Hi all
I know this is not the ideal place to ask a question about
Android--but I hope someone here can help confirm my suspicions.
I wanted to create some circuit tools using the 1/8" stereo output +
line input jack on my android phone (a droid razr). To start with,
I've got a simple signal generator (built with PdDroidParty)--and what
I found is that there is a high-pass filter on the output. It is
mostly unsurprising, but what *kind* of high-pass filter is it?
If it was just a RC impedance on the output, I could compensate for
that. However, it seems likely that there's a high-pass filter in the
software, or somewhere else.
When I drive it to clipping with a combination of sine tone and dc
offset, the noise characteristic is weird. The pitch of the distorted
tone goes down, and there's a very steep transition between the
undistorted and distorted tone. The distorted tone is also much
louder. I'm typically testing with a tone at about 1kHz, but I've
been sweeping through a lot of frequencies to see what happens. The
symptom is the same whether it's played through the speaker or a set
of headphones.
It's got to be a software filter, DAC hardware, or output amplifier
circuit. If I could identify that, it would help me figure out
whether this project is feasible or not.
I'd like to make hardware that plugs in to the 1/8" jack to create:
1. signal generator
2. RLC meter
... up to ...
linear/non-linear systems identification tools
audio spectrum analyzer
audio oscilloscope
USB hardware is mostly out of my league--and for the effort required,
there's no guarantee of success. Small steps, first.
Chuck
hey all firstly when i go to add my self to the real time group with these commands
groupadd realtime
usermod -a -G realtime sound-warrior
i get these errors from terminal
sound-warrior@the-megga-machine:~$
sound-warrior@the-megga-machine:~$ groupadd realtime
groupadd: cannot lock /etc/group; try again later.
sound-warrior@the-megga-machine:~$ usermod -a -G realtime sound-warrior
usermod: group 'realtime' does not exist
sound-warrior@the-megga-machine:~$
does any one know what these errors mean and how to solve the problem?
secondly how do you set up and use memory allocation and memory locking for audio work?
Your help with this problem and question would be most welcome.
Kind regards trahern.
On Sat, September 1, 2012 1:27 am, Paul Davis wrote:
> On Fri, Aug 31, 2012 at 10:59 AM, rosea.grammostola <
> rosea.grammostola(a)gmail.com> wrote:
>
>> On 08/31/2012 04:56 PM, Nils wrote:
>>
>>> On Fri, 31 Aug 2012 14:43:13 +0100
>>> Harry van Haaren<harryhaaren(a)gmail.com> wrote:
>>>
>>> On Fri, Aug 31, 2012 at 1:03 PM, Nils<list(a)nilsgey.de> wrote:
>>>>
>>>> The direct and naive solution would be a reversed engineered kontakt
>>>>> sample engine, yes.
>>>>> Very naive.
>>>>>
>>>>>
>>>> The community could approach NI and ask if they're intrested in
>>>> supporting
>>>> a Linux version of Kontact? I volunteer to write the email, and if
>>>> they
>>>> laugh then what harm done...
>>>>
>>>> Opinions?
>>>>
>>>
>>>
>>> Regardless if it was done or in the past or not It would be very nice
>>> of
>>> you to write an email to them.
>>>
>>
>>
>> Maybe an email from linuxaudio.org works better? Someone who speaks in
>> name of an organization?
>>
>
> NI already have inhouse versions of many of their software tools for
> Linux,
> and they use it in house for some development. I met with them in person
> several years ago when I was teaching in Berlin. They are quite big
> technical fans of JACK and of Linux, but they (probably correctly) see a
> tiny, largely irrelevant market for native releases for these platforms.
> They spent quite a bit of effort to get their standalone versions on OS X
> to talk to JACK if it is installed, but chose not to document this because
> hardly anyone wants it and when they do, they figure it out for
> themselves.
>
> A lot of people (even on this list) don't understand the extent to which
> *supporting* a piece of software is often a far bigger cost than the
> initial development, and providing support for a platform with very few
> users is an issue for companies who want their customer service reputation
> to be very good (as NI does). It doesn't work for companies like this to
> just release something "into the wild" and forget about it.
>
That's why it is time for the full scale of the Linux Audio community to
be effectively realised. I'm not talking about the people active on this
list. There is an absolutely massive installation base across the planet
of people who use Linux Audio tools but only a tiny fraction of them are
actually making the effort to directly contribute.
We should be able to make an educated guess as to how many people are
really using the tools we make based on the number of downloads of the
actual software, the number of installs of OS's that are running the
software and a general hands up accounting process.
Figuring out where to start counting when there is so much to get through
is the hard part.
For example people who use firefox or chrome would be a good start. They
are most likely the ones who will also be using Linux Audio Tools.
