David Kastrup writes:
> On another note: it would seem like a simple enough task to let the
> recording start with the first non-zero sample. ... I'd also like to prune
> trailing zeros at some point of time.
Although it is not a native ALSA app (due mainly to me not having the time
to port it yet), batchrec does this. It still works with ALSA through the
OSS emulation layer; adding native ALSA support is on my todo list for the
next few months. You can grab it from
http://www.physics.adelaide.edu.au/~jwoithe/batchrec-1.2.0.tgz
It's pretty raw (no configure script yet) but shouldn't be too hard to
compile on most systems.
Batchrec can be requested to wait for signal above a certain amplitude
before it records. It will also stop recording after the same threshold has
been present for a user-specified time (if desired); at that point recording
will stop, the file will be closed and the program will then wait for signal
above the threshold again. I've used it extensively with analog inputs
(hense the use of a threshold rather than exactly 0). The only fly in the
ointment will be whether the SPDIF input on your soundcard can be mapped in
the OSS emulation layer in such a way that OSS applications can use it.
Regards
jonathan
Hello,
All I am trying to do is compile an x86_64 kernel on i386 machine. It
dos not work:
$ ARCH=x86_64 make oldconfig
<no errors>
$ ARCH=x86_64 make
CHK include/linux/version.h
SPLIT include/linux/autoconf.h -> include/config/*
CC arch/x86_64/kernel/asm-offsets.s
arch/x86_64/kernel/asm-offsets.c:1: error: code model 'kernel' not
supported in the 32 bit mode
make[1]: *** [arch/x86_64/kernel/asm-offsets.s] Error 1
make: *** [prepare0] Error 2
What gives? Do I need a different compiler to build for x86_64?
Shouldn't this Just Work?
Lee
Hi everyone,
Can anyone give me any examples of Free audio software being used by
professionals?
Anywhere where it performs better, or simply doesn't cost two or more
body parts to use?
Quick answers get bonus points.
James
--
"I'd crawl over an acre of 'Visual This++' and 'Integrated Development
That' to get to gcc, Emacs, and gdb. Thank you."
(By Vance Petree, Virginia Power)
After several frustrating weeks playing with my Audiophile USB I'mgoing to punt and get something else.
For the archives: ALSA users interested in high-end capture should_not_ buy this device.(FWIW it sucks in windows too)
- Its only a USB 1.1 device and 24bit 96k is half-duplex only.- Its (almost) S24_3BE so you have to use plughw or jackd has to bepatched to use it. I say almost because the few times I did get thecard to record something I had to reorder the byte data to get it notto sound like white noise.- As of ALSA 1.0.10 capture is broken.
So I'm looking for a replacement. I bought this card because it had areally good recording specs (I just missed the half-duplex part) and Idont' need a pro-multi channel since all I'm tyring to do is recordLPs to CD.
Can someone recommend a 24bit 96kHz full-duplex card that they thinkhas good enough recording specs for LP's that not going to cause me topull all my hair out trying to work with ALSA.
--Richard A. Smith
Minor bug fixes since last release.
Notable changes:
* Attempt to let client timeout option work again on realtime mode
* Let jackd quit gracefully when USB soundcard or power cable is
unplugged
* Better support for US428 USB soundcard
* SSE/E3DNow mixing support. Disabled by default. Enable with --
enable-dynsimd
JACK is available at http://jackit.sf.net and our new website http://
jackaudio.org.
Hi,
I have two questions concerning the jack callback,
1. what is the preferred way of feeding data from disk to the callback?
Is there a general design pattern agreed upon? Best Practices?
2. What is the preferred way to notify the non-realtime thread that
something happened in the jack-callback?
A condition variable? How to avoid blocking? How do you do it?
I read the tutorial at http://userpages.umbc.edu/~berman3/ , it uses
mutex+condition, is it okay to do this? Are there better ways?
thx
-Richard
Hi all,
some weeks ago I sent some mails ranting about problems getting my
tascam us-122 to run on a turion64 running gentoo.
As time has gone and new kernels arrived I have reached a state where
I can play (synths, effects, ardour, etc) and "record" ie. sent the
incoming audio to effects and back out.
But: If I want to record to disk I get the same old error (should not
be here with counts=0) again and sound stops completely. The computer
freezes if I try to restart jackd...
Some tests showed that this doesn't happen if I record into ram (using
/dev/shm to save the recordings).
I modified the driver to give some more debug and to not stop at the
first error (see the attached usbusx2yaudio.c for details) but still
after some more function-calls the driver stops working. Sadly there
isn't much documentation in the source so I am a bit lost there.
Can someone give my some hints?
Thanks in advance,
Arnold
--
visit http://dillenburg.dyndns.org/~arnold/
---
Wenn man mit Raubkopien Bands wie Brosis oder Britney Spears wirklich
verhindern könnte, würde ich mir noch heute einen Stapel Brenner und
einen Sack Rohlinge kaufen.
On Dec 1, 2005, at 9:05 AM, linux-audio-dev-
request(a)music.columbia.edu wrote:
>
> it seems that gmail has an unfortunate default setting that some
> people
> are not aware of... gmail users, please fix your settings!
To be specific, in Settings, choose the "use UTF-8" option for
"Outgoing message encoding" on the General page.
---
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
lazzaro [at] cs [dot] berkeley [dot] edu
---