Hi
I released horgand-1.07
Version Changes
-------------------------
Sound Engine Changed, half CPU usage.
Recognizes MIDI Control Reverb depth message.
Recognizes MIDI Control Chorus depth message.
Fixed bug cleaning reverb.
Default Bank modified.
Changed FIFO schedule priority, now works on kernel 2.6.
Velocity Sensitivity modified.
Fixed bug in bass tune when jackd changes the default samplerate.
Improved Envelope, Reverb and Rotary effect.
Auto adjust internal sample rate to jackd sample rate.
Available in : http://horgand.berlios.de
Thanks
Josep
TAP-plugins 0.6.0 released.
TAP Reverb Editor initially released.
Homepage: http://tap-plugins.sf.net
OK, here we go:
* New plugin: TAP Fractal Doubler. Suitable for doubling tracks
containing vocals, acoustic/electric guitars, bass and just about
any other instrument. The effect is created by applying small
changes to the pitch and timing of the incoming signal. The changes
are created by a one-dimensional random fractal line producing pink
noise. Special thanks to Jan Depner for suggesting this plugin and
pointing me to useful information about fractals.
* New plugin: TAP Reflector. This plugin creates a psychedelic reverse
audio effect. It is especially worth trying this plugin on sustained
guitar and vocal tracks. Percussive sounds also create a very
characteristic atmosphere when played in a backward-ish style.
* New plugin: TAP Pink/Fractal Noise. This plugin came to life as a
secondary product of the development of TAP Fractal Doubler. It
generates pink noise by means of a one-dimensional random fractal
line generated by the midpoint displacement method.
Yes, it's Reverb time again!
* Vastly enhanced the internal workings of TAP Reverberator. As a
result, the sonic quality of reverberation got much better. (Well,
at least, now it *IS* real reverberation.) If you tried it once,
and found it sounded like hell, you definitely need to check it out
now!
* Long-long-standing denormal float (or whatever) problems causing
occasional runaway CPU-usage led to the complete re-implementation
of the internal DSP algorithm of the reverb using fixed-point
arithmetics. This inherently fixes denormal problems. However, the
option to use the previous floating-point DSP code remains as a
#define which you can set before compiling. The default is to use
fixed-point math. Very special thanks to Jan Depner for spending his
time with repeatedly testing the reverb and sharing his insights.
* A new application named TAP Reverb Editor has been written. It is a
standalone JACK app, with a GTK+-2 user interface. It works and
sounds the same as the LADSPA version (TAP Reverberator), but has
extended features that support direct editing of Reverb Types, with
immediate visual and sonic feedback. You can design new Reverb Types
easily, and 'backporting' these into the LADSPA version is also a
breeze. This program is available as a separate package called
TAP-reverbed.
* Introducing some new Reverb Types:
I made these for some acoustic guitar tracks:
* Ambience
* Ambience (Thick)
* Ambience (Thick) - HD
...and these for fun:
* Cathedral
* Cathedral - HD
Other changes:
* TAP Dynamics plugins (both Mono and Stereo) were also affected by a
runaway CPU-usage issue. It was fixed in the same way as with the
reverb: by converting the internals to use fixed point math by
default. (The #define option to use floating point math still
remains.)
* Applied patch from Luke Yelavich to clean up the Makefile a bit.
* Complete website/docs redesign. The documentation for TAP-plugins
and the user manual for TAP Reverb Editor is now available for
download as a separate package called TAP-plugins-doc.
Hope you enjoy this release. Please report any problems.
Tom
There are two ways of playing CDs.
The first one is to use the CD drive as a CD player. This way, the drive
will read and play the audio data thus converting the music/whatever to an
analog or digital signal (depending on your drive). When using this, you
have to have your cd drive connected to your soundcard/motherboard/whatever,
or use a headphone jack in the front of the drive. Also, you have to have
your soundcard mixer settings correct, so that it will mix the CD input
correctly.
The other, and much better way, is to use the CD drive to read the audio
data off the CD and use a soundcard to create the signal. You get better
quality because as you might guess, your 1010LT has better DACs than your CD
drive. This also works without connecting your CD drive and soundcard.
In XMMS this is selected by looking into:
Preferences->Audio I/O Plugins->AudioCD Reader->Output->Output mode
Here you can select either "Read Digital CD Audio", or "Play Audio CD Directly".
