Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread Frank Barknecht

Juhana Sadeharju hat gesagt: // Juhana Sadeharju wrote:

 I tried to contribute my developments to Snd, but heard nothing back
 from its author. Not a thanks, nothing. If you're not able to
 suggest and develop features to the editor, it is not that good
 for the _user_. 

I am not involved in SND development, but reading Dave Phillips'
SND/CoolEdit articles on O'Reilly Net, I get the impression, that
SND is moving in a more user friendly direction.

 If the editors would be placed on the shelf because they look good,
 then I would have no complaints; however, editors are really meant
 as tools for artists/users.

I don't get that: Are you saying, that SND looks good? It's a joke, isn't
it ;)

Ciao,
-- 
 ____
 Frank Barknecht    __    __ trip\ \  / /wire __
  / __// __  /__/ __// // __  \ \/ /  __ \\  ___\   
 / /  / /  / /  / // // /\ \\  ___\\ \  
/_/  /_/  /_/  /_//_// /  \ \\_\\_\
/_/\_\ 



Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread Nick Bailey

On Tuesday 23 Oct 2001 8:18 am, Frank Barknecht wrote:
 Juhana Sadeharju hat gesagt: // Juhana Sadeharju wrote:
  I tried to contribute my developments to Snd, but heard
  nothing back from its author. Not a thanks, nothing. If
  you're not able to suggest and develop features to the
  editor, it is not that good for the _user_.

 I am not involved in SND development, but reading Dave
 Phillips' SND/CoolEdit articles on O'Reilly Net, I get the
 impression, that SND is moving in a more user friendly
 direction.

I think that must have been an oversight, Juhana: I 
contributed a really, really dirty hack to do (very limited) 
esd compatibility, and got more thanks than I deserve.  Maybe 
some email got lost in a delete-fest.  I do that from time to 
time 8-)

Nick/



Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread delire


 I don't get that: Are you saying, that SND looks good? It's a joke,
isn't
 it ;)


It does look good ; ) As an editor I equate it with SoundEdit for the Mac.
However I've never found it to be comprehensive or flexible enough to
satisfy projects that requiring deeper editing. EG: broad sample and
bit-rate conversion with noise shaping and dithering, filter preview,
infinite undos, high bit-rate non-destructive editing, spectral analysis,
midi time-code import, extensive right click menus and a comprehensive
multitrack studio with panning and amplitude envelopes and broad mixdown
capability.
Cool-Edit Professional for windows, this is a fine editor - a good benchmark
for developments under linux. So far GSMP looks like a good
contender..though I've only spent a day with it so far

de|

Interactive Information Institute
R.M.I.T
Melbourne
Australia




Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread Bill Schottstaedt

 I tried to contribute my developments to Snd, but heard nothing back
 from its author.

This is a lie -- I never received anything from you except a copy of
some complaints you sent to SoundForge.



Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread Dave Phillips

delire wrote:

[re: Snd]

 It does look good ; ) As an editor I equate it with SoundEdit for the Mac.
 However I've never found it to be comprehensive or flexible enough to
 satisfy projects that requiring deeper editing. EG: broad sample and
 bit-rate conversion with noise shaping and dithering, filter preview,
 infinite undos, high bit-rate non-destructive editing, spectral analysis,
 midi time-code import, extensive right click menus and a comprehensive
 multitrack studio with panning and amplitude envelopes and broad mixdown
 capability.

I guess you haven't seen it lately. Certainly the undo can be as deep as
you prefer and spectral analysis features have been there for a while.
Editing is non-destructive, even at 32 bits (unless I'm not
understanding you correctly ?). Right-click popup menus are now
available for whole file and selected area. Panning and amplitude
enveloping is also available (always has been, I think).

O'Reilly Network recently published my status report on my work with
Bill Schottstaedt to externalize more of Snd's possibilities. We've
added dozens of GUI components for effects (Snd and LADSPA), cursor
control, popup menus, and so forth. If you're interested in the O'Reilly
article you can check it out here:

http://linux.oreillynet.com/pub/a/linux/2001/10/05/snd_partone.html

http://linux.oreillynet.com/pub/a/linux/2001/10/05/snd_parttwo.html

MIDI support is on Bill's TODO list. IMO, the issue of multitrack
recording seems better left to dedicated multitrack recorders (Ardour,
ecasound). Snd is an editor, that's what it aims to do and that's all it
does.

