tisdagen den 10 juni 2003 13.21 skrev Frank van de Pol:
> On Tue, Jun 10, 2003 at 08:30:39AM +0200, Robert Jonsson wrote:
> > Hi,
> >
> > > In fact the bounce feature in MusE is "realtime". It means that you
> > > have to wait the real duration of the track to be rendered.
> > > In a non "realtime" mode the track is rendered as fast as computer can.
> >
> > AFAICT the realtimeness of the bounce feature is like that because of
> > design constraints. Okay, bouncing wavetracks should be possible in
> > non-realtime, but not when using softsynths.
> >
> > This is because all softsynths use alsa-sequencer as the input interface.
> > And if I'm not missing anything, this interface is strictly realtime
> > based. (perhaps it can be tweaked by timestamping every note and sending
> > them in batches? it seems very hard though.)
>
> You are right, with the current alsa-sequencer the softsynths are driven by
> realtime events. Though an application can enqueue the events to the
> priority queues with delivery timestamp, the scheduling is handled
> internally by the alsa sequencer. This causes some problems (especially for
> sample accurate synchronisation with JACK or LADSPA synth plugins (XAP?)),
> but also for network transparency and support for MIDI interfaces which
> accepts timing hints (Steinberg LTB or Emagic AMT ... if specs of the
> protocol were available :-( ).
>
> During the LAD meeting at Karlsruhe we discussed this and sketched a
> alsa-sequencer roadmap that focusses on transition of the alsa-sequencer
> from kernel to userspace and better integration with softsynths / JACK.
> A few things from this are very much related to your track bouncing /
> off-line rendering thing:
>
> - Provide facility to delegate scheduling to the client. The implementation
> would be to deliver the events directly (without queuing) with the
> timestamp attached to the registered client port. This would allow the
> client to get the events before the deadline (time at which the event
> should be played) and use that additional time to put the events at the
> right sample position.
>
> Note that for the softsynth to get advantage of this the application
> should enqueue the events (a bit) ahead of time and pass the timestamp.
> Some of the current applications (including MusE) use the alsa-sequencer
> only as event router and drive it real-time.
>
> Since the softsynth/plugin has no notion of the acutal time (only the
> media time and sample position), rendering at arbitrary speed should be
> possible: bounce faster than realtime or even slower than realtime for
> those complex patches.
>
> - JACK is real-time, and bound to the sample rate of the soundcard. Since
> the audio sample rate can also be used as a clock master for the alsa
> sequencer this would be a good option to ensure synchronisation. The
> transport of JACK and alsa sequencer can be tied together (either one of
> the two acting as master, a run-time configurable option) to provide
> uniform transport and media time amongst the applications that hook into
> the JACK and/or alsa sequencer framework.
>
> For the offline rendering no nice scheme has been worked out yet; I guess
> it would be something along the lines where the application that owns the
> sequencer queue has full control on the transport, moving media time at the
> speed the frames are actually rendered, and the app(s) generating the
> events keeping at least one sample frame ahead of time.
>
> Frank.
Okay, I didn't know that this had been up on the table, how far has this work
progressed, was it just the Karlsruhe meeting or has more thinking occured?
(fyi I'm CC:ing LAD, it might be a more appropriate place for this
discussion..).
Regards,
Robert
Hi!
new release ...horgand-092
Program is released GNU/GPL version 2.
Horgand is a organ, Jack capable and recognizes MIDI Program Change (1-32).
Horgand is tested on a PIII 966 (Debian Sid ) and PII 300 (Gentoo).
Changes in 0.92
----------------------
-Added Reverb at this moment only presets available.
-All the sliders and dials widget reponse in realtime.
-Solved bug in keyboard level scaling, that reduced noises,
unfortunatelly sound changes a little bit.
-Small look changes and other minor bugs solved.
REQUERIMENTS:
* FAST COMPUTER
* LINUX
* ALSA
* JACK
* FLTK 1.1
Web Page :
http://personal.telefonica.terra.es/web/soudfontcombi/
Josep
I am Computer Support technician at the Marine Research Lab, near
Leigh, North Island, New Zealand. We are a remote department of the
University of Auckland.
We are building a computer based on a JENLOGIX miniature industrial
motherboard into an underwater housing in order to record on the HDD
the output from 4 hydrophones.
We have a MIDIMAN DELTA 44 four-channel sound card and hope to be
able to write a simple script to record and save the sound streams to
disk automatically whenever the computer wakes up. An external timer
would wake up the computer for 10mins every hour.
I am a newbie to Linux but have just ordered Redhat9 and intend to
get the system working on a desktop (DELL OptiplexGXa, 233MHz PII)
before installing it on the U/W computer.
I have found the snd-ice1712 driver module on the ALSA site.
Does anyone know if there are scripts already written that would do
what we want or could be readily modified?
Any other advice welcomed.
