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 ?
Hi all,
(And apologies for cross posting)
I'm pleased to announce QuteCsound version 0.4. This version
incorporates new features and bug fixes arising from the 3 previous
Release Candidate releases.
QuteCsound is a simple frontend for Csound featuring a highlighting
editor with autocomplete, interactive widgets and integrated help. It
is a cross-platform drop in replacement for MacCsound (it can open
files created in MacCsound), and aims to be a simple yet powerful and
complete development environment for Csound.
QuteCsound 0.4 has been tested on Windows, OS X, Linux and Solaris,
and it is free software released under the GPL.
You can get it here:
http://sourceforge.net/project/showfiles.php?group_id=227265&package_id=275…
There's more information here:
http://qutecsound.sourceforge.net/
Questions, comments and suggestions can be posted to the main Csound
mailing list, or join the QuteCsound users mailing list at:
http://lists.sourceforge.net/lists/listinfo/qutecsound-users
Thanks very much to all the people (particularly Joachim Heintz), who
have provided great feedback and ideas to improve QuteCsound. Also a
big thanks to Andy Fillebrown for the fixes for Windows and for
preparing the Windows installers.
Enjoy!
Andrés
Changes from version 0.4RC2:
New:
- Added "_Browse" channel for value buttons, which open a browse file
dialog and send the selected file name to any widget with the same
channel which can receive strings, like LineEdit. Added this feature
to the reserved channels example.
- New tutorials and example files, with clearer names. The examples
can now be run directly WITHOUT saving, which is very handy especially
for testing audio IO.
- Console text font color can now be changed
- Added a close tab button on the right corner of the tabs panel.
- Modified execute code for Windows (for external utilities and terminal)
- Added option to enable/disable line wrap
- Added clear option to console context menus
- MacCsound output filename is now parsed and added to the options
- Added new FM and additive examples
Fixed:
- Improved stability
- Files with extensions different to csd, orc and sco can now be opened
- External programs and terminal are now launched correctly on all platforms
- Fixed storing changes in widget panel when tab changes
- Graph widget now displays graphs generated by ftgen
- Widget values are now properly sent for the first control period
- Files now open properly on OS X when double clicked from the Finder
- Fixed problem when buffering messages
- Fixed problems with filenames with spaces when rendering or opening
in external programs
- MacCsound sections are only created if save Widgets preference is set
- New default file more consistent with current state.
- Files are now not marked as modified when changing tabs
- Fixed path of Manual for newer versions
- Fixed resizing and moving widget edit frames when changed from the
properties dialog
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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>
>
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Have tested two usb audio mic devices.
At the time of initialization, SNDRV_MIXER_OSS_PRESENT_PVOLUME bit is not
set for MIC (slot 7).
So, the same should not be checked when an application tries to do gain
control
for audio recording devices. Just check
slot->present for SNDRV_MIXER_OSS_PRESENT_CVOLUME independently.
Verified with a simple application which opens /dev/dsp for recording and
/dev/mixer for volume control.
Signed-off-by: Viral Mehta <viral.mehta(a)einfochips.com>
---
sound/core/oss/mixer_oss.c | 2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index 4690b8b..919c7eb 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -690,7 +690,7 @@ static int snd_mixer_oss_put_volume1(struct
snd_mixer_oss_file *fmixer,
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);
--
1.5.3.3
--
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__________________________________________________________________________
Hello all,
I would like to know feedback here if possible. Or is it a wrong place
to submit patches which I "think" fixes some issue in oss simulation layer ?
> 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,
>
> --- a/sound/core/oss/mixer_oss.c 2008-12-03 13:24:02.000000000 +0530
> +++ b/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);
>
>
> Thanks,
> Viral
>
>
>
>
>
>
--
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__________________________________________________________________________
Hello all,
Some good news: there definitely will be the Linux
Sound Night during LAC2009. At the moment there are
two possible locations, the choice has not yet been
made. One is a nice bar located in a new industrial
area outside the center - they only have one resident
near (100m) but we may still have to watch the SPL
because of that single resident. The other is a rather
sleezy but otherwise OK place in a post-industrial
area, here nobody will complain whatever we do.
Both places (in particular the second) are a bit to
far from the center for an easy walk, but transport
will be organised.
I'd like all who proposed to play at the LSN to
confirm their presence (drop me a line), and there
is still room for more.
Wednesday the 15th (day before the conference) there
will be the pizza evening for the early arrivals.