 Cool-Edit Professional for windows, this is a fine editor - a good benchmark
 for developments under linux. So far GSMP looks like a good
 contender..though I've only spent a day with it so far

I've been spending more time with other Linux audio editors, including
GSMP, Audacity, DAP, and others. I'm still inclined to keep working to
expose more of Snd's power rather than contribute to another project
either stalled or coming in at version 0.0.1a. Just my preference, of
course...

Best regards,

== Dave Phillips

The Book Of Linux Music  Sound at http://www.nostarch.com/lms.htm
The Linux Soundapps Site at http://sound.condorow.net

Currently listening to: O presul vere (Hildegard von Bingen)



Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread Paul Davis

MIDI support is on Bill's TODO list. IMO, the issue of multitrack
recording seems better left to dedicated multitrack recorders (Ardour,
ecasound). Snd is an editor, that's what it aims to do and that's all it
does.

   [ ... ]

I've been spending more time with other Linux audio editors, including
GSMP, Audacity, DAP, and others. I'm still inclined to keep working to
expose more of Snd's power rather than contribute to another project
either stalled or coming in at version 0.0.1a. Just my preference, of
course...

Yay! for sanity. Most people have no idea what Snd is capable of. So
to fix it, they sit down and write another editor. Dave is
encouraging/helping Bill to do a much more sensible thing: exposing
the things Snd can do so that you're less likely to run off and do
this.

I claim an exemption for myself, having spent quite a bit of time deep
inside Snd as I attempted to use it as the editor for Ardour. As Dave
notes, and others would do well to heed, editing an audio file is one
thing, multichannel work and/or audio sequencing is something else.

--p



Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread delire

just trying to install GSMP on a suse 7.1 box at another studio and
./configure produces the error:

cannot find STL file sstream

i've never come across this before..any solutions?
has all the same gtkmm / alsa / and libsigc++ libs etc as the debian box it
compiled successfully on..

de|




Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread Juhana Sadeharju

From:  Bill Schottstaedt [EMAIL PROTECTED]

 I tried to contribute my developments to Snd, but heard nothing back
 from its author.

This is a lie -- I never received anything from you except a copy of
some complaints you sent to SoundForge.

I have not sent any complaints to SoundForge. The two mails I
mailed both to you and to David described my observations on
what features an editor needs for being usable (for editing
audio files I prefer to edit) --- with my wish that those features
would find their way to Snd before I have a change to move to Snd.

Also, pointing out problems and giving a solution is not a complaint.
AND, I did _not_ get any reply from you. Does the same complaints
apply to Snd too?

Lets get the ball rolling: here are a couple of features needed for
succesful basic editing (no need to reread my mails!):

 -Software volume (up to +64 dB, say); this feature is needed
  for listening cut points between quiet fades, but also in
  noise reduction software where one needs to find the background-
  noise-only areas (i.e., the noise floor) [ as discussed here,
  Wavelab implements this with a plug-in at output path; good idea ]

 -Play feature where the region between the mouse pointer and the
  nearest edge of the selection is played; this makes it possible
  to play the ends of the selection, and check if anything important
  was accidentally left outside the selection [such a feature is
  in XWave2 (which is my version of XWave)]

 -Recording dropouts marked as red colored regions to waveform
  display so that one can see both if dropouts occured and if
  a dropout landed on the important part of the take; a close
  encounter with Alsa needed

SoundForge misses both two first features which makes it impossible
(IMHO) to make accurate edits. But I repeat my question: how people
do it without those features? I'm puzzled. (Actually, a friend
mastering CDs professionally turned volumes high up when listening
cut points near fades; occasionally he forgot to turn the volume
down, eek --- doesn't make good to speakers, nor ears.)

Bill, what is your opinion on people who don't contribute code
but only feature ideas and design? I just want to make sure
I don't waste time here.

