available at the usual place
<http://www.kokkinizita.net/linuxaudio/downloads>
New in this release:
- New command line option -L <nsamples>, tries
to compensate for 'nsamples' of processing
latency.
- Jack ports can now be given meaningful names
and optionally be connected, by commands in the
config files.
- Some new reverb IRs available in a separate
package 'jconv-reverbs'. Includes ambisonic and
stereo IRs of the concert hall of La Casa della
Musica, and the former Santa Elisabetta church,
now La Casa del Suono, in Parma.
Enjoy !
--
FA
Laboratorio di Acustica ed Elettroacustica
Parma, Italia
Be quiet, Master Land; and you, Professor,
will you be so good as to listen to me ?
Anyone on the list familiar with the code base of sonic visualizer? I'm
wondering if it would be possible to make a standalone program that works
multiplatform and can read live audio data, and return an array of the
likelyhood that each note was played in an audio stream in the previous 20ms
(not a hard limit, just an example). Similar to how the melodic range
visualization works, everytime x samples are read, check which chords, and
octives they fall into, so say there is an 88 position array of all notes
that are on a keyboard, and middle c is played, so position 60 in the array
would get a high value. I want to be able to work off acoustic data. This in
not going to be used to drive a sequencer or anything. I just want to be
able to check if said note was played.
Thanks in advance for any feedback. If there is another program that you
think would provide better results please let me know. I've tried aubio, and
it crashed all the time on me.
Nathanael
Takashi Iwai wrote:
> At Tue, 10 Mar 2009 10:00:13 +0100,
> Clemens Ladisch wrote:
>
>> (CC alsa-devel)
>>
>> Viral Mehta wrote:
>>
>>> Hi,
>>>
>>> I am trying to do gain control while RECORDING for my Audio Mic device.
>>>
>>> My application opens /dev/mixer device and calls ioctl(fdmixer,
>>> MIXER_WRITE(SOUND_MIXER_MIC), ...)
>>>
>>> But, the call fails. It traces to sound/core/oss/mixer_oss.c file and
>>> snd_mixer_oss_put_volume1() function.
>>> It never falls in "if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME)"
>>> as condition "if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) "
>>> comes as false.
>>>
>>> I think there is a bug in kernel and I think it should be like as below,
>>>
>>> --- sound/core/oss/mixer_oss.c 2008-12-03 13:24:02.000000000 +0530
>>> +++ sound/core/oss/mixer_oss.c 2009-03-09 16:22:06.548766896 +0530
>>> @@ -688,7 +688,7 @@ static int snd_mixer_oss_put_volume1(str
>>>
>>> if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) {
>>> snd_mixer_oss_put_volume1_vol(fmixer, pslot,
>>> slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right);
>>> - if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME)
>>> + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) {
>>> snd_mixer_oss_put_volume1_vol(fmixer, pslot,
>>> slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right);
>>> } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GVOLUME) {
>>> snd_mixer_oss_put_volume1_vol(fmixer, pslot,
>>> slot->numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right);
>>>
>> Your mailer mangled the patch; see linux/Documentation/email-clients.txt.
>>
>
> Also, the current code behaves intentionally so (as it's designed for
> mic-loopback volume).
>
> A more feasible fix would be to add another if check for *_CVOLUME
> instead of moving it.
>
>
>
Oh okie. It makes sense.
I just sent out a patch on LAD. Should I resubmit with what you proposed ?
> Takashi
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