Hi :)
are the coders of Calf Monosynth at that list?
Before I read "maximize flexibility while minimizing the number of
controls. Useful for synth basses", I exactly got this opinion. This is
one of the best virtual synth I know. Thanx :).
If you should have a wish list: I'm using a 61 keys keyboard and the
note numbers are fixed. I would like to have the option to transpose
Monosynth by octave steps.
Cheers!
Ralf
Hello Everyone,
In my new job I've been assigned to create a proof of concept "mp3 player"
using an existing product from our company. Trouble is, I've not done
anything with audio before and the timeframe is tight. I guess my new boss
thinks it's a piece of cake - "it's only software, after all, right?" The
main problem is that this Linux-based device doesn't have _any_ audio
capability so we're having to engineer something. The current plan is to
connect a fancy Bluetooth (CSR BC5) chip and use it for the audio-output
stage.
Although I don't think this forum is necessarily the right place to go
looking for help, I'm hoping someone can point me in the right direction for
where to get some education. I really don't understand the finer points of
how the audio signal gets to the device driver and this is where some help
would be appreciated. Specific questions I have are:
- What's the general flow of the data from the disk file (or streaming
audio source) to the device driver? Is it: MP3file -> streaming software ->
codec -> driver or// MP3file -> streaming software -> Codec -> streaming
software (e.g. via callback) -> driver? (or does it even have a standard
approach?)
- The Bluetooth part can be connected by UART or USB to the host. USB
would be better because the BT part enumerates as an audio-class device.
This would probably solve my problem, but our Linux device hasn't an
available host-side interface, I'd have to delve into USB OTG and maybe cut
and jumper some traces. That leaves UART. I don't see how a UART can be made
to look like an audio device unless I write a driver that presents as an
audio interface to Linux and exchanges the audio traffic between user space
at the top-end and the UART on its bottom-end. Is there an easier way to
create an audio interface out of a UART?
I've downloaded several packages to take a look at how they implemented, but
they all contain dozens (or hundreds) of files. Progress through will be
slow, but I'm guaranteed to find an answer. I've got copies of:
- gstreamer
- MuSE
- madplay
Between the three of these I think I'll be able to figure something out
(needs to be ported to ARM9). Is anyone able to make a recommendation?
Thanks for reading this post - I promise not to be a pest on this list (but
would be happy to share my eventual solution, if anyone wanted to know it)
Regards,
Pete Moss.
Hey guys!
Having moved to Linux I noticed a problem with holding keys. If I am using
JACK-keyboard or even tried this with a Windows sequencer through WINE,
holding a key results in triggering the note many-many times instead of just
playing it continuously, sustained. There seems to be no problem on Windows
with this. Is it a platform specific thing? Can this be somehow realized? I
would love to just press a key, hold it and have the note sustained.
--
Louigi Verona
http://www.louigiverona.ru/
Hi,
I'm having troubles posting to the list. Hopefully this mail will make it
through but I can never be sure. Switching SMTP server sometimes help, sometimes
not.
Some days it works, some days it doesn't. Yesterday, I tried several times to
answer to Geoff on the "pulsating noises" thread, but it didn't work. So that
Ralf (who was in CC) answered my post before mine arrived, which eventually
found its way to the list today.
Here's the error I always get:
<linux-audio-dev(a)lists.linuxaudio.org>: host
a.mx.lists.linuxaudio.org[198.82.152.114] said: 550 5.7.1
<linux-audio-dev(a)lists.linuxaudio.org>: Recipient address rejected: Mail
appeared to be SPAM or forged. Ask your Mail/DNS-Administrator to correct
HELO and DNS MX settings or to get removed from DNSBLs; please relay via
your ISP (samalyse.com) (in reply to RCPT TO command)
I'm participating to a rather important number of lists, and LAD is the only
place where I have this problem (dunno about LAU and LAA, since I haven't posted
there for a while). Also, it never happens in all my other personal mails.
--
Olivier
This is probably a stupid question.
My guess is that quantisation noise is only something present between
the input signal and its digital representation, and hence no change of
the digital representations can do anything about it.
