Cakewalk UA-1G

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ATTENTION: This page is still under construction. This part will be removed I finished all the tests described on this page. Main problem is I don't get dmix / softvol to work.

Product page with pictures

This is a card from probably March 2009. The format of this page is based on the UA-4FX page, because that page is laid out so well.

The Roland Cakewalk UA-1G is a good quality audio device, with the following features:

I hope that this little howto can guide you and save you some time. The logic of the Roland Cakewalk UA-1G is that it is operated by hands, directly on the device. There are some hardware limitations, but it is frankly a very nice tool.


The Roland Cakewalk UA-1G is not yet supported by Alsa. With the patch below the device is recognised by the snd_usb_audio module. You can play and record at 44.1kHz and 16-bit samples if you have set the Advanced Driver switch to off. Full support for 24-bit playback and recording at rates up to 96 kHz has been successfully tested.


Unfortunately, Alsa has little control over this device.

In fact, it is not a problem of operating systems (Linux / FreeBSD / Window\$), but rather a hardware issue. For example, Input level, Phone volume, mixing, even sample rate are controlled at the hardware level. There is no software control. Not even with the original Mac/Window\$ drivers. Some users may like this old-fashion way of managing a sound card. Others may dislike it. Analyse your needs before purchasing this sound device.

The Roland Cakewalk UA-1G is designed to be a simple mixing board. That is why the input and output levels are controlled by knobs and not software.

Fortunately, Alsa offers a variety of features and plugins, which emulate at software level the missing features. This is the advantage of Alsa over other sound systems.

Contents

Understanding the Roland Cakewalk UA-1G logic

At first, you should have a look at the back pane. Or look at the photo of the device. Try to locate three important buttons:

Advanced Driver toggle switch

On the left side of the UA-1G, there is an on/off switch called "Advanced Driver", which controls the USB mode of the device. You have to remove your sound card prior changing this switch.

Sample rates switch

On the left side of the UA-1G, there are two switch buttons called "SAMPLE RATE". The switches have four values: 32 kHz, 44.1 kHz, 48 kHz, 96 kHz and one switch called "96 kHz MODE" for RECORD or PLAY at 96 kHz:

Cold/Hot reboot

Whenever you switch from Advanced ON to Advanced OFF, change sample rates, or change from 96 kHz RECORD to 96 kHz PLAY, you will either need to:

Lost?

To understand advance mode, sample rates and REC/PLAY, it is recommended to use aplay and arecord in verbose (option -v) mode. This is what we will do during the howto. This will clearly show you how the audio devices plays and records sound.

Basic Alsa configuration

alsaconf is not required to use the Roland Cakewalk UA-1G. UDev is able to recognise the audio device. Just plug and play.

Assigning audio system rights

GNU/Linux is a secure system. To play sound, you need audio system rights. To query your systems rights:

$groups
my_username adm disk dialout fax cdrom floppy tape dip video plugdev powerdev scanner

In this example, I don't have enough rights to play/record sound. To assign rights either use your distributions configuration tools (like drakconf, yast). Find your own user account and add the group "audio" to it. Alternatively you can do it manually like this:

sudo adduser my_username audio

Will add the user my_username (replace with your username) to the audio group. You should be able to play music.

Naming the Roland Cakewalk UA-1G device

If the UA-1G is the only device of your computer, you can address the device using the plughw:0,0, but it is not very convenient. We recommend using the alphanumeric name of the device. To query the name of your device, type:

$cat /proc/asound/cards
 0 [UA1G           ]: USB-Audio - Cakewalk UA-1G
                      Roland Cakewalk UA-1G at usb-0000:00:1d.1-1, full speed

Here, you should use plughw:UA1G rather than plughw:0,0

Testing sound output

Test the card output. This command plays a woman voice on 2 channels ("Front Right", "Front Left"):

speaker-test -c2 -D plughw:UA1G -twav

This program needs to be installed separately on some distributions.

Playing sound

You can play any sound and it will be played with the sample rate set via the SAMPLE RATE switches.