Best regards,

Juhana



Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread Dave Phillips

Emiliano Grilli wrote:
 
 I read part one of your tutorial and found it *very* interesting. Thank you
 also for your site, which is a cornerstone in my bookmarks.
 Unfortunately, I can't find the part two of the snd tutorial, and the link
 you provided in this email seems to be broken.
 Please tell me the correct URL of the article, if you have time...

My bad, sorry. Here's the URL for Part 2 of my article:

http://linux.oreillynet.com/pub/a/linux/2001/10/18/snd_parttwo.html

Best regards,

== Dave Phillips

The Book Of Linux Music  Sound at http://www.nostarch.com/lms.htm
The Linux Soundapps Site at http://sound.condorow.net

Currently listening to: Vos flores rosarum (Hildegard von Bingen)



Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread Rene Rebe

Hi.

On Tue, 23 Oct 2001 23:18:49 +1000
delire [EMAIL PROTECTED] wrote:

 just trying to install GSMP on a suse 7.1 box at another studio and
 ./configure produces the error:
 
 cannot find STL file sstream
 
 i've never come across this before..any solutions?
 has all the same gtkmm / alsa / and libsigc++ libs etc as the debian box it
 compiled successfully on..

This is a check if the STL (Standard Template Library) of C++ is up to date.
It seems that it is at a wrong location on this SuSE system (or not all
c++ development packages are installed ??)

(Since I use ROCKLinux (www.rocklinux.org) I expected such compile errors
and I like to be cc'ed in private, too ...)

 de|

k33p h4ck1n6 René

-- 
René Rebe (Registered Linux user: #127875)

eMail:[EMAIL PROTECTED]
  [EMAIL PROTECTED]

Homepage: http://www.rene.rebe.myokay.net/

Anyone sending unwanted advertising e-mail to this address will be
charged $25 for network traffic and computing time. By extracting my
address from this message or its header, you agree to these terms.



Re: [linux-audio-dev] low latency + mp3

2001-10-23 Thread Jelle

On Mon, Oct 22, 2001 at 10:34:11PM +1000, David Burrows wrote:
 
 The bonus question is about pitch control.  I understand that this can be
 achieved by simply changing the sampling rate, however, I'm wondering if
 anyone has knowledge of fast or high quality resampling algorithms?

I think you can do spectral encoding with loads of bands to get very
high quality resampling plus you can do all kinds of other funky stuff
with it. however, this technique is nearly impossible to do realtime on
std computer hardware, atleast with a load of bands.

to do spectral encoding you seperate the freq spectrum into a load of
bands and analize the sample and record the amplitude for each band at
time t. then you can play back that recording resynthesizing the
original sample using sine waves but with a altered speed. you can stop
time and stuff like that.

i suppose bspline interpolation works equally well ;)

cheers
-- 
jelle herold (defekt)   http://defekt.nl/
seeing digital  http://channelthree.net/



[linux-audio-dev] Audio streaming over network

2001-10-23 Thread Ryan Mitchley

Hi all

Does anyone here know of a library or API for streaming audio over a
network? The library should take into account the inevitable clock drift
between the machine generating the stream and the machine receiving it. I
presume this would involve resampling/reclocking of some kind.

Thanks for all replies!

Ryan





Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread Rene Rebe

Hi.

On Tue, 23 Oct 2001 07:01:20 -0700
Bill Schottstaedt [EMAIL PROTECTED] wrote:

  IMO, the issue of multitrack
  recording seems better left to dedicated multitrack recorders (Ardour,
  ecasound).
 
 I agree completely -- I haven't had time yet to try out ecasound,
 but Fernando showed me Ardour and it is beautiful.  I'm very tempted
 to remove the Record option from Snd!  As a bit of boring history,
 that dialog dates back to the SGI days, and was ported to Linux
 at a time when Soundblaster cards were about as good as you could
 get; I was trying at the time to fill an obvious need.

Some short points about why I developed another Audio Editor (and to
end the discussion why another one).