This will be at 'Alfonso' (no relation) on Via Emilia
Est, according to some the best pizzeria in Parma.
It's also near the Auditorium Paganini making it a
good place for a pre-concert pizza. Please let me
know if you will be there as I'll have to book the
places. Sunday evening we'll have the unofficial
LAC2009 dinner.
At the moment I'd like everyone who plans to come
to LAC2009 to register so we can have an idea of
the size of the crowds to expect, and of the number
of proceedings to print.
<http://lac.linuxaudio.org/2009>
See you all in Parma,
--
FA
On Mon, 9 Mar 2009 11:09:34 -0700, Mark Knecht <markknecht(a)gmail.com>
wrote:
> In general I agree with what I understand about linuxaudio.org and
> your point of view.
>
> Probably I need to make a more nuanced distinction between
> linuxaudio.org and linux-audio-user(a)lists.linuxaudio.org. It's likely
> not true for you running all this stuff but from the outside I see the
> two as very intertwined. If the business of linuxaudio.org were to
> become a large part of what is discussed on this list, or the list was
> to start pushing folks to become Debian based as a policy, then I'd be
> disappointed. I think it's best if this list is distro neutral. What
> linuxaudio.org does is a different thing.
Well, hello again, this time from the list-admin part of me.
I would like to clarify this for those who might be here since recently and
who might not know of the history of our 4 favorite lists. The hosting at
linuxaudio.org is relatively recent, and was chosen in an effort to make
things look a little more organized. Our lists are kindly hosted there by
the Virginia Tech Department of Music and DISIS. They were previously
hosted at the University of Columbia (at the time there were 3 lists ;
linux-audio-tuning did not yet exist). The first list created was, if I'm
not mistaken, linux-audio-dev, a long time ago, in a galaxy far far away
(1998 ?).
While our lists are hosted at linuxaudio.org, they do not belong to anyone
but the whole community that make them as lively and entertaining as they
are. Moreover, they cannot be in any way related/associated/linked/whatever
to one particular distribution. So feel free to talk about any distro you
feel is good for audio, even slackware if you wish :p
And one last thing, just so everything is crystal clear, since these list
belong to everyone, you are also entitled to fire the list admin (that
would be me) at anytime if I start being really bad at that ;)
Kind regards,
Marc-Olivier Barre,
The guy who gets to read a lot of spam to keep those lists clean.
> How does that effect me for instance as a Gentoo user? Please consider
> whether LAU would still be a place of value to those of us that aren't
> Debian/Ubuntu based.
>
> I don't mind the occasional distro specific conversation. I wouldn't
> want to see that ever become more than 1% of what's discussed on a
> list like this.
>
> - Mark
Ideally, we would also want to have Gentoo involved in linuxaudio.org.
However please note that I cannot (nor do I want to) coerce anyone to join
linuxaudio.org. Yet, here we are with a potential opportunity to have Debian
join us and my thoughts are we should not pass this up. I can only hope that
Gentoo will join us as well--but that is not my call...
Best wishes,
Ico
Michael, Paul has answered you on jack-devel. See below.
Note: I'm cross-posting this mail to linux-audio-dev, since Michael has recently
subscribed to it. At least, we should be able to discuss together in there.
Paul Davis wrote :
> I have no idea how any of you have kept this conversation going when one
> of the main protagonists is not even subscribed to one of the two
> mailing lists, and i suspect that one of the others isn't subscribed to
> the other. perhaps the FFmpeg-devel list doesn't require membership.
>
> anyway, i've finally had enough of reading Michael's stuff about
> buffers, timing and so forth and felt obligated to comment.
>
> michael - i would politely request that you stop shooting off at the
> mouth about stuff that the JACK community has been dealing with for more
> than 9 years.
>
> you do not need to write your own ring buffer code - JACK's is LGPL'ed
> and you are free (and probably even recommended) to copy it.
> furthermore, you would be very foolish to imagine (especially based on
> your incredibly naive comments about uint8_t) that you understand the
> subtleties of these buffers. the JACK community (and a couple of other
> exclusively audio-focused development groups) have been working with
> this buffer design for many, many years, and we are absolutely confident
> that our buffers are (a) SMP/multi-core safe (b) as efficient as they
> can be without using assembler. you should also be aware that there are
> very good arguments for the current structure of the ringbuffer code,
> which explicitly does NOT use any memory barriers. if you don't
> understand why it works without them, then you should probably refrain
> from commenting on the design of these buffers at all.
>
>