Hello, list!
I'm a regular GNU & Linux user. I'm also a musician. I would like to
know:
How can I make a cool sampler for live performance using GNU & Linux?
I know you guys know. Help me out on this and I volunteer to publish a
HowTo somewhere.
Please, remember that the sampler should be sophisticated enough... not
just a "working sampler".
Any cool ideas and stuff are welcome!
--
Renich Bon Ciric <renich(a)woralelandia.com>
Woralelandia
On 9/25/07, Fons Adriaensen <fons(a)kokkinizita.net> wrote:
> On Mon, Sep 24, 2007 at 09:59:22PM -0700, Maitland Vaughan-Turner wrote:
>
> > Erik de Castro Lopo <mle+la(a)mega-nerd.com> sez:
>
> > > That means the only sensible place to do DSD processing efficiently
> > > is in silicon; either FPGAs or ASICs.
> >
> > ooo, I've been meaning to get one of those! Can you (or anyone) point
> > me toward a FOSS-friendly FPGA?
>
> Sony used to sell a line of HW modules for DSD processing some years
> ago. Not sure if they still exist - I'd be surprised.
Ugh, no I just want an FPGA to play with and do whatever: make a DSD
processor or maybe an evil atomic powered robot. Anyway, I was just
wondering if anyone can tell me of any FPGA's that are made by
Free/Open-Source friendly companies.
I basically just switched topics randomly, there, ahaha sorry about that.
But about the Sony thing, you are talking about the Sonoma? or
something else? Haha, I think the Sonoma is out of my price-range
(for now...). Besides I hate Sony, (although I"m thinking about
buying a PS3. hahaha Anybody wanna help me write a Wii emulator for
PS3?)
~Maitland
Erik de Castro Lopo <mle+la(a)mega-nerd.com> sez:
> Jussi Laako wrote:
>
> > In some cases, direct DSP processing of the DSD stream would be
> > feasible.
>
> Really?
>
> Current CPUs are designed to work on integer or float samples. They
> are not designed to work on single bits. Attempting to do simgle bit
> processing on regular CPUs will be slow.
yeah, well it's time we rethink the way CPUs work anyway. It looks
like to me that we are running into the ceiling as far as Moore's law
goes. How about a generic, digitally controlled analog ALU as a
coprocessor for non-critical operations? (like real-time audio
processing). That could be fast... (no throwing rocks, please ;)
>
> That means the only sensible place to do DSD processing efficiently
> is in silicon; either FPGAs or ASICs.
>
ooo, I've been meaning to get one of those! Can you (or anyone) point
me toward a FOSS-friendly FPGA?
~Maitland
On 9/24/07, Paul Davis <paul(a)linuxaudiosystems.com> wrote:
> On Mon, 2007-09-24 at 13:13 -0700, Maitland Vaughan-Turner wrote:
> > oh yeah, why is that? acoustic waves are continuous, analog
> > representations are continuous. The more samples we can get the more
> > closely digital representation can mimic the analog which is far more
> > like the pressure waves than a series of pulses could ever be.
>
> there are lots of reasons why its wrong. information theory is one angle
> to take: how much information is being delivered per unit time. biology
> is another angle to take: how the human ear actually decodes acoustic
> pressure waves. non-linearities in pressure transducers (speakers etc)
> are another angle. the moment you convert an acoustic pressure wave into
> an electrical signal, its properties start to change. leaving it in
> analog form doesn't change it a lot. converting it to digital of any
> type changes the properties quite a bit, but this makes no difference if
> a symmetrical operation is possible when converting back to an analog
> electrical.
>
> but basically: "more pulses with less information per pulse" isn't
> equivalent to "less pulses with more information per pulse" and it
> certainly isn't equivalent to "continuously varying analog signal".
I dig what your saying, but when was the last time you listened to
just a single sample? Granted a 24 bit sample contains a lot more
data than a 1 bit sample. This is totally obvious. But when you look
at a whole chunk of samples, only the first several samples of the 1
bit stream are a question mark. After several samples it falls into
line and the amplitude can be accurately represented.
Now, I understand that 1-bit 2.8 Mhz can not achieve the dynamic range
of 24 bit samples, but it can surely represent more detailed
waveforms. Besides, there is only one variable to maximize (instead
of two). What happens when we turn that mega into a giga? (I know, I
know, an even bigger processing nightmare... hahaha)
Oh, and btw, just because I like DSD doesn't mean I don't like PCM! I
totally dig your work, and I use Ardour all the time. (well, ok, it's
broken on my box right now, but when it's working I use it all the
time! :) I mean, c'mon dude, you're like a celebrity! Thanks for
even talking to me =)
~Maitland
> On Tue, 22 Apr 2003 19:54:09 -0500
> "Dustin Barlow" <duslow at hotmail.com> wrote:
>
> > I read an interesting article on Direct Stream Digital (DSD) / Pulse Density
> > Modulation (PDM) entitled "A Better Mousetrap" by Brian Smithers in the May
> > 2003 issue of Electronic Musician. Since, Brian did a good job explaining
> > PDM/DSD in quasi-layman terms, I'll just quote snippets from his article to
> > set the stage for my questions.