Sampo
Quoting Joseph Dell'Orfano <fullgo(a)dellorfano.net>:
> Probably a stupid oversight on my part, but I am having trouble
> listening to audio CDs. I have the jack plugin enabled on xmms and with
> jack running, I get no audio from xmms when playing a cd from either one
> of my drives (/mnt/cdrom or /mnt/cdrom1). xmms plays .wav files with no
> problem through the jack plugin. Any ideas?
>
> In case you were wondering, all else (including ardour, jack, etc) works
> fine on my machine. It is fedora core 1 with low latency kernel patches
> and a delta 1010LT card. DAW stuff works great as do my video apps. Just
> discovered this CD problem when trying to test a cd that I just burned.
>
> Thanks,
>
> -Joe Dell'Orfano
>
>
Probably a stupid oversight on my part, but I am having trouble
listening to audio CDs. I have the jack plugin enabled on xmms and with
jack running, I get no audio from xmms when playing a cd from either one
of my drives (/mnt/cdrom or /mnt/cdrom1). xmms plays .wav files with no
problem through the jack plugin. Any ideas?
In case you were wondering, all else (including ardour, jack, etc) works
fine on my machine. It is fedora core 1 with low latency kernel patches
and a delta 1010LT card. DAW stuff works great as do my video apps. Just
discovered this CD problem when trying to test a cd that I just burned.
Thanks,
-Joe Dell'Orfano
>
>
> ------------------------------
>
> Message: 3
> Date: Sat, 19 Jun 2004 22:33:31 -0400
> From: Alastair Couper <kalepa(a)shaka.com>
> Subject: [linux-audio-user] Re: Newbie question--
> what can I do with
> just a PC keyboard, my Linux box, and a SB 16 PCI
> card?
> To: linux-audio-user(a)music.columbia.edu
> Message-ID: <cb3i05$d6d$1(a)sea.gmane.org>
> Content-Type: text/plain; charset=us-ascii
>
> Try SpiralSynthModular. A softsynth with keyboard
> input included. Very
> stable app.
>
>
>
=== message truncated ===
Thanks for all the responses. I confess I was using
the earlier Rosegarden version, as Chris figured out.
I take it MIDI config. under rosegarden-4 is
different: must investigate further. Back to the
docs!
(I sincerely do try to RTFM, but I need FM's written
for FMorons. :-)
-MMW
=====
--
Seek professional help! Ask a librarian.
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Egads, I know I'm being a bit lazy and irresponsible tossing such a basic
question into this list, but I'm hoping to cut through most of the forest
of data to hone in to the right area.
I think I'm "done" with messing about with various motherboard audio
hardware - it's pretty noisy and has quite a few limitations I'm starting
to find annoying.
I have a (genuine) Aureal A3D Vortex2 PCI card in my stockpile of hardware,
but the support on both Wintendo and Linux seems iffy at best.
I have modest requirements:
1) Stereo in/out's fine, but multi-channel output is a pure bonus.
2) It should be as clean as practical - I realise outboard DAC/ADC is the
way to go if the godliness of Clean is sought, but I don't need better than
a -93dB noise floor.
3) It should have reprogrammable bitclocks so that resampling is not
necessary at any bitrate of sampling OR playback. Creative Labs seems to
have gone the way of fixed-rates and overrelies on DSP hardware to "patch
up" support for various user configurable bitrates. 96KHz support would be
awesome, but 48K wouldn't be a problem. Likewise, with sample sizes, 16's
all I need, but 20 or 24 would be awesome.
4) Intelligent and flexible audio-routing [monitoring would be nice, but
isn't required] so that I don't have minefields of which channels are
shadowed to which channels when recording from LINE OUT. I ran into so
real minefields with some motherboard chipsets where it was quite difficult
to record LINE IN audio without having LINE OUT audio mixed in with it.
5) Rich and excellent ALSA support, of course.
6) It should be a well-performing card with good bus-mastering DMA support
so driving it doesn't take too many CPU cycles.
I've seen Turtle Beach hardware on sale for reasonable prices, and a
generally positive writeup on these in various ALSA groups. How are they
for audio quality and performance?
7) Under US$200 price tag. Preferably under US$100.
Am I asking for something that exists? Thanks for your patience.
=MB=
PS: My negative bias against Creative Labs is not cast in stone - if my
impressions or perceptions are mistaken, I'm happy to correct them.
PPS: The Hammerfall cards look *AWESOME*, however I sense they are way over
my price range.
--
A focus on Quality.
On Sat, Jun 19, 2004 at 04:30:26 -0400, Chris Pickett
wrote:
> At first glance that seems fairly honest. It does
also suggest that
you
> wouldn't appreciate the benefits with a normal audio
CD, so in that
> respect it seems pointless, but maybe I'm missing
something. As for
the
> Nyquist frequency I read some discussion that some
people can hear up
to
> 23 kHz, and that there may even be psychoacoustic
effects up to 30
kHz,
> but I didn't try to find any references on this.