To play a sound:

aplay -D plughw:UA1G foo.wav

For a better understanding, it is recommended to play in verbose mode:

aplay -v -D plughw:UA1G foo.wav

Several lines of text will explain what Alsa is doing:

Playing WAVE 'foo.wav' : Signed 24 bit Little Endian, Rate 48000 Hz, Stereo
Plug PCM: Linear conversion PCM (S24_3LE)
Its setup is:
  stream       : PLAYBACK
  access       : RW_INTERLEAVED
  format       : S24_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 24000
  period_size  : 6000
  period_time  : 125000
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 6000
  period_event : 0
  start_threshold  : 24000
  stop_threshold   : 24000
  silence_threshold: 0
  silence_size : 0
  boundary     : 1572864000
Slave: Hardware PCM card 1 'Cakewalk UA-1G' device 0 subdevice 0
Its setup is:
  stream       : PLAYBACK
  access       : MMAP_INTERLEAVED
  format       : S24_3LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 24
  buffer_size  : 24000
  period_size  : 6000
  period_time  : 125000
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 6000
  period_event : 0
  start_threshold  : 24000
  stop_threshold   : 24000
  silence_threshold: 0
  silence_size : 0
  boundary     : 1572864000

The sound file has a 48.000 Hz sample rate and 24 bits of resolution.

When playing two sounds at the same time, an error message is displayed:

aplay: main:546: audio open error: Device or resource busy

You can play sounds at higher sample rates and 24-bit precision using Advanced mode.

Recording sound

With the UA-1G device set to Advanced OFF, you can use the arecord utility from the Alsa package to record any sound from the microphone:

$arecord -f cd -t wav -D plughw:UA1G foobar.wav

For a better understanding, try the same command in verbose mode:

arecord -v -f cd -t wav -D plughw:UA1G foo.wav

The resulting message:

Recording WAVE 'foo.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
Plug PCM: Rate conversion PCM (48000, sformat=S24_3LE)
Its setup is:
  stream       : CAPTURE
  access       : RW_INTERLEAVED
  format       : S16_LE
  subformat    : STD
  channels     : 2
  rate         : 44100
  exact rate   : 44100 (44100/1)
  msbits       : 16
  buffer_size  : 22050
  period_size  : 5512
  period_time  : 125000
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 5512
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 22050
  silence_threshold: 0
  silence_size : 0
  boundary     : 1445068800
Slave: Hardware PCM card 1 'Cakewalk UA-1G' device 0 subdevice 0
Its setup is:
  stream       : CAPTURE
  access       : MMAP_INTERLEAVED
  format       : S24_3LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 24
  buffer_size  : 24001
  period_size  : 6000
  period_time  : 125000
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 6000
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 24001
  silence_threshold: 0
  silence_size : 0
  boundary     : 1572929536

The Roland Cakewalk UA-1G is able to record in 24 bits, at the sample rate of 96 Khz.

With Advanced mode enabled, playing a recorded sound at 96 Khz with the "96 kHz MODE" switch set to PLAY should work:

aplay -v -D plughw:UA1G foobar.wav

At 96.000 Hz, the audio device can play OR record, but not both.

If aplay complains of "No such file or directory," it might be because the "96 kHz MODE" switch is set to RECORD. Try changing it to PLAY. You might then need to cold or hot reboot.

Now you should be able to play the sound:

aplay -v -D plughw:UA1G foobar.wav

Changing the switch (and enabling Advanced mode) is all that is needed to see something like this:

