I have come into possession, and still have most of this stuff:
1)
http://asia.yamaha.com/en/products/proaudio/interfaces/mini_ygdai_cards/my4…
2 x MY4-DA digital > analog cards
1 x MY4-AD analog > digital card
2 x MY8-AD analog > digital card
2) Modular Synthesis Plug-in System --
http://asia.yamaha.com/en/search/?query=PLG150-AN
2 x PLG150-AN Analog physical modeling plug-in board
2 x PLG150-DX Advanced DX/TX plug-in board
1 x PLG150-VL Virtual acoustic plug-in board
1 x PLG150-PF Piano plug-in board
1 x PLG100-XG XG plug-in board
3) SY-85 Performance Data Disk for SY-85 synthesizers (3.5" floppies) -
obsolete, no longer on Yamaha website
1 x VD-8501 "Top 40" -- 128 performances & 256 voices
1 x SD-8501 "Rock Band" sound data disk -- 17 waveforms, 128 voices, 64
performances
2 x SD-8502 "Sax & Brass" sound data disk -- 6 waveforms, 128 voices, 64
performances
4) gone
5) SY-22 Synthesizer Voice & Memory Data Cards (about the size of old PCMCIA
cards)
3 x VC2252W -- "Vector 2 Studio Selection" -- 64 voices
3 x MCD32 -- 32KB RAM cards
6) Music Cartridges (beyond obsolete)
1 x EMS MC107 / OM 23945 -- "Jazz" Music Cartridge Styles
1 x EMS MC105 / OM 23943 -- "Classic & Folk" Music Cartridge Styles
I need to get them moved, either to a home that wants them or to the dump. If
anyone is into this stuff and would like any of it (just pay shipping and
similar costs, or come to the Bahamas and pick them up yourself!) let me
know.
This stuff is basically unused and in original boxes.
all the best,
drew
Hello again Q!
OK, then your learning is progressing very nicely. :-) In the bas drum I
think I mainly heard EQ'ing nicetiesand in the snare, it was mainly the
compression, that struck me favourably.
You might call 11/8 and 108 trivial, which it might be for the rhythmically
advanced. But there were rather complentary elements in the drums and rhythmic
instruments, that keep me busy. :-) At least that way it guarantees to be a
source of new discoveries for a little longer, before I get to the stage of
being familiar with it and returning to an old friend. So it's the good old
school of prog really. :-)
(a)Rhythmically swinging yours
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
Hello everyone!
I've still got problems joining my twosoundcards (Delta 1010LT and EMU
1212m). I've created the multi device for capture and playback, taken care of
the bit width conversion and still it doesn't work. See the .asoundrc below.
With Delta and EMU's analog I/O (device 0) JACK will start, but gives a lot
of ALSA_PCM xruns. With the S/PDIF and/or ADAT/DSP part of the EMU card
included JACK won't start or will shutdown immediately. Also ecasound/arecord
won't work.
Having tried the EMU S/PDIF on its own with arecord, I found, that it would
only allow 8kHz samplerate, instead of the requested 48kHz. The EMU card is
however setup to take the clocksource from the delta card, which works
perfectly with 48kHz. OK, even it takes its clock from the outside (Lindy A/D
converter).
To test the S/PDIF in on the EMU I tried playing back something to the
Delta's S/PDIF out, but only got silence.
So it looks, like there is more than one problem. But I can't spot it. I've
looked at the .asoundrc documentation again and the list of ALSA plugins. OK,
there is the route plugin for creating interleaved audio, using the ttable,
but for a start, I've read, that this will cause great latency. Anyway, I've
built my JACK version myself and I don't see, why it shouldn't include complex
MMAP. But that is a little further along the way.