1. I started to play with such stuff in the i386 times under DOS
2. When I started to use Linux (3? years ago) there was no audio
   editor solution
3. All the other applicatoins that came up during this time had a
   few probelms:
   a) not powerfull enough
   b) not useable

The exceptions are at least Ardour and snd. Ardour never compiled on my system
and targets another area (seems to be everything-in-one-place(tm) hard-disk-
recorder emulation). SND was one of the not-powerfull/not-useable tools
some years ago and I took a look onit a few month ago - and I gave up compiling
it.

gcc -c -DHAVE_CONFIG_H -g -O2 -I/usr/include -I/usr/X11R6/include  clm.c
clm.c:40: gsl/gsl_complex_math.h: No such file or directory
In file included from clm.c:5955:
/usr/include/gsl/gsl_sf_bessel.h:8: gsl_mode.h: No such file or directory
/usr/include/gsl/gsl_sf_bessel.h:9: gsl_precision.h: No such file or directory
/usr/include/gsl/gsl_sf_bessel.h:10: gsl_sf_result.h: No such file or directory
make: *** [clm.o] Error 1

(Yes I have a development system (and my whole distribution compiles on this
system!) - and the only programs I have compile prolems with, were Ardour and
SND ... ?!)

We would have contributed to a project - if there would have been one (not C
hacked) one year ago.

GSMP would have been released last autumn - if the book C++ TPL from Bjarne
Stroustrup hadn't sliped through my hands ...

IMO this is the most interesting point on GSMP. Is is fully obejct-oriented, features
a very cool plugin and activator system, many GUI componets are generated
dynamically ...

It is not just another wave-editor, and we WILL continue developing it.

(BTW: We have already produced a CD with it ;-)

k33p h4ck1n6 René

-- 
René Rebe (Registered Linux user: #127875)

eMail:[EMAIL PROTECTED]
  [EMAIL PROTECTED]

Homepage: http://www.rene.rebe.myokay.net/

Anyone sending unwanted advertising e-mail to this address will be
charged $25 for network traffic and computing time. By extracting my
address from this message or its header, you agree to these terms.



RE: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread STEFFL, ERIK *Internet* (SBCSI)

 -Original Message-
 From: delire [mailto:[EMAIL PROTECTED]]
 
 just trying to install GSMP on a suse 7.1 box at another studio and
 ./configure produces the error:
 
 cannot find STL file sstream
 
 i've never come across this before..any solutions?
 has all the same gtkmm / alsa / and libsigc++ libs etc as the 
 debian box it
 compiled successfully on..

  on a debian unstable:

jojda:~locate sstream
/usr/include/g++-3/sstream
jojda:~dpkg -S sstream
libstdc++2.10-dev: /usr/include/g++-3/sstream
jojda:~

erik



Re: [linux-audio-dev] Audio streaming over network

2001-10-23 Thread M. Edward (Ed) Borasky

On Tue, 23 Oct 2001, Ryan Mitchley wrote:

 Hi all

 Does anyone here know of a library or API for streaming audio over a
 network? The library should take into account the inevitable clock drift
 between the machine generating the stream and the machine receiving it. I
 presume this would involve resampling/reclocking of some kind.

 Thanks for all replies!

 Ryan

Check out sfront at

http://www.cs.berkeley.edu/~lazzaro/sa/

--
[EMAIL PROTECTED] (M. Edward Borasky) http://www.aracnet.com/~znmeb
Relax! Run Your Own Brain with Neuro-Semantics!
http://www.aracnet.com/~znmeb/Flyer.htm

What phrase will you *never* hear Miss Piggy use?
You can't make a silk purse out of a sow's ear!




Re: [linux-audio-dev] Audio streaming over network

2001-10-23 Thread John Lazzaro

 On Tue, 23 Oct 2001, Ryan Mitchley wrote:

 Hi all

 Does anyone here know of a library or API for streaming audio over a
 network?

 M. Edward (Ed) Borasky writes

 Check out sfront at

 http://www.cs.berkeley.edu/~lazzaro/sa/


Soon, but not quite yet -- at the moment, sfront networking does
MIDI resilently for low-latency situations, and while you could
concievably hack this to do audio (sending samples encoded as
MIDI change-control events, and reassmbling on the other side),
you really don't want to go there.

