With my previous kernels (2.6.29 and 3.2.29), I tried plugging the
Motif XF to my PC via USB, but it wasn't recognized. After my latest
system upgrade to Slackware 14.1 and kernel 3.10.17, I decided to try
again, to see if something had changed. To my surprise, the MIDI ports
were recognized.
$ aplaymidi -l
Port Client name Port name
14:0 Midi Through Midi Through Port-0
28:0 Virtual Raw MIDI 3-0 VirMIDI 3-0
29:0 Virtual Raw MIDI 3-1 VirMIDI 3-1
30:0 Virtual Raw MIDI 3-2 VirMIDI 3-2
31:0 Virtual Raw MIDI 3-3 VirMIDI 3-3
32:0 YAMAHA MOTIF XF8 YAMAHA MOTIF XF8 MIDI 1
32:1 YAMAHA MOTIF XF8 YAMAHA MOTIF XF8 MIDI 2
32:2 YAMAHA MOTIF XF8 YAMAHA MOTIF XF8 MIDI 3
32:3 YAMAHA MOTIF XF8 YAMAHA MOTIF XF8 MIDI 4
MIDI recording and playback work fine, but now I want to know if it
can be made to work as a soundcard. I know that the Motif XF can work
as a USB audio interface on Windows and Mac. Here, it even happens to
be listed as one in "/proc/asound/cards"
$ cat /proc/asound/cards
0 [PCH ]: HDA-Intel - HDA Intel PCH
HDA Intel PCH at 0xf7910000 irq 42
1 [NVidia ]: HDA-Intel - HDA NVidia
HDA NVidia at 0xf7080000 irq 17
2 [Loopback ]: Loopback - Loopback
Loopback 1
3 [VirMIDI ]: VirMIDI - VirMIDI
Virtual MIDI Card 1
4 [XF8 ]: USB-Audio - YAMAHA MOTIF XF8
Yamaha YAMAHA MOTIF XF8 at usb-0000:00:1a.0-1.2,
full speed
but, problem is, it doesn't show on "aplay -l" nor "arecord -l" (nor
any other audio app for that matter).
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: ALC662 rev3 Analog [ALC662 rev3 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 2: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM]
Subdevices: 8/8
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
Subdevice #2: subdevice #2
Subdevice #3: subdevice #3
Subdevice #4: subdevice #4
Subdevice #5: subdevice #5
Subdevice #6: subdevice #6
Subdevice #7: subdevice #7
card 2: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM]
Subdevices: 8/8
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
Subdevice #2: subdevice #2
Subdevice #3: subdevice #3
Subdevice #4: subdevice #4
Subdevice #5: subdevice #5
Subdevice #6: subdevice #6
Subdevice #7: subdevice #7
Trying to force using it as soundcard (based on info from
"/proc/asound/cards") doesn't work either. For example:
$ aplay -D plughw:XF8 file.wav
aplay: main:722: audio open error: No such file or directory
Same result with "plughw:4" instead of "plughw:XF8".
I read that problems like this could happen if the USB soundcard was
plugged into a USB 3.0 port. Indeed, I have USB 3 ports here, but I
tried plugging it into USB 2 ports and the result was the same.
More command outputs:
$ lsusb
Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub
Bus 002 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub
Bus 003 Device 010: ID 046d:c315 Logitech, Inc. Classic Keyboard 200
Bus 003 Device 029: ID 046d:c077 Logitech, Inc.
Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub
Bus 001 Device 010: ID 0499:105c Yamaha Corp.
# lsusb -v
[...]
Bus 001 Device 010: ID 0499:105c Yamaha Corp.
Device Descriptor:
bLength 18
bDescriptorType 1
bcdUSB 1.10
bDeviceClass 0 (Defined at Interface level)
bDeviceSubClass 0
bDeviceProtocol 0
bMaxPacketSize0 8
idVendor 0x0499 Yamaha Corp.
idProduct 0x105c
bcdDevice 1.00
iManufacturer 1 YAMAHA Corporation
iProduct 2 YAMAHA MOTIF XF8
iSerial 0
bNumConfigurations 1
Configuration Descriptor:
bLength 9
bDescriptorType 2
wTotalLength 99
bNumInterfaces 1
bConfigurationValue 1
iConfiguration 0
bmAttributes 0xc0
Self Powered
MaxPower 0mA
Interface Descriptor:
bLength 9
bDescriptorType 4
bInterfaceNumber 0
bAlternateSetting 0
bNumEndpoints 2
bInterfaceClass 255 Vendor Specific Class
bInterfaceSubClass 0
bInterfaceProtocol 255
iInterface 0
** UNRECOGNIZED: 07 24 01 00 01 51 00
** UNRECOGNIZED: 06 24 02 02 01 00
** UNRECOGNIZED: 06 24 02 02 02 00
** UNRECOGNIZED: 06 24 02 02 03 00
** UNRECOGNIZED: 06 24 02 02 04 00
** UNRECOGNIZED: 09 24 03 02 01 01 01 01 00
** UNRECOGNIZED: 09 24 03 02 02 01 01 01 00
** UNRECOGNIZED: 09 24 03 02 03 01 01 01 00
** UNRECOGNIZED: 09 24 03 02 04 01 01 01 00
Endpoint Descriptor:
bLength 7
bDescriptorType 5
bEndpointAddress 0x01 EP 1 OUT
bmAttributes 2
Transfer Type Bulk
Synch Type None
Usage Type Data
wMaxPacketSize 0x0040 1x 64 bytes
bInterval 1
Endpoint Descriptor:
bLength 7
bDescriptorType 5
bEndpointAddress 0x82 EP 2 IN
bmAttributes 2
Transfer Type Bulk
Synch Type None
Usage Type Data
wMaxPacketSize 0x0040 1x 64 bytes
bInterval 1
Device Status: 0x0001
Self Powered
Any ideas? Anything else I can try?