--
Regards,
Philipp
--
"Wir stehen selbst enttäuscht und sehn betroffen / Den Vorhang zu und alle Fragen offen." Bertolt Brecht, Der gute Mensch von Sezuan
Hello,
I have developed an audio recording application for Android, and a user is
reporting a weird noise issue. There are near-zero chances that this comes from
my code, and I'm looking for the possible cause of these noises.
Here is the sample, it's a musical flute and violin recording:
http://sound.samalyse.org/tapemachine/pulsations.wav
Listen carefully and you will here small pulsations.
I've also asked him to record a 440Hz sine. This one isn't distorted:
http://sound.samalyse.org/tapemachine/record440hz.wav
I suspect that this is a hardware issue, but I'm no expert so I can't really
tell. The phone is an HTC Incredible, which unless I'm wrong, means that the
sound components are embedded into a Qualcomm Snapdragon QSD8650 chip.
One other is that, although Android is a linux-based system, and use alsa
underneath, there's a terrible high-level Java API, which is full of various
problems which I have worked around to obtain that kind of quality. Thus, I
can't really tell what happens in there.
Have you already heard such things, and know what the cause, whether hardware or
software, is likely to be?
--
Olivier
I couldn't notice (in Germany we call it) 'pumping'. I don't think it's
caused by auto-leveling. OTOH the pulsing noise might be caused by a
strange limiter because of some peaks, but try to reproduce such a sound
using LADSPA limiters, compressors on your computer. Ok, it might be
caused on the analog domain, anyway, for analog limiters and compressors
it's the same.
Regarding to a 'half level' recording. If the user hasn't got the
equipment to reduce the level, a pocket handkerchief might help
(unfortunately it would be a pass filter too ;).
Because it's a telephone, it might be all kind of strange noise. I guess
we all know the sound a cellphone could cause to radios etc.. How do
they prevent that such noise has no impact to the cellphone's audio itself?
Hi LAD,
Since the days I installed my RME HDSP card combined with a Multiface II IO box, I have never been able to suspend / resume my PC because the multiface ends up in a weird state at resume time. I have to shut down and power off the the PC to reset everything.
Is there any improvement in recent driver iterations ? I would really like to be able to suspend / resume my PC sometimes.
Thanks for any info on the matter.
J.
Drumstick is a C++ wrapper around the ALSA library sequencer interface using
Qt4 objects, idioms and style. ALSA sequencer provides software support for
MIDI technology on Linux. Complementary classes for SMF and WRK file
processing are also included. This library is used in KMetronome, KMidimon
and KMid2, and was formerly known as "aseqmm".
Changes:
* Fixed a drumstick-sysinfo crash, when retrieving information for an
unavailable timer module.
* Fixed WRK file read implementation, allowing compilation on more
architectures.
* Added standard arguments to all the utilities/example programs.
* Added man pages for all the utilities/example programs.
* New utility/example program: drumstick-drumgrid, a simple MIDI drum pattern
editor/player.
Copyright (C) 2009-2010, Pedro Lopez-Cabanillas
License: GPL v2 or later
Project web site
 http://sourceforge.net/projects/drumstick
Online documentation
 http://drumstick.sourceforge.net/docs/
Downloads
 http://sourceforge.net/projects/drumstick/files/0.3.2/
Regards,
Pedro
On 5/29/10, akjmicro(a)gmail.com <akjmicro(a)gmail.com> wrote:
> Hey all,
>
> Yes, grepping for the port type which appears underneath with a 'jack_lsp
> -t' will be more consistent and dependable. Or, using a python-jack lib
> function and not depending on any system shell calls. The problem then
> becomes, is the jack lib for python well documented? If so, I think that's
> the real future of jackctl.py
>
> PS Qjackctl may be lightweight, but installing the entire QT toolkit just to
> use it is not!
>
> AKJ
>
> Sent from my Verizon Wireless BlackBerry
>
> -----Original Message-----
> From: Robin Gareus <robin(a)gareus.org>
> Date: Sat, 29 May 2010 14:00:52
> To: Julien Claassen<julien(a)c-lab.de>
> Cc: Aaron Krister Johnson<aaron(a)akjmusic.com>;
> <linux-audio-user(a)lists.linuxaudio.org>;
> <linux-audio-dev(a)lists.linuxaudio.org>
> Subject: Re: [LAD] [LAU] like "qjackctl", but trimmed of all fat
>
> Hi Julien, Hey Aaron,
>
> read 'jack_lsp --help'.
