| Message written at 28 Apr 1997 22:30:31 +0100
--- Copy of mail to tre@indy1.calarts.edu ---
In-reply-to: <199704241405.HAA00683@indy1.calarts.edu> (message from Tom Erbe
on Thu, 24 Apr 97 07:06:09 -0700)
>>>>> "Tom" == Tom Erbe writes:
>> I'm using the the Mac CornBucket score generator via Linux Executor: it
>> works okay, but the formatting is bad. Here's what I get:
>>
>> f1 0 8192 10 1^Mi1 0.000000 0.000100 10200.301758 50.250000 0.500000
>> 0.546584^Mi1 0.001000 0.000096 9985.474609 50.348255 0.499969
>> 0.543260^M
>> [etc. for a gazillion lines all run together like that]
>>
Tom> This is a problem with Mac/Unix line feed incompatibility. You can fix it
Tom> with BBEdit on the Mac side (open the file, "Save As...", select
Tom> "Options...",
Tom> select "Line Breaks->Unix",save). You could also probably fix this with
Tom> sed
Tom> on the Unix side. It has been a while since I have had to use sed to do
Tom> this,
Tom> can anyone suggest the proper invocation to convert Mac line-feeds to
Tom> Unix
Tom> line-feeds?
tr is the command and the arguments are
tr "\015" "\012" < old > new
==John
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In-reply-to: <9704281139.ZM7518@ulysses.stanford.edu> (message from Tobias
Kunze on Mon, 28 Apr 1997 11:39:11 -0700)
Subject: Re: lcl name list error
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The "Bath" sources comment that out as I had the table extensible some
time ago.
Do not just comment it out -- take a recent version
==John
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From: Matthew Biddulph
To: csound@maths.ex.ac.uk
Subject: MIDIfying the DX7 ORCs
Date: Tue, 29 Apr 1997 12:17:37 GMT
Reply-To: matthew.biddulph@balliol.ox.ac.uk
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I've downloaded the DX7 files for Csound from
http://www.parnasse.com/dx72csnd.shtml , including the bug-fixed ORC
files. I'd like to convert the ORCs so that they can be controlled via
MIDI messages instead of from the SCO file - can anyone give this
newbie an outline of how to do so? Following the simple template for
MIDIfying given in MIDIFY1.{ORC,SCO} doesn't seem to work, perhaps
because the DX7 ORC files are set up to use octave.note pitch values?
Please help if you have a spare moment.
Thanks,
Matt.
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In-Reply-To: <9704281139.ZM7518@ulysses.stanford.edu> (message from Tobias Kunze on Mon, 28 Apr 1997 11:39:11 -0700)
Subject: Re: lcl name list error
Date: Tue, 29 Apr 97 13:50:25 BST
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The "Bath" sources comment that out as I had the table extensible some
time ago.
Do not just comment it out -- take a recent version
==John
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In-Reply-To: <3364FF38.ABD@protozoa.com> (message from John Boyd on Mon, 28 Apr 1997 12:49:12 -0700)
Subject: Re: New versions (linux)
Date: Tue, 29 Apr 97 15:34:52 BST
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Yes, the version I am preparing includes diskin (which is teh name I
was given for soundin2).
ELF binaries will depend on one of the LINUX people making it --
usually Dave Phillips or someone (my memory seems to have paged out).
It also has Michael Clarke's FOG opcode, and that still needs a
little tweak according to him. And I want to include Richard Dobson's
fixes.
==John ff
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Date: Tue, 29 Apr 1997 11:27:56 +0200
From: Gabriel Maldonado
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In version 1.7 uploaded in 27 Apr 1997 19:34:55 +0200, there was a bug
concerning analysis utilities called by -U flag: cvanal and pvanal are
called erroneously as "lpanal".
Now I fixed the bug and the correct executable is avalaible at the same
url:
http://www.agora.stm.it/G.Maldonado/download.htm
I'm sorry for those persons which had downloaded the program before than
today.