So here's my .asoundrc:
# The 8 ADAT ins of EMU 1212m, conversion to S16_LE
pcm.emu_adat {
type linear
slave {
pcm "hw:4,2"
format S32_LE
}
}
# The 2 Ins, 2 Outs of EMU 1212m, they have S16_LE
pcm.emu_analog {
type hw
card 4
device 0
}
# The 2 S/PDIF Ins and 2 S/PDIF Outs of EMU 1212m,they have S16_LE
pcm.emu_spdif {
type hw
card 4
device 1
}
# The 16 DSP out channels of EMU 1212m, they have S16_LE
pcm.emu_dsp {
type hw
card 4
device 3
}
# The M-Audio Delta 1010 LT, conversion from S32_LE to S16_LE
pcm.delta {
type linear
slave {
pcm "hw:1"
format S32_LE
}
}
# Multi capture device including EMU ADAT, Analog and S/PDIF channels and
# Delta analog and S/PDIF in channels
pcm.multi_capture {
type multi
slaves.a {
pcm "delta"
channels 12
}
slaves.b {
pcm "emu_analog"
channels 2
}
#slaves.c {
# pcm "emu_spdif"
# channels 2
#}
#slaves.d {
# pcm "emu_adat"
# channels 8
#}
bindings.0.slave a
bindings.0.channel 0
bindings.1.slave a
bindings.1.channel 1
bindings.2.slave a
bindings.2.channel 2
bindings.3.slave a
bindings.3.channel 3
bindings.4.slave a
bindings.4.channel 4
bindings.5.slave a
bindings.5.channel 5
bindings.6.slave a
bindings.6.channel 6
bindings.7.slave a
bindings.7.channel 7
bindings.8.slave a
bindings.8.channel 8
bindings.9.slave a
bindings.9.channel 9
bindings.10.slave b
bindings.10.channel 0
bindings.11.slave b
bindings.11.channel 1
#bindings.12.slave c
#bindings.12.channel 0
#bindings.13.slave c
#bindings.13.channel 1
#bindings.14.slave d
#bindings.14.channel 0
#bindings.15.slave d
#bindings.15.channel 1
#bindings.16.slave d
#bindings.16.channel 2
#bindings.17.slave d
#bindings.17.channel 3
#bindings.18.slave d
#bindings.18.channel 4
#bindings.19.slave d
#bindings.19.channel 5
#bindings.20.slave d
#bindings.20.channel 6
#bindings.21.slave d
#bindings.21.channel 7
}
# Control device for multi_capture, I chose card 1 (Dela), because it gives
# the clock signal.
ctl.multi_capture {
type hw
card 1
}
# Multi playback for EMU 1212m Analog, S/PDIF and DSP out channels as well as
# Delta 1010 LT's analog and S/PDIF out channels
pcm.multi_playback {
type multi
slaves.a {
pcm "delta"
channels 10
}
slaves.b {
pcm "emu_analog"
channels 2
}
slaves.c {
pcm "emu_spdif"
channels 2
}
#slaves.d {
# pcm "emu_dsp"
# channels 16
#}
bindings.0.slave a
bindings.0.channel 0
bindings.1.slave a
bindings.1.channel 1
bindings.2.slave a
bindings.2.channel 2
bindings.3.slave a
bindings.3.channel 3
bindings.4.slave a
bindings.4.channel 4
bindings.5.slave a
bindings.5.channel 5
bindings.6.slave a
bindings.6.channel 6
bindings.7.slave a
bindings.7.channel 7
bindings.8.slave a
bindings.8.channel 8
bindings.9.slave a
bindings.9.channel 9
bindings.10.slave b
bindings.10.channel 0
bindings.11.slave b
bindings.11.channel 1
bindings.12.slave c
bindings.12.channel 0
bindings.13.slave c
bindings.13.channel 1
#bindings.14.slave d
#bindings.14.channel 0
#bindings.15.slave d
#bindings.15.channel 1
#bindings.16.slave d
#bindings.16.channel 2
#bindings.17.slave d
#bindings.17.channel 3
#bindings.18.slave d
#bindings.18.channel 4
#bindings.19.slave d
#bindings.19.channel 5
#bindings.20.slave d
#bindings.20.channel 6
#bindings.21.slave d
#bindings.21.channel 7
#bindings.22.slave d
#bindings.22.channel 8
#bindings.23.slave d
#bindings.23.channel 9
#bindings.24.slave d
#bindings.24.channel 10
#bindings.25.slave d
#bindings.25.channel 11
#bindings.26.slave d
#bindings.26.channel 12
#bindings.27.slave d
#bindings.27.channel 13
#bindings.28.slave d
#bindings.28.channel 14
#bindings.29.slave d
#bindings.29.channel 15
}
# Control device for multi_playback, I chose card 1 (Delta) again as it is
# the clock source
ctl.multi_playback {
type hw
card 1
}
Any idea anyone? Could it be a specific configuration issue of the EMU 1212m
or is there a special kernel option - not directly related to sound -, which
might help?