I think that there has been quite a lot of work done
on the optimal theoretical sampling rate for audio and
as I remember it was somewhere around 60kHz (I may be
wrong). There are good reasons why it's not as simple
as saying that the limit of hearing is, say, 23kHz so
we need a max sample rate of 46 (as per Nyquist).
Unfortunately, the A/D/A process uses filters (e.g.
decimation filters) which can introduce distortion at
frequencies which may be audible. Thus, one benefit of
higher sampling rates is that these distortions are
pushed up to higher frequencies well outside the range
of our ears. Thus, while a system running at 44.1 may
introduce distortions at, say, 20kHZ, a system running
at 96kHz would produce analogous distortions at, say,
44kHz which is way above the audible range. Thus 44.1
is probably not as transparent as 88, 96, etc.
I'm away from my desk at the moment so I can't list
any references but I do know that Dan Lavry has
recently written an AES paper about this subject.
While it's mainly about the reasons why 192kHz is bad
(limits of engineering), it provides an excellent
background in sample rates, etc. I think the web site
is lavryengineering.com
Psychoacoustics is just a bit too frightening for me
I'm afraid. Although I would urge everybody to read
Yost's book and if they like this stuff try Moore's
psychoacoustics book. Prepare to be amazed though!
Greg
___________________________________________________________ALL-NEW Yahoo! Messenger - sooooo many all-new ways to express yourself http://uk.messenger.yahoo.com
Running jack will work if qjackctl was active on shutdown. Restoring the
session on login will start jack in a usable manner. Jack will shutdown and
restart with the session. If X crashes, jack will still be running and the
new session will have qjackctl hang up. Need to kill jackd and try again. The
instance of jackd is not usable in the new session.
I have had similar problems starting jackd from anacron. Starts fine but
applications cannot connect to it and will hang.
There's got to be a correct and better way. Any ideas?
> > Most people aren't aware of much above
> > 16k, however the ear/brain is surely capable of
> perceiving differences, so a
> > higher sample rate is going to sound smoother in
> the way that faster film
> > looks smoother, the ear will perceive curves
> rather than digital grainyness.
>
I don't think it's as simple as saying that smoothness
is related to higher sample rate. I think that the way
in which we perceive music is related to distortions
introduced by the signal processing path and the
amount of detail which we can resove.
> There's certainly some evidence in favour of that,
> but consider this
> counter-argument:
> A 'grainy' signal could be regarded as the sum of a
> perfect signal plus
> a small distortion signal. If you can demonstrate
> that the
> distortion signal is inaudible then arguably it also
> doesn't have an
> audible effect whan added to a sound that is
> audible. In fact the well
> known auditory phenomenon of masking shows the
> reverse: a
> sound that by istelf *is* audible can be rendered
> inaudible in the
> presence of a simultaneous louder sound.
>
But what about effects such as stochastic resonance?
Add a (miniscule) amount of distortion and we hear
more of the music. Perhaps this is why some people
like the "sound" of DSD.
Greg
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Thank you tim and Alastair for your replies.
The problem seems to lie with getting jack (jackit 0.94) to work on my
system.
Starting Jack with Qjackctl gives an error message: ...
JACK: unable to mlock() port buffers: Operation not permitted
cannot set thread to real-time priority (FIFO/20) (1: Operation not
permitted)
cannot use real-time scheduling (FIFO/10) (1: Operation not permitted)
12:24:50.512 Could not connect to JACK server as client.
Starting Jack from the command line < jackd -d alsa -d hw:0 > results in:
loading driver ..
creating alsa driver ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|rt|32bit
control device hw:0
configuring for 48000Hz, period = 1024 frames, buffer = 2 periods
Couldn't open hw:0 for 32bit samples trying 24bit instead
Couldn't open hw:0 for 24bit samples trying 16bit instead
Couldn't open hw:0 for 32bit samples trying 24bit instead
Couldn't open hw:0 for 24bit samples trying 16bit instead
after which the system monitor (KDE System Guard) lists jackd as active
and other audio functions which are not compatible with jack do not work
properly. However jack-reliant applications still fail to respond. For
example starting QSynth, whether or not jackd is running, gives an error
message: ... failed to create the audio driver (jack) ....
I will hunt about on the jack homepage for a bit and try to make sense
of the situation, however any guidance is always appreciated.
David