>
> '-t' does not take any arguments; it just makes jack_lsp print the type.
> the filter-string only acts on the port-name (BTW, not only the
> beginning of the port-name; but it's case-sensitive: strstr() )
>
> Anyway I can reproduce the problem, some jack-midi ports show up in the
> audio-tab of jackctl20100528b.py.
>
> jackctl20100528b checks for lowercase 'midi' in the port-name instead of
> looking up the port-type. So a2jmidi for example with an upper-case M
> "Midi.." ends up in the audio-panel.
>
> Your suggestion to parse the output of 'jack_lsp -t -c' is spot on.
> the (currently 2) possible return values are (indented by tab):
>
> #define JACK_DEFAULT_AUDIO_TYPE "32 bit float mono audio"
> #define JACK_DEFAULT_MIDI_TYPE "8 bit raw midi"
>
> ..or as you suggest using the python-module for JACK may also simplify
> things and make jackctl easier to maintain.
>
> Cheers!
> robin
>
> PS. Oh, and which of qjackctl's features makes it 'fat'? it's not
> bloated in any way. I'd rather put it the other way 'round and say that
> jackctl is 'slim'. Sorry could not resist.
>
>
> On 05/29/2010 12:23 PM, Julien Claassen wrote:
>> Hello Aaron and Jack-Team!
>> There seems to be a bug in my jack_lsp. I just started a2jmidid and
>> j2amidi_bridge. when I do a jack_lsp I get all the ports.
>> When I do: jack_lsp -t midi I only get one port from jack_midi_clock,
>> but none of the other ones.
>> When I type: jack_lsp -t, I can't see a difference between the
>> jack_midi_clock port and the others:
>> jack_lsp -t
>> [...]
>> a2j:Virtual Raw MIDI 0-0 [16] (capture): VirMIDI 0-0
>> 8 bit raw midi
>> a2j:Virtual Raw MIDI 0-0 [16] (playback): VirMIDI 0-0
>> 8 bit raw midi
>> a2j:Virtual Raw MIDI 0-1 [17] (capture): VirMIDI 0-1
>> 8 bit raw midi
>> a2j:Virtual Raw MIDI 0-1 [17] (playback): VirMIDI 0-1
>> 8 bit raw midi
>> a2j:Virtual Raw MIDI 0-2 [18] (capture): VirMIDI 0-2
>> 8 bit raw midi
>> a2j:Virtual Raw MIDI 0-2 [18] (playback): VirMIDI 0-2
>> 8 bit raw midi
>> a2j:Virtual Raw MIDI 0-3 [19] (capture): VirMIDI 0-3
>> 8 bit raw midi
>> a2j:Virtual Raw MIDI 0-3 [19] (playback): VirMIDI 0-3
>> 8 bit raw midi
>> a2j:M Audio Delta 1010LT [20] (capture): M Audio Delta 1010LT MIDI
>> 8 bit raw midi
>> a2j:M Audio Delta 1010LT [20] (playback): M Audio Delta 1010LT MIDI
>> 8 bit raw midi
>> j2a_bridge:playback
>> 8 bit raw midi
>> a2j:j2a_bridge [129] (capture): capture
>> 8 bit raw midi
>> Jack MIDI Clock:midi_out
>> 8 bit raw midi
>>
>> Or is the argument "midi" only seen as the start of a port_name?
>> If so, Aaron, you must rewrite this part of jackctl (I guess you do
>> what I described, because I get exactly your output). You should rewrite
>> it using:
>> jack_lsp -t
>> And then parse the type info underneath each name. I think a simple
>> grabbing for "audio" or "midi" will do. But I guess, that in the long
>> run, using the python module for jack, will be more efficient and easy
>> to use.
>> Kindly yours
>> Julien
>>
>
--
Aaron Krister Johnson
http://www.akjmusic.comhttp://www.untwelve.org