I'm actively working on SASL networking for sfront at the moment --
SASL is the more general-purpose control language for Structured
Audio, provided as a companion to MIDI control. SASL is well 
suited for writing custom audio codecs in Structured Audio, so
once the SASL packetization is ready, sfront should be a viable
platform for these sorts of experiments. But not for another few
months ... 

--jl



Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread D. Stimits

Rene Rebe wrote:
 
 On Tue, 23 Oct 2001 13:47:34 -0600
 D. Stimits [EMAIL PROTECTED] wrote:
 
 [...]
 
  If it is a matter of location, use locate g++-3/sstream to find it
 
 It shouldn't be a matter of location. We use:
 
 configure.in: AC_CHECK_HEADER(sstream,,AC_MSG_ERROR(missing STL file sstream))
 and in the sources: #include sstream
 
 So his Linux system hasn't one in the paths the cpp searches through.
 - So there is even no STL on it ?? (or an really old one containing
 sstream.h??)

Until recently, there wasn't *any* sstream. He can have STL, but
upgrades have to be done on Redhat 6.2 and earlier to get this
particular one. I guess the key is that if there is an include
subdirectory g++-3 then it should be there. Before this version,
sstream did not exist on Linux.

D. Stimits, [EMAIL PROTECTED]

 
  (run updatedb if needed before this).
 
  D. Stimits, [EMAIL PROTECTED]
 
 k33p h4ck1n6 René
 
 --
 René Rebe (Registered Linux user: #127875)
 
 eMail:[EMAIL PROTECTED]
   [EMAIL PROTECTED]
 
 Homepage: http://www.rene.rebe.myokay.net/
 
 Anyone sending unwanted advertising e-mail to this address will be
 charged $25 for network traffic and computing time. By extracting my
 address from this message or its header, you agree to these terms.



Re: [linux-audio-dev] Audio streaming over network

2001-10-23 Thread M. Edward (Ed) Borasky

On Tue, 23 Oct 2001, John Lazzaro wrote:

 Soon, but not quite yet -- at the moment, sfront networking does
 MIDI resilently for low-latency situations, and while you could
 concievably hack this to do audio (sending samples encoded as
 MIDI change-control events, and reassmbling on the other side),
 you really don't want to go there.

 I'm actively working on SASL networking for sfront at the moment --
 SASL is the more general-purpose control language for Structured
 Audio, provided as a companion to MIDI control. SASL is well
 suited for writing custom audio codecs in Structured Audio, so
 once the SASL packetization is ready, sfront should be a viable
 platform for these sorts of experiments. But not for another few
 months ...

Sorry ... I guess I was getting ahead of you :-). I just wish there were
more publicly available SASL-controlled instruments available. Almost
everything I've found so far uses MIDI control, which is a pain to hack
around.
--
[EMAIL PROTECTED] (M. Edward Borasky) http://www.aracnet.com/~znmeb
Relax! Run Your Own Brain with Neuro-Semantics!
http://www.aracnet.com/~znmeb/Flyer.htm

What phrase will you *never* hear Miss Piggy use?
You can't make a silk purse out of a sow's ear!




Re: [linux-audio-dev] W64 file format

2001-10-23 Thread ljp

On Tuesday 23 October 2001 04:00 pm, Erik de Castro Lopo wrote:
 Hi People,
 I currently have one example of this file format and I need more. The
 single file I have is a 16 bit stereo file. I need some more files and what
 I'm looking for (if they are available) is as follows:

- a mono file of any bitwidth
- files containing more than 2 channels
- files containing 8, 24 or 32 bit PCM data
- files containing float or double data
- files containing looping and other information
- files containing encoded data (ie not PCM nor float/double)

 If anybody wishes to help this Free Software project by supplying examples
 files I would be very pleased to hear from them. Please don't email files
 to me without asking first as I would like to prevent my mail box from
 overflowing with multiple copies of the same file.

I can send you some, it appears that theres no support for multi channel 64 
bit wave files.