>
> <snip>
>
> > DSD/PDM appears to be a superiour technique for recording and playing audio
> > material.
>
> Having been around digital audio and digital signal processing for over 10
> years, I am still far from convinced.
>
> > Granted, this technology may never catch on because of all the
> > hardware and software changes that would be required to mirror what a
> > typical PCM based DAW currently does. But, if DSD/PDM does catch on, and
> > DAWs start being produced, how will this effect current audio DSP
> > techniques?
>
> I have not looked into the maths behind algorithm development in DSD/PDM,
> but I doubt it is anywhere near as easy as with PCM.
>
> > The article mentions a program called Pyramix (Windows) which features DSD
> > support. However, for Pyramix to do EQ, dynamics, reverb processing, and to
> > display waveforms and vu levels, it converts DSD to a "high quality" PCM
> > format.
>
> That should tell you something :-).
<snip>
So..? Most PCM converters utilize a 1-bit stream also. Why not
utilize all the tools available for the task at hand?
As for processing, you can look at a PCM representation of a waveform
to ease the processing load and then just apply the changes to the
orignal DSD stream without ever having to process in the 1-bit domain
directly (which is way more processor intensive since you have to look
at a huge chunk of the stream in order to extract the amplitude data
that is available in each multi-bit sample).
IMHO, though, the hippest alternative at present is to process a DSD
stream in the analog domain and re-record it to DSD. This results in
a very "analog" sound. These days you can get analog gear with a
respectable dynamic range for a song (Mackie Onyx anyone?). When you
can get a 130 dB S/N ratio in the analog domain you really don't lose
too much converting back and forth from 1-bit domain. It's freakin
sweet!
If you haven't tried recording 1-bit. Do yourself a favor and demo
one of the new Korg recorders. It really is really good, no kidding.
~Maitland
On 9/24/07, Paul Davis <paul(a)linuxaudiosystems.com> wrote:
<snip>
>
> i'm afraid that you first have to convince a group of people with far
> too much to do already that there is something worthwhile about this.
> i've been following DSD for 4-5 years, and i not only unconvinced, i am
> certain that its nothing but marketing BS.
heh, that's ok if I don't convince anyone to do anything. If you just
spend some time thinking about it, that's cool to me. Even if that
thinking is just trying to explain to me why you think it sucks.
...mmmm electric kool-aid, lol
~Maitland
>
> ------------------------------
>
> Message: 2
> Date: Mon, 24 Sep 2007 11:28:17 +1000
> From: Erik de Castro Lopo <mle+la(a)mega-nerd.com>
>
>
> First off, why are you reprising a 4 year old thread if you don't have
> anything new to add?
Uhh, what about experience? I've actually used a 1-bit recorder. You
can talk about theory and quote papers all day, but that doesn't mean
*anything* compared to actually making recordings all day.
Also, things have changed a lot in 4 years. Now 1-bit recording is
available to everyone. Unfortunately, however, there are no linux
alternatives for dealing with 1-bit files. I just thought a bit of
discussion on this list might help with that.
>
> Maitland Vaughan-Turner wrote:
>
> > So..? Most PCM converters utilize a 1-bit stream also. Why not
> > utilize all the tools available for the task at hand?
>
> Yes, but you need to analyze what you are doing or the next thing
> you know you are paying mega dollars for triple gold plated single
> direction, rare earth metal interconnect cables and other such
> snake oil.
>
> > As for processing, you can look at a PCM representation of a waveform
> > to ease the processing load and then just apply the changes to the
> > orignal DSD stream without ever having to process in the 1-bit domain
> > directly (which is way more processor intensive since you have to look
> > at a huge chunk of the stream in order to extract the amplitude data
> > that is available in each multi-bit sample).
>
> Problem : the conversion from DSD and PCM is lossy [0], hence doing
> DSD -> PCM -> DSD -> PCM is a bad idea.
that's true, it's not the route I would take, but doing some effects
in PCM and mixing those with the original stream (either analog or
pure DSD mixing) results in more clarity than working wholely in the
PCM domain.
>
> > IMHO, though, the hippest alternative at present is to process a DSD
> > stream in the analog domain and re-record it to DSD.
>
> Two problems:
>
> - A-to-D converters are limited to about 20 bits of SNR due
> to things like silicon junction noise and those noisy
> electron thingys.
heh, i didn't say it wasn't lossy I just said it was hip. Like, it
sounds cooler (to me) than using a DAW.
