I got about half way through an ecasound set up for controlling many of
hermes' 52 parameters with midi controllers and noticed the "input gain"
control. I was under the impression hermes could run as a stand alone
synth. But, I guess the name it bears, "hermes filter", might suggest
otherwise. I tried to catch Steve on irc today, but kept missing him.
Does anyone know if the input is optional for hermes?
If input is required, can I feed it an ecasound null input and still
get sounds out of it?
I'll use it however it's supposed to work. I just want to understand my
options.
Thanks,
Eric Rz.
Hi All,
I'm wondering if there is an app which can scan through audio files to
identify ones with similar characteristics to a reference file? A bit
like the apps which can identify duplicate images.
Cheers
Dylan
--
"I see your Schwartz is as big as mine"
                 -Dark Helmet
Hi.
I released ZynAddSubFX software synth todays
(standalone linux/windows and VST).
For those who don't know about it: zynaddsubfx is a
powerful opensource synth with many features (require
few pages just to make a list of them ) and the
homepage is at http://zynaddsubfx.sourceforge.net
News: - Solved some problems regarding VST
- added an advanced clipboard/preset module. On each
windows/object that support this, will appear 2
buttons "C" for copy and "P" for paste (from
clipboard/preset). You can make a collection of
waveforms, LFOs, ressonance functions, envelope,etc,
and share to others or use as favorites
- Completely removed the old format support (you can
use the prereleases
- bugs removed
- others
Also a very good news is that it comes with about 250
instruments (from usual organ/strings to the most
weird sounds ever heard )
Hope you like it.
I have intention to stop coding for a while and to
start to write the documentation and make some real
good sound examples.
Paul
__________________________________
Do you Yahoo!?
Yahoo! Mail - 50x more storage than other providers!
http://promotions.yahoo.com/new_mail
I'm trying to get a sequencer like rosegarden working
under knoppix 3.4 (a debian flavor linux live CD
distro). I am unable to get a midiport like
/dev/sequencer to work.
knoppix loads the OSS driver by default, and it
recognizes the soundcard and I'm able to hear audio
waves thru the soundcard, but no midi. I can
alternatively boot with ALSA but then I don't even
hear audio, so it seems that it may be easier with OSS
since I'm halfway there.
I'd like to get the external mpu401 midiport working,
and also access the onboard FM sounds. I am using an
ALS100 soundcard (and can resort to another ESS1869 if
I need to).
Here is my system info:
"lsmod" shows uart401, sound, and soundcore
cat /dev/sndstat displays nothing.
/etc/modules.conf doesnt seem to have anything
relevant (grep didnt find midi or sound anywhere)
What is the next step to enable my midiport? I can
promptly post back any requested info. Thanks!
__________________________________
Do you Yahoo!?
Take Yahoo! Mail with you! Get it on your mobile phone.
http://mobile.yahoo.com/maildemo
So after much pondering, I started writing this thing a few hours ago. As I
wrote the first paragraph, I figured I'd do it to provide something I could
never seem to find....Lot's of simple detail for Noobs! Damn!!! Now I know
why no one ever want's to write that basically; there's seventy kajillion
things you need to esplain!!! :)
Oh well....It's on the rails and headed out of the shed! I belive Mark
bellied up as a proof reader and some others...who were they now? I need to
put this in front of a few folks to make sure I'm not losing my mind!
R~
Hello,
It has turned out that the output of 44100 Hz files on my Intel 865 AC97
is only distorted on SPDIF. The analog output is all right!
Looks like it's not rate conversion at all, but SPDIF status. I saw
something like that in ALSA 1.0.7rc1 changelogs (but can't find these
changelogs again; I wonder why even ALSA tarballs don't include
changelogs).
I am now trying to compile the 2.6.8.1 kernel with alsa 1.0.7 rc1. It
may be strange idea, but I'm somewhat reluctant to to go to an -rc
kernel as yet. I'll report whether the problem is gone.
Yours, Mikhail Ramendik
Hi Erik,
Although I did say that sndfile-resample sounded ever so slightly better,
I also indicated that it was also ever so slightly less accurate.
The sndfile-resample sounded ever so slightly better than either the
original or the FFT-overlap resampled version (which sounded identical
to each other). I detected almost-imperceptible smoothing. It was
very difficult to hear any difference at all, so I think it did a very
good job on the voice. I have not attempted to evaluate the clicks and
pops wherein one might expect to hear more differences.
Best regards,
Dave.
Erik,
I do not low-pass filter in my resampler, yet it works fine. The reason
is that I assume that the input is band-limited, and this is usually true
for my own work. Not only is it band-limited, but usually also tapered
in the frequency domain, i.e. already effectively low-pass filtered. I use
a raised cosine window in the time domain and no window (other than the
rectungular truncation "window" for the case of downsampling) in the
frequency domain. Effectively I am assuming oversampling in the time
domain. I also have filtering capability, but separately from resampling,
for cases in which I do have higher frequencies. (However, this is also
very crude from a user point of view...)
On the presence or absence of higher frequencies: If there is further
processing down the road and the higher frequencies are missing, then the
results may be inaccurate (for example absent cross-products in the audible
region which are not the result of aliasing). Throwing out high frequencies,
or merely altering them somehow at every stage is not necessarily advisable,
so I caution people who extol the virtues of low-pass filtering. Now in
a library resampler, such as yours, I'm sure it's a good idea to enable it...
I wouldn't volunteer to answer your email otherwise!
Best regards,
Dave.
Hi All,
I'm setting up some cron jobs to record radio programmes while I'm on
holiday. I've got a basic command:
sox -t ossdsp -r 44100 -l -c 2 /dev/dsp test.wav
which records IIUC at CD quality (44.1K, 16bit, stereo) from my sound
card input. Levels are set independently by aumix.
Is this a good set of options? Is there anything else I should consider?
Disk space is no particular issue as I have 120Gig available and this
seems to use 10MB/minutes - meaning I have close on 200 hours if my
calculations are right...
Any suggestions welcome!
Cheers
Dylan
--
"I see your Schwartz is as big as mine"
                 -Dark Helmet