Googling about using the Motif XF on Linux, I found this:
http://www.motifator.com/index.php/forum/viewthread/451381/
I also found this thread, but it's about MOX and not Motif XF:
http://www.motifator.com/index.php/forum/viewthread/458166/
which lead me to this patch:
http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg28116.html
The discussions are a bit old, but it seems that the patch was
included upstream and MOX is working. Maybe the problem with the XF is
similar?
--
____________________
Blog: http://aiyumi.warpstar.net/
Hi,
Apparently Matt Tytel released initial version of Cursynth, a
ncurses-based polysynth.
http://www.gnu.org/software/cursynth/
Since there are a few visually impaired users around here, I'm
curious: does it work well for you?
Alexandre
hi all,
for a project, i'm working with 192 khz, but unfortunately, none of my
audio interfaces (rme multiface and fireface ucx) support 192khz. the
ucx supports 192khz in hardware, but the usb/alsa/class-compliant mode
only seems to go up to 96khz. so i wonder, is there any way to perform
upsampling within jack?
tia,
tim
Hello, I want to make my first release of a software based audio
production environment. I did a 30 day roadmap therefore. Every feedback
will be considered.
Please contact if your interested in helping to advance Advanced Gtk+
Sequencer development.
https://sourceforge.net/p/ags/wiki/release-0_4_0
Hint: there's currently no midi binding
thanks in advance
Joël
Hi all,
A new version of aubio, 0.4.1, is out.
aubio is a library of functions to perform audio feature extractions
such as:
- note onset detection
- pitch detection
- beat tracking
- MFCC computation
- spectral descriptors
This version is mostly focusing on media file input and output. Here is
a quick overview of the changes.
The most interesting feature in this release concerns aubiocut. Thanks
to the sponsoring of Mark Suppes, the python script to slice sound
steams was extended to be sample accurate, cut overlapping segments, and
work on multiple channels.
New source and sink objects have been added to let aubio read and write
WAV files, even when built with no external libraries. This should
simplify the use of aubio on platforms such as Android or Windows.
Existing sources and sinks have been extended to read and write from and
to multiple channels. This makes python-aubio one of the fastest and
most versatile Python module to read and write media files.
This release also comes with a stack of bug fixes and code clean-ups.
Note: this version is API and ABI compatible with 0.4.0. Since it only
adds new features to the existing interface, your existing source and
binary code will keep working without any modifications.
To find out more about aubio and this release:
Project homepage:
http://aubio.org/
Post announcing aubio 0.4.1:
http://aubio.org/news/20140312-1953_aubio_0.4.1
ChangeLog for aubio 0.4.1:
http://aubio.org/pub/aubio-0.4.1.changelog
Source tarball, signature and digests:
http://aubio.org/pub/aubio-0.4.1.tar.bz2http://aubio.org/pub/aubio-0.4.1.tar.bz2.aschttp://aubio.org/pub/aubio-0.4.1.tar.bz2.md5http://aubio.org/pub/aubio-0.4.1.tar.bz2.sha1
API Documentation:
http://aubio.org/doc/latest/
Happy hacking!
Paul
On Sun, 2014-03-16 at 19:10 +0100, tim wrote:
> don't tell anyone that they should reduce their production
> quality to your standards.
48 KHz isn't my standard, for good reasons it's a common studio standard
(as long as live usage latency isn't an issue). If you want analog
sound, than use analog gear. If you want to have perfect digital quality
48 KHz / 24 bit is all you need, for production 32 bit float _might_ be
better.
On Sun, 2014-03-16 at 18:21 +0100, tim wrote:
> >> fwiw, for digital synthesis (non-standard or distortion synthesis) i
> >> ended up rendering my compositions at 3mhz ... which was a good
> >> compromise between computation time and sound quality.
> >
> > And I prefer analog gear over digital gear
>
> maybe because you did not like the aliasing artifacts and/or
> quantization noise?
I'm a dino and used to analog gear. When I need to use digital gear I
stay with 48 KHz, if I have a choice, I use analog gear. I don't know
why I prefer analog gear.
I'm pleased to say this is now available from:
http://sourceforge.net/projects/yoshimi/files/1.2/yoshimi-1.2.0.tar.bz2/dow…
Apart from a number of bug fixes we have:
Circle and Spike AddSynth waveshapes
MIDI bank and program change - with extra configuration in 'settings'
Included patch set additions and updates.
Next on the roadmap is midilearn and extra control exposure.
Discussion would be welcome on:
yoshimi-user(a)lists.sourceforge.net
--
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