Looks like these are supported :

ACELP.net
A-law
U-law
True Speech
GSM6.10
IAC2
IEEE float (uncompressed)
IMA ADPCM
LH CELP 4.8kbit/s
LH SBC 12
LH SBC 16LH SBC 8
MS ADPCM
MS G.723.1
mp3
PCM


which ones do you want? What length?
Can't do multi channel

ljp



Re: [linux-audio-dev] multitrack and editor separate?

2001-10-23 Thread delire

As Dave
 notes, and others would do well to heed, editing an audio file is one
 thing, multichannel work and/or audio sequencing is something else.


...sure, but Cool-Edit Professional for windows shows how often a static
mixing [multitrack] environment is used as an intergral part of the overall
editing project - you make a mix of several samples and dump it back in the
editor for further processing. maybe afterwards you send it back to the
multitracker to layer it up with other samples.
sequencing, well that is for a very particular kind of composition.

syntrillium [producer] obviously recognises that a multitracker should in
fact come with the editor, and does a very fine job of making this work.
in work that i do, for instance [and i am not an exception] , it isn't
uncommon to have 70 or 80 samples open simultaneously in a composition; by
being able to double-click a wave in the multitrack window to take me over
to the waveform where i edit it further [without saving] and then return to
the multitrack, provides for a highly efficient working environment. having
to open up the file in a separate editor, then save it, then re-open it up
in the multitrack makes larger compositional projects, like electroacoustic
work, special fx and film scores very difficult. they don't have to be
together in the same window [as in SoundEdit16 on the mac for instance], so
much as a click apart; as in many other application environments where you
have eg: a 'objects' [editor] and 'scenes' [multitrack].

for this reason the ABC in my country, and several universities are
replacing other professional editing/multitrack packages with Cool-Edit
Pro..it is testimony that there is a wide demand for an editor and
multitracker to be put together in an intuitive relation.

if you are, however, lucky enough to be one of those rare composers that can
produce all their samples first before taking them over to the
multitracker - then a separation of these two parts of the composition
process is not a problem. for the rest of us howver, composition is so often
about trying out arrangements of timbres, textures, acoustics, amplitudes,
harmonics and noise beds during the working process. for this reason a
multiracker / editor combination is ideal.


de|

Interactive Information Institute
R.M.I.T
Melbourne
Australia




Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread delire

fair enough, must admit i haven't looked at snd in quite a while... what's
promised in the o'reilly articles looks amazing - i'll download the latest
version and check it out.

de|


- Original Message -
From: Dave Phillips [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, 23 October 2001 10:46
Subject: Re: [linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1



 I guess you haven't seen it lately. Certainly the undo can be as deep as
 you prefer and spectral analysis features have been there for a while.
 Editing is non-destructive, even at 32 bits (unless I'm not
 understanding you correctly ?). Right-click popup menus are now
 available for whole file and selected area. Panning and amplitude
 enveloping is also available (always has been, I think).






[linux-audio-dev] alsa latency statistics

2001-10-23 Thread dave willis

i just finished testing several different patched and unpatched linux
kernels, and was surprised by the results:  all basically the same.

alsarange = 33.4% to 33.8%1.336 ms to 1.352 msdiff = 0.016 ms
oss-emu range = 91.6% to 91.9%3.664 ms to 3.676 msdiff = 0.012 ms

as you can see, the differences are negligible and probably irrelevant if
more tested were done, and a standard 2.4.12 kernel is just as good as any
patched kernel.

one note:  when i did a real-world test of latency by looping a signal
through my card and dividing the difference in time by 2 (because of input
and output latency), i got 3.675 ms latency using kernel 2.4.8-ll with oct
1 alsa-cvs, and that corresponds with the oss-emu latency for some reason
instead of the alsa latency.  i don't know why, and i'll test this on
other kernels soon.

my system is an amd athlon 900 mhz running at 1006 mhz with redhat 7.1 and
an  m-audio delta audiophile 2496 soundcard.  i used alsa-cvs from october
20 and takashi iwai's modified version of latencytest0.42+alsa.  the
kernels tested are 2.4.8-ll, 2.4.9-ll, 2.4.10-ll, 2.4.10-pe, 2.4.12,
2.4.12-ac2, 2.4.12-ac2-ll, 2.4.12-ac3, and 2.4.12-ac3-pe.
ll = low-latency
pe = pre-emptive
ac2 and ac3 are alan cox's patches

latencytest was ran as follows:
./latencytest -t alsa -d hw -q none 3 256
./latencytest -t oss -q none 3 256