Bye and ... happy csounding :)
--
Gabriel Maldonado
mailto:g.maldonado@agora.stm.it
http://www.agora.stm.it/G.Maldonado/home2.htm
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From: Richard Dobson
Message-Id: <199704291637.RAA26784@talisker.pact.srf.ac.uk>
Subject: Re: a hardware question....
To: csound@maths.ex.ac.uk
Date: Tue, 29 Apr 1997 17:37:28 +0100 (BST)
In-Reply-To: <199704251800.NAA04322@ho13.eng.ua.edu> from "jwilder" at Apr 25, 97 01:00:30 pm
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>
> I wanted to post this question to the group to see if there are any soundcard pros out there..
>
> For my Masters of Science in Engineering thesis at the University of Alabama, I've been working on building a quad-output
> soundcard for the PC that will work with Csound. Basically, the idea is that it will take any digital sound file that
> contains 16 bit samples and convert that to analog information ==> sound. The thesis idea came from a music prof. here
> that's been wanting to have electronic music quad output for a long time and suggested that if I wanted to do a thesis
> project for the music dept., that I build one for him. I know there are some devices out there that will do quad, but I
> haven't been able to find anything that's specifically used just for quad output. Anyway, enough background info. and
> on to my question.
>
> I'm using a hardware interrupt to signal an interrupt software routine to port out 4 samples to my card, where each of the
> samples is loaded into a different digital-to-analog converter. My hardware interrupt is a clock that equals 44.1kHz, so
> in theory, within one clock cycle of 44.1kHz (about 22 micro-secs.), the interrupt software routine (ISR) should access
> the digital sound file(which is dang large as we all know) four times within 22usec. The problem is that hard drive access
> time isn't quick enough to keep up with my 44.1kHz clock; hence, within 22usec I could never hope to make 4 reads to the
> hard drive, and the samples to the DACs aren't making it there fast enough! I thought about loading my samples first into
> RAM, but that wouldn't be very efficient and would place a severe limit on the size of the sound file(100 secs of quad
> music takes around 35 Megabytes). Does anyone know how a sound card achieves its output? Surely it uses some kind of
> buffering (maybe with DMA?) that will keep on filling up a buffer with the samples, and then the samples are just
> ported out from the buffer, thus allowing the critical 22usec time frame to be achieved so that all DACs are filled and
> outputed according to the 44.1kHz clock before the next set of samples comes in.
>
> Any knowlege on sound card output would be appreciated...
>
> Thanks in advance,
> Joel Wilder
>
>
There are several general issues here:
Firstly, are you designing for DOS, for Windows 3.1, or Windows95 (or even NT)?
For DOS you can access a hardware device such as a soundcard directly, and
use either or both of interrrupts (you could use more than one), and DMA.
However, old DOS dma is pretty slow, and brobably cannot handle four-channel
stuff too well. These days, you reaaly want to use the full facilities of
Windows 3.1 at least, which means writing a device driver (HORRIBLY difficult!)
which can then also be used under Windows95.
Secondly, yes absolutely use buffering. There are two levels of buffering when
playing a sound from disk. The first in handled by the disk driver/operating system,
which deals in blocks of data - you can ask the system to get four samples from
the disk, but in fact it HAS to get at least one sectors-worth - say 512 bytes.
Often the minimum amount is much more ( the 'cluster size' - perhaps 8K. Windows
will add a further layer of buffering to this. Then you write a block (say 32K)
to the card, which then streams it out at the required rate. The trick is to do
so-called 'double-buffering', where you send one block of data while the card is
still busy playing out the previous one. This is because, though disk access for
bytes is horribly inefficient, access for large blocks is wonderfully efficient.
It is quite possible, with a good card design, to transfer data simply by
polling, and not using interrupts or dma at all. I have a Motorola DSP56K
development board, which has three blocks of 3k words of fast memory. One of
these blocks is used to buffer data being sent to the card, and the clock to
write the data to the output port comes (in my case) from a SONY PCM. The main
loop on the card receives samples from the PC and stores them in a circular buffer,
and the DSP interrupt arrives each 44100Hz cycle to write from that buffer.