- - - - - - - - - - - - - - - - - - - - - - - - - - - - -
Jo Evans, Network Manager, Computer Support ph ext.3601
University of Auckland ph 64 - 9 - 422 6111
Leigh Marine Laboratory fax 64 - 9 - 422 6113
P.O. Box 349, Warkworth j.evans(a)auckland.ac.nz
NEW ZEALAND - - - - - - - - - - - - - - - - - - - - - -
Last week there was a thread on LAD about DSP kits and integration
with Linux, and I mentioned that I knew of a few small DSP projects
with promise, but couldn't find the URL.
Found the URL today, and it is:
http://www.gweep.net/~shifty/death/
And:
http://www.gweep.net/~shifty/ezkit/
Seems like this might give some of you some inspiration in the "put a
cheap DSP into Linux, integrate it ..." department.
--
;
Jay Vaughan
r&d>>music:technology:synthesizers - www.access-music.de/
Hi!
I have ported Sebastien Metrot's libakai to Linux some couple of weeks ago.
Until it will be in CVS one day you can get it from:
http://stud.fh-heilbronn.de/~cschoene/projects/libakai/
I added some code to the demo application to extract samples from a Akai disc
yesterday, so you can now actually not only see but also hear what's on a
disc.
You will notice that there's still a small bug, I will fix it ASAP, but I
won't complain if someone else will look for it ;)
Best regards.
Christian
Hi! i'm doing some studying on dsp, and one thing I could never properly
understand is the term of "excitation signal" I seem to find it associated
to environment or natural sources, but I cant really find a definition.
Could someone with enough knowledge on this subject please
give me a brief on this? Thanks in advance!
Juan Linietsky
JACK 0.72.4
JACK is a low-latency audio server, written primarily for the GNU/Linux
operating system. It can connect a number of different applications to
an audio device, as well as allowing them to share audio between
themselves. Its clients can run in their own processes (ie. as normal
applications), or can they can run within the JACK server (ie. as a
"plugin").
JACK is different from other audio server efforts in that it has been
designed from the ground up to be suitable for professional audio work.
This means that it focuses on two key areas: synchronous execution of
all clients, and low latency operation.
**CHANGES**
* Updated documentation
* Bug fixes
* MacOSX port. Includes a ProjectBuilder file to help compilation.
Requires PortAudio to be installed.
* Ringbuffer example files added
* New example client: simple transport master to demonstrate Jack's
transport API. Requires GNU readline to compile.
* Removed software monitoring and improved hardware monitoring
semantics.
Taybin Rutkin
> I think someone else (Paul?) hit it on the head. I fyou load 2 8bit
> samples and pass them to a soundcard that is expecting 16-bit samples,
> you'll get a LOT of garbage - like white noise - and it will be 1/2 as long
> as your input sample.
>
> See, each pair of 8 bits will become a 16 bit sample.
>
> input 8 8bit samples:
> dec: 0 64 127 64 0 -64 -127 -64
> hex: 0x00 0x40 0x7f 0x40 0x00 0xc0 0x81 0xc0
>
> read as 4 16bit samples (ignore endianness):
> hex: 0x0040 0x7f40 0x00c0 0x81c0
> dec: 64 32576 192 -32320
>
>
> Notice how the waveforms don't resemble each other at all.
>
>
> Clearer, now?
I See.. So if I wanted to convert to a 16bit, how would you recommend I do
this? It would seam I would need some type of Filler.... err. white noise or
just blank noise to fill in the extra 8bits. I guess I could convert every
bit read from the file from an 8bit (unsigned char) to a 16bit (signed short)
then write it to the /dev/dsp?
Bare with me here, I'm not a verteran.. So I would have to convert the
unsigned char to a signed char, then to a signed short... So Not knowing how
the conversion is done in the OS, I'm assuming that the resulting signed
short would be padded with 'off' bits. which would come out as silence
correct? ( but it's in the same sample, so you really wouldn't hear the
silence )
Werd?
--
It looked like something resembling white marble, which was
probably what it was: something resembling white marble.
-- Douglas Adams, "The Hitchhikers Guide to the Galaxy"
This is a first for audio applications.
Native Instruments Traktor just appeared on Freshmeat! :)
I guess some people are starting to take notice of the OSS community. Though
it isn't OSS in itself and not available for Linux (OS X, + a few marginal
os:es).
But the most interesting thing was that they used Freshmeat as a marketing
channel.
/Robert
Hi,
I'm trying to write a patch editor and librarian for Roland's CM-32L
synth module. This was a repackaged version of the popular MT-32 with 8
part multimbral support, separate drums and a basic reverb facility. The
only controls on the front panel are a power switch and master volume
knob - everything else has to be done via MIDI.
I have written a simple program to change which preset each part uses,
and can also trigger notes. Now I want to get a SysEx dump so I can put
together the patch editor. However, whenever I send a Request Data
message the unit doesn't appear to respond. It definitely receives the
message as the MIDI activity light comes on. But when I try to read from
the MIDI device file my call blocks forever.