>
> - All commonly used audio A-to-D converters are 1-bit so every
> time you go to analogue and then into a digital effects box
> you are doing another DSD to PCM conversion.
Is this a bad thing? I never said PCM sucks or anything. Haven't you
ever used an effects send/return, anyway? You don't have to send the
entire signal through the box. The affected signal is mixed with the
original signal.
I'm also using DSD to record a guitar running through a 20bit PCM
hybrid effects and a 16 bit keyboard and occasionally a V-Drum set.
Oh the blasphemy! :)
>
> See reference [0] below.
>
> > This results in a very "analog" sound.
>
> So, you have a way of objectively measuring how "analog" something
> sounds? I'd be interested in hearing your methodology.
not objectively. I just use my ears. I record my band and the detail
and clarity of the recording in DSD format is striking (compared with
PCM). This kind of detail I can only hear when playing vinyl records.
That's why I say it sounds more "analog."
Intuitively, one could also say that more sample points yield a
waveform that is closer to a continuous, analog waveform. Thus it
sounds more analog.
>
> > These days you can get analog gear with a
> > respectable dynamic range for a song (Mackie Onyx anyone?). When you
> > can get a 130 dB S/N ratio in the analog domain you really don't lose
> > too much converting back and forth from 1-bit domain.
>
> Where is you analogue to digital converter which also has 130dB SNR?
well, right, I'm saying the analog gear (that I have) is higher
quality than the digital gear. Of course, there is going to be some
change in sound everytime you switch domains, but is it a loss or is
it a gain?? I think it mostly depends on the skills of the audio
engineer.
>
> Erik
>
> [0] Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality
> Applications,
> Stanley P. Lipshitz and John Vanderkooy
> http://sjeng.org/ftp/SACD.pdf
Thanks for the link. My whole point of digging up this old thread
though, was to say that I've tried it, and my ears tell me that the
papers are incorrect.
IMO, 1-bit recording is LSD not snake oil...
...have some of the kool-aid, it's the good stuff ;)
~Maitland
I am trying to get the ESI MIDImate (EGO SYstems) to work.
I am running 2.6.17-5mdv and here is the /proc/bus/usb/devices entry:
T: Bus=01 Lev=01 Prnt=01 Port=00 Cnt=01 Dev#= 3 Spd=1.5 MxCh= 0
D: Ver= 2.00 Cls=00(>ifc ) Sub=00 Prot=00 MxPS= 8 #Cfgs= 1
P: Vendor=0a92 ProdID=1001 Rev= 1.04
S: Manufacturer=ESI
S: Product=ESI MIDI Mate
C:* #Ifs= 2 Cfg#= 1 Atr=80 MxPwr= 20mA
I: If#= 0 Alt= 0 #EPs= 0 Cls=01(audio) Sub=01 Prot=00 Driver=(none)
I: If#= 1 Alt= 0 #EPs= 2 Cls=01(audio) Sub=03 Prot=00 Driver=(none)
E: Ad=81(I) Atr=02(Bulk) MxPS= 4 Ivl=0ms
E: Ad=01(O) Atr=02(Bulk) MxPS= 4 Ivl=0ms
The problem appears to be that the snd-usb-audio driver will not load.
My /etc/hotplug/blacklist includes usb-midi to avoid driver conflicts.
The dmesg output is
usb 1-1: new low speed USB device using uhci_hcd and address 4
usb 1-1: configuration #1 chosen from 1 choice
unknown device speed 1
snd-usb-audio: probe of 1-1:1.0 failed with error -5
unknown device speed 1
snd-usb-audio: probe of 1-1:1.1 failed with error -5
/proc/asound/version says ALSA v1.0.12 is installed.
How do I get the driver to load?
===
Mark Watkins
<email suppressed>
Snd-ls v0.9.8.2
===============
Snd-ls is a distribution of Bill Schottstaedt's sound editor SND.
Its target is people that don't know scheme very well, and don't want
to spend too much time configuring Snd. It can also serve
as a quick introduction to Snd and how it can be set up.
Snd-ls also serves as base code for the San-Dysth softsynth
(http://www.notam02.no/~kjetism/sandysth/) and the Snd-rt music
programming language (http://www.notam02.no/arkiv/doc/snd-rt)
Changes 0.9.7.12 -> 0.9.8.2
---------------------------
-The rt_readin_tag startup bug is finally fixed. Thanks to Josh Lawrence,
Luke Hammon, Martin Rumori and Renick Bell for helping me finding it.
-Improved the build system a bit.
-Guile >=1.8.0 is now required to build and run Snd-ls.
-Fixed bug that caused snd to fail starting if no previously used
soundfile was opened during startup.
-Updated Snd from 8.4/12.9.2006 to 9.3/30.7.2007. Many important fixes.
Download from http://www.notam02.no/arkiv/src/snd/