On the PC end, I have a simple loop which asks if the card is ready to receive
data (this is the 'poling' part), by reading a bit in a control register.
(the card appears to the PC as eight bytes of memory starting at Port address
$340). When the card say 'yes' I send the card a block of data, 32KB each.
Beacause my first machinw was so slow, I coded the PC end in assembler, but
on a faster modern machine coding in C will be fine.
So long as the disk is fast enough, this works fine.My first machine was a 386
with a very slow drive, and I could not get data of it fast enough. Once I had
upgraded to a 486 and a more modern drive, 44100Hz stereo played back easily.
This enable you to do the job without handling any interrupts or dma, or even
writing a device driver. This is viable up to and incluing Windows95 (if you
have a compiler which allows you to use 'outp()' calls - Visual C++ 32bit
doesn't!); for Windows NT you HAVE to use a device driver, as it does not
permit any direct access to the hardware, for security reasons.
The advantage of a device driver is that it separates the hardware from the
computer to a degree. If you improve the card, you only have to supply a new
driver, not a whole new application. Also, you can publish the driver functions
to third parties (the 'public interface'), without having to expose all your
clever low-level code if you choose to keep it to yourself.
I hope this helps. good luck with the design.
Richard Dobson.
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From: Richard Dobson
Message-Id: <199704291728.SAA27745@talisker.pact.srf.ac.uk>
Subject: Re: Reinit soundin
To: csound@maths.ex.ac.uk
Date: Tue, 29 Apr 1997 18:28:45 +0100 (BST)
In-Reply-To: from "Matt J. Ingalls" at Apr 15, 97 03:37:52 pm
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>
>
> > Can anyone help me understand WHY soundin ug seems not to reinit?
> > Well, soundin goes on quietly reading its file without reinitialize.
>
> here's the first line in the init code (which i assume is called by
> reinit):
>
>
> if (p->fdch.fd != 0) return; /* if file already open, rtn
> */
>
>
> bypassing all the file opening stuff (AND not reiniting the skip time,
> which is why you are doing all this, right?)
>
>
> -matt
I think there may be a cunning bug here, which I've found elsewhere -
the Csound code is full of tests against fopen() - it assumed
that a return value of zero signalled failure, whereas in fact it is -1.
That is, an fd of 0 in fact signals a valid file. the test should be modified
to recognize zero as a valid fd, therefore. If this proves to be the case,
I will send the fix to John Fitch, to go in the next version (if he hasn't
already done it!).
Richard Dobson
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Date: Tue, 29 Apr 1997 13:31:30 -0500 (CDT)
From: Kent Williams
To: Richard Dobson
Cc: csound@maths.ex.ac.uk
Subject: Re: a hardware question....
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There is a card from Frontier designs (Wavecenter) that has two stereo
outputs, that can handle 4 signals at once.
Without putting intelligence on the card itself, you're pretty much limited
to using DMA to send data out the port. If you're doing hardware engineering,
you should be up on how DMA works. If it was me, I'd set up the
card to use one DMA channel, have the driver software interleave the
samples, and use a counter to select which DAC to load.
DMA programming is a little weird but not bad once you get used to it.
You can (I believe) set up the DMA to transfer synchronized to your
digital world clock. To transfer 4 16 bit samples every 1/44100 of a second
the clock will just run at 176.4 kilohertz.
On the PC side you simply fill a buffer from disk, initiate the transfer,
then fill a second buffer during the transfer of the first transfer. When
you get an I/O Complete interrupt, you set up a transfer for the second
buffer, and fill the first.
How big you make each buffer depends on a latency/CPU usage tradeoff.
If your buffers are too small you will have very low latency, but
will have to spend more time handling interrupts and setting up DMA
requests. If your buffers are too large, then you will spend less time
servicing interrupts, but the time quantum of each transfer is larger,
introducing a time lag between when some other real time event (like a
midi note) can affect the output.
These are great days though -- modern processors are fast enough to
do lots of work besides pumping out audio. Back in the old day, you had
to do things in assembly language and cross your fingers in order to
keep up with the demands of high bandwidth devices.