Playing WAVE 'foobar.wav' : Signed 16 bit Little Endian, Rate 96000 Hz, Stereo
Plug PCM: Linear conversion PCM (S24_3LE)
Its setup is:
 stream       : PLAYBACK
 access       : RW_INTERLEAVED
 format       : S16_LE
 subformat    : STD
 channels     : 2
 rate         : 96000
 exact rate   : 96000 (96000/1)
 msbits       : 16
 buffer_size  : 48000
 period_size  : 12000
 period_time  : 125000
 tick_time    : 1000
 tstamp_mode  : NONE
 period_step  : 1
 sleep_min    : 0
 avail_min    : 12000
 xfer_align   : 12000
 start_threshold  : 48000
 stop_threshold   : 48000
 silence_threshold: 0
 silence_size : 0
 boundary     : 6755399441055744000
Slave: Hardware PCM card 0 'UA-1G' device 0 subdevice 0
Its setup is:
 stream       : PLAYBACK
 access       : MMAP_INTERLEAVED
 format       : S24_3LE
 subformat    : STD
 channels     : 2
 rate         : 96000
 exact rate   : 96000 (96000/1)
 msbits       : 24
 buffer_size  : 48000
 period_size  : 12000
 period_time  : 125000
 tick_time    : 1000
 tstamp_mode  : NONE
 period_step  : 1
 sleep_min    : 0
 avail_min    : 12000
 xfer_align   : 12000
 start_threshold  : 48000
 stop_threshold   : 48000
 silence_threshold: 0
 silence_size : 0
 boundary     : 6755399441055744000

Remember, at 96.000 Khz, you can either play OR record, but not both.

Also remember some programs don't offer plughw or might give you an error message concerning the "sample format". Be sure to exactly configure in this case the record parameter as put out above, especially setting the format to S24_3LE (signed 24 bit little endian), 2 channels and the configured sample rate will be required for recording.

Advanced Alsa configuration

The recommended settings in this HOWTO are now:

This will record/play sound in 24 bits, at the frequency set via SAMPLE RATE switches.

Full-duplex mode

The Roland Cakewalk UA-1G is a full-duplex device up to 48.000 Hz. With Advanced mode enabled, you can also set the UA-1G to 96 kHz PLAY or 96 kHz RECORD. The Edirol_UA-25 page suggests using the asym plugin to get full-duplex 96 kHz operation, but the hardware does not support it. Someone, please verify that the UA-25 asym setup is reasonable and update this section.

Custom softvol PCM

$alsamixer -c 0 to use mixer settings on card 0 will return:

no mixer elems found

Unfortunately these is no software control over the hardware mixers on the device (like in any USB device), nothing will show up in mixer programs.

Fortunately, Alsa offers the softvol plugin to create a software volume control. We will also define this control as the default mixer.

The below example does not work for me (in my system I have two soundcards, so I changed card to "1". Alsamixer always gave me the error "no mixer elems found" no matter what I tried. Also pcm "pasymed" is not defined anywhere in this example (dmix/plugin:dmixer does not work for me either). It seems the author of UA-4FX has not tested this or left out important parts of it. If you have a working example with internal and external soundcard please drop me a note.

Here is my first try:

pcm.!default {
   type             plug
   slave.pcm       "softvol"
}
pcm.softvol {
   type            softvol
   slave {
       pcm         "pasymed"
   }
   control {
       name        "SoftMaster"
       card        0
   }
}

Let us have a look at our software mixer:

$amixer
Simple mixer control 'SoftMaster',0
 Capabilities: volume
 Playback channels: Front Left - Front Right
 Capture channels: Front Left - Front Right
 Limits: 0 - 255
 Front Left: 255 [100%]
 Front Right: 255 [100%]

Recording left and right input channels seperately

When recording from two mono inputs (Input 1/L and Input 2/R), the sound is mixed into a stereo stream at hardware level. Again, there is no software control over this audio device.

This is a problem when you only record from one mono microphone. The resulting stereo sound includes a muted channel with noise. At software level, you may downmix this sound to mono, but this degrades quality because of the muted channel with noise.

A simple solution is to record from left and right input channels seperately, as explained in the dsnoop howto. The better solution is to use the Mic/Guitar input for a mono dynamic or Mic Plug-In Powered input for a monaural or the R/L input for a stereo microphone (stereo might require input level adjustments outside the UA-1G!).

pcm.record_left {
   type        dsnoop
   ipc_key 234884
   slave {
       pcm     "plughw:UA1G"
       channels 2
   }
   bindings.0  0
}
pcm.record_right {
   type        dsnoop
   ipc_key 2241234
   slave {
       pcm     "plughw:UA1G"
       channels 2
   }    
   bindings.0  1
} 

Digital signals

The same rules apply here. Alsa has very little control over the UA-1G hardware. Digital Record must be enabled on the back with the switch "REC SOURCE" set to DIGITAL. Note: Not tested by me

Digital In

To record from digital source, you must select DIGITAL from the REC SOURCE switch on the back side of the UA-1G (near the SAMPLE RATE switches). You can then record using any Alsa tool, on the pasymed or SoftMaster PCMs. Note: Not tested by me

Digital Out

With the Roland Cakewalk UA-1G in Advanced mode off, you can play 16 bits, SAMPLE RATE (switches) digital output.