Warm regards and thanks for any hints
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
Hello all!
I've recently compiled a new kernel and since then my "alarm clock", doesn't
work anymore. My alarm clock is mplayer started by cron. I know, I once had
that issue before, but beat me, if I know, what I've changed since then.
My mplayer uses JACK by default and the system it's running on, still runs
JACK and mplayer from a root account normally. I'd much rather not reconfigure
that.
So any idea, whichkernel option might have stopped my mplayer from being
triggered by cron?
Thanks for any advise.
Kindly yours
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
When sounds are competing in the mix, the frequently given advice is
usually to carve out frequency ranges for them with EQ.
But what do you do when its a thick analog keyboard part that spans a
4-octave range, and the range it's playing is the whole musical point of
it? What do you do for those parts that won't fit in an EQ shoebox?
--
+ Brent A. Busby + "We've all heard that a million monkeys
+ Sr. UNIX Systems Admin + banging on a million typewriters will
+ University of Chicago + eventually reproduce the entire works of
+ James Franck Institute + Shakespeare. Now, thanks to the Internet,
+ Materials Research Ctr + we know this is not true." -Robert Wilensky
List,
I've created a Wiki page with my findings of using real-time,
low-latency audio on the RPi: http://wiki.linuxaudio.org/wiki/raspberrypi
It's not complete and the RPi needs some more hacking to squeeze out the
last bits of processing power that can be dedicated to things like
samplers, synths, effects and guitar amp emulators.
Thought I'd share the link, saw some questions regarding the RPi pass by
the last few months. To show off what the RPi is capable of:
http://www.youtube.com/watch?v=u6XKaRe_MEA&webm=1 (if you still use
Flash remove '&webm=1')
Regards,
Jeremy
Linux audio seems to be the new popular target for worldwide hackers.
This time linuxmao.org <http://www.linuxmao.org> is their last victim, I
hope the admin team will put back the old cherished website online quickly!
olivier
On Tue, April 16, 2013 9:36 am, Julien Claassen wrote:
> Hello everyone!
> So I've got the e-mu 1212 soundcard working. it should have two
> analogue ins
> and two analogue outs and then S/PDIF I/O and ADAT - i.e. 8 channels in
> and 8
> channels out. That doesn't seem to be the case.
12 and 12 at 48000 or 44100s/s.
> With a normal JACK call to hw:4 I get 2 ins, 2 outs (I suppose the
> analogue
> part.
> With JACK using the syntax given in
> Kernel/Documentation/sound/alsa/emu10k1-jack.txt
> using hw:4,2 and hw:4,3 I get 8 ins and 16 outs, which is 3/4 of what
> was to
> be expected (16 ins and 16 outs).
I know for the delta (1712 based cards) the maximum i/os always shows up
12/10.
I am not so sure with this unit. even with the delta1010/66/44 if you are
using the gui controller (I know you have not seen that) It shows only the
number of adcs and dacs there are (I have a D66). In the monitor mixer
section it shows all the outputs, but only real inputs. You may have some
of the same.
There may be some reason the adat shows only outputs until it is
connected... then it may be 8, 4 or 2 channels depending on rate. (only 8
inputs up to 48k) Or it may be 2 inputs if it is being used as a second
spdif (total of 6 i/o). The real question in my mind is what inputs are
really there in the 8 channels you have now? The pcie1010 reconfigures
itself quite dramatically depending on what is plugged in. Plugging the
dock in makes some of the plugs on the pcie card inactive for example. The
dock and the daughter card can not be used together either.
I think you will find the 16 outputs are there just because they are mixer
inputs. The inputs? I would assume either spdif l/r then ch 1/2 NC 3/4/5/6
or ch1/2 NC 3/4/5/6 and spdif l/r.
I think the adat would have to be plugged in before the system start and
already set to the rate you were going to use as I don't think (correct me
if I am wrong) alsa can dynamically change the number of ports a device
has and jack would find that a problem too. Do you have the bits to test
what is what?
--
Len Ovens
www.OvenWerks.net