Another example, Behringer ships Audacity with every single product they
sell. Clearly the global market leader for audio production hardware sees
some value in open source too. We can get an idea of their market base by
looking at their end of year or quarterly financial reports.
--
Patrick Shirkey
Boost Hardware Ltd
Hi,
I wrote a little tuner for my android thing.
I didn't find free-software tuners for android.
It also works with gtk/jack. Slightly less
CPU eater than gxtuner on my host but much
less pretty.
http://sed.free.fr/android/index.html#tuner
for source and apk.
Big thanks to Lorenzo "Icon" Sutton for the lovely icon!
If you try it and it fails to work, tell me.
Regards,
Cédric.
Hi,
I'm doing more and more live stuff this days and personally and together in
a
band, I'm/we are using backingband played from some variations of music
players (stereo out).
I'm in a need of a player (or something that acts like a player) for
backing-
band and monitoring purposes(the music plus ticks and individual mixes and
so
on).
Does it exists a program and/or well established practice in Linux that can
do
this stuff in an easy, predictable and quick way when using a computer with
a
sufficent numbers of audio channels:
*Start a song with a foot pedal (space button on a keyboard
is good enought). This is start, stop pause through a play list.
*Providing several channels out (jack/soundcard) for
in house mixers, click track, individual monitor mixes and so on
Ordinary DAWs are probably not good for this, so any suggestions?
Thank you.
Jostein
Can I output all audio from one computer on another computer's audio
device? If so, is networking the two computers via ethernet and
running jack with RT priority the best way to do this with as little
latency as possible?
- Grant
On Mon, September 3, 2012 12:44 pm, SxDx wrote:
>> From: "Lorenzo Sutton" <lorenzofsutton(a)gmail.com>
>
>> >>>> From: "Ralf Mardorf" <ralf.mardorf(a)alice-dsl.net>
>> >>>> The East West Choirs sing your lyrics. It's not another Ahhhh or
>> >>>> Ohhhh
>> >>>> sample ;).
>> >>>>
>> >>>> http://www.youtube.com/watch?v=aI5Gg2-mhmU
>
>> I think you only need the resources/time/skill to sample all the
>> phonemes for various choir combinations, add glue and logic, the
>> interface, some usual ADSR stuff and filtering, reverb here and there, a
>> pinch of randomness, and your done :)
>>
>> Another way could be to have 4 singers sing the lines various times
>> recording them multi-trak, each take with different miking positions and
>> EQing (or keeping the mike still in the room and having them move around
>> each time). If the singers were good enough they'd be able to change
>> their voice quality slightly at each take to mimic multiple people. Of
>> course you need 4 singers if you wanted a full SATB choir, probably just
>> a good baritone and dark soprano would suffice to have a mixed choir
>> effect for a monophonic or 2-voice line.
>
> Maybe that's how they do it in the video. I gave a try to mbrola and the
> result is much more realistic than espeak. Mbrola uses recorded
> phonemes (if I understand correctly).
Yes, and that is the same technique the vocaloid uses too.
There has been a lot of funding and research undertaken by Inria and at
one point they even got a song onto the charts ;-)
Festival also has a singing mode and it can use espeak or mbrola.
I'm sure in a couple of years there will be a large database of singers
for the lauloid project. If we took it a step further lauloid could be
integrated with linuxsampler too.
--
Patrick Shirkey
Boost Hardware Ltd
I'm pleased to announce the 20120903 release of WhySynth, a DSSI
softsynth plugin.
New since my last release announcement:
* One new oscillator mode (Wavecycle Chorus) and four new filter modes
(resonz, high-pass and band-reject, with thanks to Luke Andrew).
* Some new patches.
* An icon for desktop use.
* A bug fix for an ugly click that would occur when using portamento in
monophonic mode.
* WhySynth development is now hosted on Github.
Find WhySynth here:
http://smbolton.com/whysynth.html
More information on the DSSI plugin standard, available hosts
and plugins can be found here:
http://dssi.sourceforge.net/
WhySynth is written and copyright (c) 2012 by Sean Bolton,
under the GNU General Public License, version 2.
> People have managed it in the past with varying degrees of success. The >
standard method these days is to use pulseaudio's jack-sink over dbus.
FWIW, I had epic fails with dbus in Ubuntu 12.04.
> > I don't know why people still insist on making their lives more
difficult > that necessary. Pulse works just fine as a jack client.
> because pulse is a dog, and its latency blows?
Different configurations are right for different use cases. I run pulse
through jack for daily use, but I wouldn't do that in a concert. For
instance, I might want to have Skype open in a normal session, and that
seems to play best with pulseaudio. But I wouldn't use it in a performance,
in which case it's simple to start the jack server but not pulse.
hjh