Has anyone had succes in getting SysEx data flowing back and forth
between Roland sound modules and their computer? I have attached my
simple test program in case I'm doing something obviously wrong. By the
way, I'm working on NetBSD which has slightly different MIDI device
filenames to Linux.
Chris
--
http://www.btinternet.com/~chris.wareham/
/* get_sysex_dump.c
* cc -Wall -Wformat -g -o get_sysex_dump get_sysex_dump.c
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <errno.h>
#include <fcntl.h>
#include <unistd.h>
#define SYSEX_START 0xf0
#define SYSEX_END 0xf7
struct sysex {
unsigned int length;
unsigned char *data;
};
static struct sysex *sysex_new(void);
static void sysex_delete(struct sysex *);
static void sysex_set_length(struct sysex *, unsigned int);
static void sysex_read(int, struct sysex *, int);
static void sysex_write(int, struct sysex *);
static int sysex_checksum(struct sysex *, int, int);
int
main(int argc, char *argv[])
{
int fd, i;
struct sysex *se;
fd = open("/dev/rmidi2", O_RDWR|O_SYNC);
if (fd == -1) {
fprintf(stderr, "open() failed: %s\n", strerror(errno));
return 1;
}
se = sysex_new();
sysex_set_length(se, 11);
se->data[0] = 0x41; /* Roland's ID */
se->data[1] = 0x10; /* Device Number - default appears to be 0x10 */
se->data[2] = 0x16; /* Model ID */
se->data[3] = 0x11; /* Command ID - "Request Data" */
se->data[4] = 0x04; /* Address MSB */
se->data[5] = 0x01; /* Address */
se->data[6] = 0x76; /* Address LSB */
se->data[7] = 0x00; /* Size MSB */
se->data[8] = 0x01; /* Size */
se->data[9] = 0x76; /* Size LSB */
se->data[10] = 0x0e; /* Checksum */
sysex_write(fd, se);
sysex_set_length(se, 252); /* 8 bytes to wrap data, and 246 bytes of data */
sysex_read(fd, se, 1);
printf("Read %d bytes\n", se->length);
for (i = 0; i < se->length; i++)
printf("0x%0x\n", se->data[i]);
sysex_delete(se);
if (close(fd) == -1) {
fprintf(stderr, "close() failed: %s\n", strerror(errno));
return 1;
}
return 0;
}
/* Receives a MIDI System Exclusive message */
void
sysex_read(int fd, struct sysex *se, int autostop)
{
int n, length = 0;
unsigned char c;
if (fd == -1) {
fprintf(stderr, "sysex_read() failed: invalid fd\n");
return;
}
if (se == NULL) {
fprintf(stderr, "sysex_read() failed: null sysex argument\n");
return;
}
while ((n = read(fd, &c, 1)) == 1 && c != SYSEX_START)
usleep(40);
if (n != 1)
return;
while (length < se->length) {
n = read(fd, &c, 1);
if (n == 1) {
se->data[length++] = c;
if (autostop && se->data[length - 1] == SYSEX_END)
break;
}
}
if (length < se->length) {
se->length = length;
se->data = realloc(se->data, se->length);
}
}
/* Sends a MIDI System Exclusive message */
void
sysex_write(int fd, struct sysex *se)
{
unsigned char *data;
if (fd == -1) {
fprintf(stderr, "sysex_write() failed: invalid fd\n");
return;
}
if (se == NULL) {
fprintf(stderr, "sysex_write() failed: null sysex argument\n");
return;
}
data = calloc(se->length + 2, sizeof(unsigned char));
data[0] = SYSEX_START;
memcpy(data + 1, se->data, se->length);
data[se->length + 1] = SYSEX_END;
write(fd, data, se->length + 2);
free(data);
}
static struct sysex *sysex_new(void)
{
struct sysex *se;
se = calloc(1, sizeof(struct sysex));
return se;
}
static void sysex_delete(struct sysex *se)
{
if (se->length != 0)
free(se->data);
free(se);
}
/* Sets the length of a MIDI System Exclusive message */
void
sysex_set_length(struct sysex *se, unsigned int length)
{
if (se == NULL) {
fprintf(stderr, "sysex_set_length() failed: null sysex argument\n");
return;
}
se->length = length;
se->data = realloc(se->data, se->length);
}
/* Determine a Roland System Exclusive checksum */
int
sysex_checksum(struct sysex *se, int start, int length)
{
int checksum, i;
if (se == NULL) {
fprintf(stderr, "sysex_checksum() failed: null sysex argument\n");
return 0;
}
checksum = 0;
for (i = start; i < start + length; i++) {
checksum += se->data[i];
if (checksum > 127)
checksum -= 128;
}
checksum = 128 - checksum;
return checksum;
}