-dave

details below:
---
sample rate = 48000
NUMBER of OVERRUNS = 0  PIXEL_PER_MS=112
fragment latency = 1.33 ms  buffer latency= 4.000 ms
fragment size=256   total buffer size=768
cpu_load= 80.%  cpu latency = 1.07 ms
with alsa, 100% of samples are within 1ms of buffer
used alsa-cvs from oct 20, oss = alsa's oss-emu

2.4.10-ll
output type = ALSA max latency=1.3 ms  factor=33.4 % of buffer

output type = OSS max latency=3.7 ms  factor=91.6 % of buffer
1MS factor=83.4679932MS factor=99.937850

2.4.10-pe
output type = OSS max latency=3.7 ms  factor=91.9 % of buffer
1MS factor=83.3676982MS factor=99.965636

output type = ALSA max latency=1.3 ms  factor=33.6 % of buffer

2.4.12-ac2-ll
output type = ALSA max latency=1.3 ms  factor=33.7 % of buffer

output type = OSS max latency=3.7 ms  factor=91.7 % of buffer
1MS factor=83.4008102MS factor=99.932524

2.4.12-ac2
output type = ALSA max latency=1.3 ms  factor=33.4 % of buffer

output type = OSS max latency=3.7 ms  factor=91.7 % of buffer
1MS factor=83.3729222MS factor=99.966067

2.4.12-ac3
output type = ALSA max latency=1.4 ms  factor=33.8 % of buffer

output type = OSS max latency=3.7 ms  factor=91.7 % of buffer
1MS factor=83.3613922MS factor=99.971942

2.4.12-ac3-pe
output type = ALSA max latency=1.3 ms  factor=33.5 % of buffer

output type = OSS max latency=3.7 ms  factor=91.7 % of buffer
1MS factor=83.3902162MS factor=99.943117

2.4.12
output type = ALSA max latency=1.3 ms  factor=33.4 % of buffer

output type = OSS max latency=3.7 ms  factor=91.7 % of buffer
1MS factor=83.3728282MS factor=99.960506

2.4.8-ll
output type = ALSA max latency=1.3 ms  factor=33.6 % of buffer

output type = OSS max latency=3.7 ms  factor=91.6 % of buffer
1MS factor=83.4028362MS factor=99.972199
2.4.9-ll
output type = ALSA max latency=1.3 ms  factor=33.4 % of buffer

output type = OSS max latency=3.7 ms  factor=91.7 % of buffer
1MS factor=83.3850932MS factor=99.948240




Re: [linux-audio-dev] multitrack and editor separate?

2001-10-23 Thread Paul Davis

...sure, but Cool-Edit Professional for windows shows how often a static
mixing [multitrack] environment is used as an intergral part of the overall
editing project - you make a mix of several samples and dump it back in the
editor for further processing. maybe afterwards you send it back to the
multitracker to layer it up with other samples.
sequencing, well that is for a very particular kind of composition.

syntrillium [producer] obviously recognises that a multitracker should in
fact come with the editor, and does a very fine job of making this work.

except that this is linux, where fork(2) is cheap, and IPC is the most
efficient available.

what reason is there for requiring a user to use *your* sample editor
when your main focus was on multitrack editing and sequencing? why not
let the user choose a 3rd party editor? suppose i prefer bias peak to
cool edit for sample editing, for example?

in work that i do, for instance [and i am not an exception] , it isn't
uncommon to have 70 or 80 samples open simultaneously in a composition; by
being able to double-click a wave in the multitrack window to take me over
to the waveform where i edit it further [without saving] and then return to

yep. just fork snd, or audacity, or gsmp, or sweep, or gnoise, or DAP
or whatever you want. 

the downside is that you don't get non-destructive editing. see, the
dedicated waveform editor in your windows tool isn't really doing
sound *file* editing, its doing the same kind of non-destructive
editing that the rest of the program is doing (i.e. just rearranging
playlists).

for this reason the ABC in my country, and several universities are
replacing other professional editing/multitrack packages with Cool-Edit
Pro..it is testimony that there is a wide demand for an editor and
multitracker to be put together in an intuitive relation.

the relationship, as you noted, has more to do with 1 click away
than it does with anything else.

however, i would agree that it is harder to make them work together if
the relationship has to be based around a stored audiofile rather than
a playlist/EDL.

personally speaking, i might just hack this feature in ardour tonight :)

--p



[linux-audio-dev] What version of Broadcast that is still available is the least buggy?

