Hello again!
I though about my MIDI problem again and I concluded that the only mistake can be my alsa
version. Nothing else could hang up the system, right ?
Well, then I'd like to know howto use MIDI via the OSS-layer, because alsa-0.9.7 ( current version)
seems to need some "magic" to get started and I am MUCH impatient right now ;))))
Any tips, links or something interesting where I could start with ?
Howto say OSS that there is a MIDI-keyboard ? ANY applications uses MIDI-IN signals via OSS ?
greez, Sascha Retzki
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-------- directBOX Reply ---------------
From: david(a)olofson.net
To : linux-audio-dev(a)music.columbia.edu
Date: 10.10.2003 19:17:01
On Friday 10 October 2003 17.41, hexe_2003(a)directbox.com wrote:
[...]
> But well, If I synthize ( ? spelling) two signals and want to send
> them at the same time to the device, my first idea'd be:
> 1.) Open /dev/dsp, maybe O_NONBLOCK | O_WRONLY
> 2.) thread
> 3.) two threads calculate the both signals
> 4.) sync them
> 5.) two threads write their signals to /dev/dsp
> 6.) threads close
>
> certainly some of this. I mean mp3blaster, for example, have
> several pids while playing audio streams, and it is basicly
> OSS-programming and not alsa.
> Am I so wrong ? ;)
Yep! ;-)
Synths, trackers, games and other "polyphonic things" at some point
mix everything into a single stream that is sent to the audio device.
In general, all audio processing is done in a single thread, but in
some cases, it can be pretty hard to get the processing deterministic
enough that it can run in the audio thread without causing drop-outs.
This is especially troublesome if you need low latency, as in games
and other interactive applications.
This is probably why mp3blaster is using multiple threads; one for
each mp3 stream (decoding into raw audio data) and one for mixing and
output. There is most probably substantial buffering between the
decoding threads and the mixer thread, as this would allow solid
playback with less buffering (ie less laggy mixing controls) between
the mixer and the audio device.
//David Olofson - Programmer, Composer, Open Source Advocate
..- Audiality -----------------------------------------------.
| Free/Open Source audio engine for games and multimedia. |
| MIDI, modular synthesis, real time effects, scripting,... |
`-----------------------------------> http://audiality.org -'
--- http://olofson.net --- http://www.reologica.se ---
--------------
quiet interesting ;)
Could you give me a short idea howto mix two signals/stream/oscilators/whatever ??
If it is not pthread, what is it ?
Sascha Retzki
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-------- directBOX Reply ---------------
From: S.W.Harris(a)ecs.soton.ac.uk
To : linux-audio-dev(a)music.columbia.edu
Date: 13.10.2003 09:28:51
On Sun, Oct 12, 2003 at 01:30:52 +0200, hexe_2003(a)directbox.com wrote:
> But this wasn't your question ;) .
> jack does NOTHING with quality itsself, because jack is a user-space daemon other
> apps connect to and send data to, which is then outputted via the currently used audio-driver.
> And this is alsa ;)
Thats not /quite/ true, JACK will dither the output (if asked to) which
will improve the audio quality, and if the output of your program is going
to be used by another then JACKs float data will help.
Essentially though, it makes little difference.
- Steve
-----------
Woow, thats interesting! Well, then jack might be the best way to code your OGG-player ;)
Sascha Retzki
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-------- directBOX Reply ---------------
From: Vincent.Torri(a)math.u-bordeaux.fr
To : linux-audio-dev(a)music.columbia.edu
Date: 12.10.2003 12:30:38
Hello,
I am a newbie in the linux audio world and I have some stupid questions:
I want to wrote an ogg player with great quality playback. I've already
look at some other linux players and I've search this mailing list to
find some answers. It seems that two APIs are interesting: ALSA and Jack.
But i have not find some comparisons between these (and others) APIs
concerning the output quality (regardless the sound card).
Can someone give me some advice(s) about what API I should use
(comments, links, etc...) so that I could start working on it?
I would like to mention that I am not concerning about portability (only
linux platform) or easy programming, and I just want the best quality
output for this player.
I hope this is not the most stupid question of this mailing list :)
thank you
best regards
V.T.
------------
Well, ( like I understood it ) the quality of both are equal. If yes, you just have to know:
1.) jack ( http://jackit.sourceforge.net/ ) is easier to code, and the USER decide
which "options" are used, not your programme.
2.) alsa ( http://www.alsa-project.org/ ) is goes "deeper" into the hardware space,
your programme can decide which "options" are used and its sometimes harder
to code.
But this wasn't your question ;) .
jack does NOTHING with quality itsself, because jack is a user-space daemon other
apps connect to and send data to, which is then outputted via the currently used audio-driver.