It also gets a lot easier if you don't mind putting a DSP or
a microcontroller on the card, and implementing a shared memory window.
Then you can just blast data off the hard disk directly into the dual
port buffer. This is what the Turtle Beach Hurricane architecture does,
and they were getting a reliable 4 tracks of 44khz sound off of hard disk
back in the dark days of 25mhz 386's.
---------------------------------------------------------------------
Kent Williams kent@inav.net -- http://soli.inav.net/~kent
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From: Tobias Kunze
Message-Id: <9704291310.ZM9835@ulysses.stanford.edu>
Date: Tue, 29 Apr 1997 13:10:24 -0700
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Hi-
Here is another one I've run into before, but can't work around this time.
I get the message "GOTOS list is full..extending" and then the allocation
of instr 2 (which has the many if statements) fails and csound returns.
Any idea what's going wrong here? I guess it's a mrealloc failure somewhere
but don't really have the time to track the bug down.
-Tobias
| sorting score ...
| ... done
| orch compiler:
| 803 lines read
| instr 1
| instr 2
| instr 3
| instr 4
| GOTOS list is full..extending
| MIT Csound: 3.44 (Mar 2 1997)
| orch now loaded
| displays suppressed
| audio buffered in 64 sample-frame blocks
| reading 256-byte blks of shorts from devaudio
|
| PMLqueuesize = 8192
| writing 256-byte blks of shorts to devaudio (AIFF)
| SECTION 1:
| ftable 1:
| ftable 2:
| ftable 3:
| new alloc for instr 1:
| new alloc for instr 2:
--
______________________________________________________________________
Tobias Kunze t@kunze.stanford.edu
CCRMA, Stanford University http://www.stanford.edu/~tkunze
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Date: Tue, 29 Apr 1997 22:41:08 -0400 (EDT)
From: Paul Koonce
To: t@ulysses.stanford.edu
Cc: Csound mailing list
Subject: Re: GOTOS list is full..extending
In-Reply-To: <9704291310.ZM9835@ulysses.stanford.edu>
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On Tue, 29 Apr 1997, Tobias Kunze wrote:
>
> Hi-
>
> Here is another one I've run into before, but can't work around this time.
> I get the message "GOTOS list is full..extending" and then the allocation
> of instr 2 (which has the many if statements) fails and csound returns.
>
> Any idea what's going wrong here? I guess it's a mrealloc failure somewhere
> but don't really have the time to track the bug down.
>
>
> -Tobias
>
>
I hit the same bug on both GOTOS and LABELS. Changing the macro defines and
recompiling will work around it, but the mrealloc is still a problem.
-Paul Koonce
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Subject: M68K MAC
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Message written at 29 Apr 1997 21:31:23 +0100
Am I correct in my belief that teh MILLS system only runs on PowerMAC,
or am I just misinformed? I ask as if I am right, users of old MACs
have been deserted! I am willing to rebuild the 68K version again,
from my latest sources, but only if there is a reason.
==John ff
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Date: Wed, 30 Apr 97 12:03:03 BST
Subject: Re: Reinit soundin
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Message written at 29 Apr 1997 21:23:15 +0100
--- Copy of mail to rwd@pact.srf.ac.uk ---
In-reply-to: <199704291728.SAA27745@talisker.pact.srf.ac.uk> (message from
Richard Dobson on Tue, 29 Apr 1997 18:28:45 +0100 (BST))
Much as I dislike disagreeing with my friend Richard, ANSI C says that
fopen returns NULL if the open fails for any reason. fopen returns a
variable of type FILE*, and any C compiler must arrange that the
result is only NULL (usually zero) when there is an error.
I think you are confusing fopen with open(2), which returns -1 for an
error, but in not an ANSI function.
On UNIX of course a file descriptor of zero is standard input, and so
would not be returned by open(2).
However Richard is correct that a file descriptor in UNIX can be zero,
and as this function only exists elsewhere as a sop to unix hackers,
it is a bug.
==John
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