Hi,
I'm doing a study on audio mastering. Hopefully this
letter will generate some correspondance from which
I'll learn enough to augment a GFDL licensed document
that I've been working on.
The test job is an album I recorded several years ago.
The source I've decided to use is 16bit 44100 audio
CD. I'm restricting the applications I use to jack
clients.
I guess the process of copying audio and data from CD
doesn't introduce any opportunity to compromise the
sonic quality of the source. Am I right or wrong?
Having opened the songs in Rezound, I've discovered
that some of the songs have an inordinate amount of
clipping--represented by vertical red lines on the
time line. For the purposes of testing this is
excellent because they're an opportunity to solve a
common problem. The clips cause a series of questions
for which I absolutely do not have definitive answers.
Is what I am seeing clipping or is there a more
accurate term to describe what I'm seeing?
Perhaps someone could provide a technical explanation
of clipping or a link to a definition.
What tools do you use for eliminating clipping that
already exists in a source? I don't care at all about
preventing the problem.
For the moment I am using the Rezound Arbitrary Fir
Filter to identify the hz where the clips occur and am
performing a decibal cut on the problem range. The
interface for this filter enabled me to do some
detailed work. Reguardless of how detailed I get,
there's an audible consequence to eliminating the
clips.
I configured a preset with the minimal settings
required to eliminate the clips but the result is
audibly unacceptable. Another preset eliminates about
80% of the clips and audibly is marginally acceptable
if applied to the entire file.
What's interesting about this specific set of clips is
that they are mostly inaudible. The clipping occurs
around 10kHz -> 15kHz and are almost all within the
high hat.
This causes me to wonder:
*how Rezound is configured to conclude that there are
clips
*should measuring for these types of problems be user
configurable or does a technical specification define
when a clip occurs
*can engineers safely ignore inaudible clips and tell
their clients that there's room to fudge and not to
worry
Is a Fir filter a good tool for addressing the problem
of digital clips or is there something better?
Are there alternative Fir filter algorithms that
produce better results than the one being used in
Rezound? I haven't a clue what Rezound uses.
Anyway, there's a few of my questions which I realize
are probably enough to exhaust anyone's patience.
Reguardless, I really would appreciate your thoughts.
ron
__________________________________
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Hi! again ...
Hi!
New Release : horgand 1.01
News in v1.01 (21/06/2003)
-------------
-Fixed Bass frequencys, now are tuned :-)
-Master Transpose transposes the bass line too.
-Master MasterTune tunes the bass line too.
-Added more chords to recognition.
-Fixed small bugs in split, chords ...
REQUERIMENTS:
* FAST COMPUTER
* LINUX
* LIBSNDFILE
* ALSA
* JACK
* FLTK 1.1
Web Page :
http://personal.telefonica.terra.es/web/soudfontcombi/
Josep
I've been trying to get mplayer to work with my delta44. I have
a working plug pcm type:
pcm.dplug {
type plug
slave.pcm ice1712
}
and
$ mplayer -ao alsa9:dplug foo.avi
will play audio, but no more than a few frames of video. If I look
at the verbose output I see;
alsa-init: requested format: 48000 Hz, 2 channels, Signed 16-bit (Little-Endian)
alsa-init: compiled for ALSA-0.9.4
alsa-init: soundcard set to dplug
alsa-init: pcm opend in block-mode
alsa-init: chunksize set to 1024
alsa-init: current val=7, fragcount=7
alsa-init: unable to set periods: Invalid argument
alsa-init: got buffersize=26212
The number of periods the driver requests is 7 which the code
obtains by calling snd_pcm_hw_params_get_periods_max(). So
my question is, what should this value be for a plug, should it
be the same as the underlying slave. Is the ice1712 not reporting
the correct maximum? The maximum jackd will start with is 5
periods so there seems to be something wrong here.
--ant
Hi,
I have a MIDI card (Yamaha DB50XG) that I use as a performer for my old
Ensoniq controller. It works fine. However, it seems that the Ensoniq
keyboard sends the velocity information at only half what it should be
-- if I pound on the keyboard, the volume is increased from normal
playing, but still rather quiet. If I play a midi file through tse3play
or something similar, the volume is full.
To get around this problem, I have two sysex files; I cat
volume_high.syx > /dev/midi when I want to use the keyboard, and cat
volume_low.syx > /dev/midi when I want to play midi files. The sysex's
each set the midi master volume to a level which is comfortable to use
with either.
However, this feels like a kludge. In addition, sometimes annoying
things happen like after I've been playing the keyboard at its
comfortable volume, I visit a web page with a MIDI on it, and it plays
at 250% volume and blasts my ear out. Or I play a game like DOOM which
has hardware MIDI support and have to lunge for the volume on my mixer
to keep from disturbing the neighbours. :)
The keyboard is a Ensoniq SDP-1 from 1986 or so. I tried the volume
setting on the keyboard in the hopes that it would modify the volume of
the notes sent to the midi card, but it seems to have no effect. (Is it
broken possibly?)
I was thinking about hacking the mpu401 driver so that when midi data is
received externally, it rewrites the velocity somehow before it reaches
the midi device. Or if that isnt possible, when a file is played to
/dev/midi, after the file sets master volume, reset it to a lower value.
Thoughts? Suggestions? This has been annoying me for a while now. :)
Thanks,
--
Ryan Underwood, <nemesis at icequake.net>, icq=10317253
at the risk of degrading what I'm appreciating - I just wanted to send a
note thanking y'all for the great discussions and high signal-to-noise
ratio of this group. I've been lurking for three months, waiting till I
have the finances to buy equipment to try all this stuff - and sitting on
this list has been highly educational. the recent mastering conversation
is really diggable. thanks!
--
Rev. Dan Easley
http://burntpossum.com
Hi,
How do I set the ouput line level of my delta 66 to -10dbv? Under
windows this can be done using the delta control panel but I can't see
any option for this in envy24control nor alsamixer.
thanks
--
rob <mailingLists(a)pangolin.org.uk>
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I'm looking for inexpensive soundcard but with high quality... for
recording dan monitoring
I found two candidates:
- - Soundtrack DSP24 (seems hard to find it in local store)
- - SB Audigy 2
Can some1 share experience on these two soundcard in linux...
regards,
khad
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I'm having trouble getting my audio hardware to work under linux. I'm
not sure if this is the correct place to ask a qustion like this
though... I'm getting noise on the audio input with a faint shadow of
what I'm trying to record. I'm trying to get one sound card working
first and hope to add one or two more.
Michael Tench
For those list members who are going to the postponed LinuxUser &
Developer Expo next week in Birmingham (that's England, not Alabama)
is there interest in rescheduling the informal linux-audio-users
meeting I originally proposed?
I'm going to be there all three days, but I'd prefer to meet up on the
Thursday, say 12.30pm for lunch? Meet at the Rosegarden stand?
Hopefully it will have calmed down a bit by then and I'll be able to
get away from our own stand...
Cheers
Daniel
Hi All:
I'm wondering if Dave Phillip's book 'Linux Music and Sound' is still a good
buy. Given that it was published 3 years ago, would much of the material now be
out of date?
TIA
Mick