And this is alsa ;)
( Or I think so - MY jack version ( Debian unstable ) does NOT have other possibilities,
my jack can only use alsa. Maybe this changed in newer versions! )
Hoped I helped
Sascha Retzki
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I'm searching for an audio API to parse a .wav file and show its waveform.
Someone knows something similar?
If not, someone could tell me how to do it by myself?
I mean if I open the file .wav and get one buffer at time, what I should do
to know its waveform?
Thank you,
J_Zar
Romanin Gianluca
Hello,
I am a newbie in the linux audio world and I have some stupid questions:
I want to wrote an ogg player with great quality playback. I've already
look at some other linux players and I've search this mailing list to
find some answers. It seems that two APIs are interesting: ALSA and Jack.
But i have not find some comparisons between these (and others) APIs
concerning the output quality (regardless the sound card).
Can someone give me some advice(s) about what API I should use
(comments, links, etc...) so that I could start working on it?
I would like to mention that I am not concerning about portability (only
linux platform) or easy programming, and I just want the best quality
output for this player.
I hope this is not the most stupid question of this mailing list :)
thank you
best regards
V.T.
Hi!
I bought a MIDI-keyboard a couple of days ago. It did not work correctly, and
I described my prob at lau. They told me that my ESS-Solo1 card is not able
to use MIDI ( under Linux or generally ?). Well, ok, I exchanged my card with my
fathers ( ES-1379 or simular ). Alsa supports it, I configured it and can listen to sounds.
If I start timidity in alsa mode ( "$ timidity -iA") and then attach my MIDI device with
timidiy via "$ acconnect 64 128", my system completely hangs!!! I must reboot via
the reset-button!! That's exactly the problem I had while I tried it ( the same way) with
my ESS-Solo1. Well, I am using Debian unstable, alsa version 0.9.0.beta12, have a
2.4.21-xfs kernel installed. Secondly, RG4-sequencer is NOT able to start MIDI and
I have no sound at all. Is it alsa's fault ? Is it mine ?
greez Sascha Retzki
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-------- directBOX Reply ---------------
From: magusofthedark(a)yahoo.com
To : linux-audio-dev(a)music.columbia.edu
Date: 11.10.2003 00:45:38
> >The prob. w/ portaudio is it abstracts to OSS, not
> >ALSA and I'd rather not take the least-common
> >denominator road.
>
> No. PortAudio abstracts to a callback model. Its
> closer to JACK than
> either OSS or ALSA. It isn't any closer to ALSA than
> it is to OSS.
I phrased that poorly. Say someone has ALSA and
doesn't have OSS compatibility. Would an app written
in portaudio work on their sys? If it doesn't, then
that's what I meant by that it doesn't abstract to
ALSA. If it does, then What I said was wrong, my bad.
--------
Like I understood portAudio's hp, portaudio is able to work with OSS.
Secondly, alsa is able to deal with OSS-apps because of an, well, OSS-emulation.
With other words OSS-applications are able to work with alsa, and they don't know
that they are working under alsa ;).
Ergo portAudio ( with it compatibility to OSS ) runs under OSS & ALSA and so
under Linux generally.
Sascha Retzki
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-------- directBOX Reply ---------------
From: paul(a)linuxaudiosystems.com
To : linux-audio-dev(a)music.columbia.edu
Date: 10.10.2003 17:37:43
>Hello !
>
>I've read somewhere that it is not possible that threads write different data to /
>dev/dsp.
>But how does Cheesetracker, for example, make it then?
i don't how cheesetracker does it, but it requires user-space
synchronization (e.g. pthread_mutex_(un)lock). with ALSA, you don't
need to to do this - any thread can write to the device at any
time. that's the theory, at least. in general, its probably a bad
design to be doing that, much like it is for GUI programming, where
its better to have a single thread mediate between the "device" and
the rest of the program.
------------------
But well, If I synthize ( ? spelling) two signals and want to send them at the same time
to the device, my first idea'd be:
1.) Open /dev/dsp, maybe O_NONBLOCK | O_WRONLY
2.) thread
3.) two threads calculate the both signals
4.) sync them
5.) two threads write their signals to /dev/dsp
6.) threads close
certainly some of this. I mean mp3blaster, for example, have several pids while playing audio streams, and it is basicly OSS-programming and not alsa.
Am I so wrong ? ;)
Sascha Retzki
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>OSS is dead. You should not be writing apps with OSS. If you want
>portability use PortAudio.
Yes, if heared many good stuff about PortAudio .... where is the homepage ?
Is it portable to NetBSD ( NetBSD is my "2nd OS" ;) )
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