2001-10-23 Thread Ivica Bukvic

Hi, I've recently installed Mdk 8.1 and Bcast2000 (c version) is having
problems displaying waveforms (it is completely out of whack displaying it
way beyond the reasonable range, so I need to zoom out the y axis all the
way in order to get somewhat of a visible waveform). Is this the case with
all of the versions or is there a version that has this working properly. In
another words, which version is the best out there (since it has been pulled
I am looking for the best remaining version)? Thanks for your help!
Sincerely,

Ico Bukvic


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Andre Pang
Sent: Sunday, October 21, 2001 3:37 AM
To: [EMAIL PROTECTED]
Subject: Re: [linux-audio-dev] preemptive kernel patch


On Sat, Oct 20, 2001 at 02:33:09PM -0500, dave willis wrote:

 where can i get the preemptive kernel patch?  i've tried searches but get
 everything but...

http://www.tech9.net/rml/linux/


--
#ozone/algorithm [EMAIL PROTECTED]  - trust.in.love.to.save





[linux-audio-dev] RE: What version of Broadcast that is still available is the least buggy?

2001-10-23 Thread Ivica Bukvic

Crap, please disregard this last post, it seems I was trying to import a
non-normalized file (DOH!). Sorry for cluttering your mailboxes!

-Original Message-
From: Ivica Bukvic [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, October 24, 2001 12:21 AM
To: [EMAIL PROTECTED]
Subject: What version of Broadcast that is still available is the least
buggy?


Hi, I've recently installed Mdk 8.1 and Bcast2000 (c version) is having
problems displaying waveforms (it is completely out of whack displaying it
way beyond the reasonable range, so I need to zoom out the y axis all the
way in order to get somewhat of a visible waveform). Is this the case with
all of the versions or is there a version that has this working properly. In
another words, which version is the best out there (since it has been pulled
I am looking for the best remaining version)? Thanks for your help!
Sincerely,

Ico Bukvic


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Andre Pang
Sent: Sunday, October 21, 2001 3:37 AM
To: [EMAIL PROTECTED]
Subject: Re: [linux-audio-dev] preemptive kernel patch


On Sat, Oct 20, 2001 at 02:33:09PM -0500, dave willis wrote:

 where can i get the preemptive kernel patch?  i've tried searches but get
 everything but...

http://www.tech9.net/rml/linux/


--
#ozone/algorithm [EMAIL PROTECTED]  - trust.in.love.to.save





[linux-audio-dev] Re: [Alsa-user] gsmp release 0.0.1

2001-10-23 Thread Frank Barknecht

I wrote:

 You really should announce this on linux-audio-dev, too. 
Uhm, you did. Procmail outsmarted me.
-- 
 Frank Barknecht  



[linux-audio-dev] anyone using code based on audioengine

2001-10-23 Thread Paul Davis

i just uncovered a subtle bug in the audioengine inner loop.
it might not affect you, but then again, if it does, it will be nasty.

when i wrote audioengine, the idea was to guarantee that all clients
would never be asked to process more frames than was specified in the
last call to their set_block_size() method. unfortunately, the engine
does not honor this guarantee: in the event of a scheduling delay
which causes us to be notified of N periods worth of data/space at
once, rather than a single period's worth, the clients are asked to
process it all at once, instead of subdividing into N calls to
process().

i have fixed the sources, and a fix will be in CVS tonight. it would
be worth thinking about whether any code you might be using that was
based on this stuff is affected. i know that i have to fix JACK as
well, for identical reasons. i'll put out a new version of JACK
tomorrow.

--p