Let us try Alsa utility speaker-test:

speaker-test -c 2 -D softvol0 -twav

Connect an optical cable:

In the mixer of the Terratec Aureon, select Input 2 and check IEC958 In.

Record the digital stream using the following command:

arecord -v -f S16_LE -c 2 -D plughw:1,0 foobar.wav

Note: Not tested by me

Low latency (to be written)

Investigating. To be written.

Device information

The following section may help Alsa hackers:

cat/proc/bus/usb/devices

Please note that I removed all other USB devices from the below device output.

$cat /proc/bus/usb/devices

T:  Bus=03 Lev=01 Prnt=01 Port=00 Cnt=01 Dev#=  2 Spd=12  MxCh= 0
D:  Ver= 1.10 Cls=ff(vend.) Sub=00 Prot=ff MxPS= 8 #Cfgs=  1
P:  Vendor=0582 ProdID=00e9 Rev= 1.00
S:  Manufacturer=Roland
S:  Product=UA-1G
C:* #Ifs= 1 Cfg#= 1 Atr=80 MxPwr=200mA
I:* If#= 0 Alt= 0 #EPs= 0 Cls=ff(vend.) Sub=02 Prot=01 Driver=(none)
I:  If#= 0 Alt= 1 #EPs= 1 Cls=ff(vend.) Sub=02 Prot=01 Driver=(none)
E:  Ad=82(I) Atr=05(Isoc) MxPS= 608 Ivl=1ms

cat /proc/asound/devices

Please note that I removed all other audio devices from this output.

$cat /proc/asound/devices
 0: [ 0]   : control
 1:        : timer
16: [ 0- 0]: digital audio playback
24: [ 0- 0]: digital audio capture

cat /proc/asound/cards

Please note that I removed all other audio devices from this output.

$cat /proc/asound/cards
0 [UA1G           ]: USB-Audio - Cakewalk UA-1G
                     Roland Cakewalk UA-1G at usb-0000:00:1d.1-1, full speed

aplay -l

$aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: UA1G [Cakewalk UA-1G], device 0: USB Audio [USB Audio]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

$aplay -L
front:CARD=UA1G,DEV=0
    UA-1G, USB Audio
    Front speakers
surround40:CARD=UA1G,DEV=0
    UA-1G, USB Audio
    4.0 Surround output to Front and Rear speakers
surround41:CARD=UA1G,DEV=0
    UA-1G, USB Audio
    4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=UA1G,DEV=0
    UA-1G, USB Audio
    5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=UA1G,DEV=0
    UA-1G, USB Audio
    5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=UA1G,DEV=0
    UA-1G, USB Audio
    7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=UA1G,DEV=0
    UA-1G, USB Audio
    IEC958 (S/PDIF) Digital Audio Output

Getting Advanced mode to work

Dch24: I worked through these steps to get almost 100% support for Advanced mode. All that is missing is software-selectable Input Monitor when the switch is set to AUTO. (When it is set to ON, inputs are always sent to outputs.)

First, I checked /proc/bus/usb/devices. (Only relevant device(s))

$cat /proc/bus/usb/devices
T:  Bus=03 Lev=01 Prnt=01 Port=00 Cnt=01 Dev#=  2 Spd=12  MxCh= 0
D:  Ver= 1.10 Cls=ff(vend.) Sub=00 Prot=ff MxPS= 8 #Cfgs=  1
P:  Vendor=0582 ProdID=00e9 Rev= 1.00
S:  Manufacturer=Roland
S:  Product=UA-1G
C:* #Ifs= 1 Cfg#= 1 Atr=80 MxPwr=200mA
I:* If#= 0 Alt= 0 #EPs= 0 Cls=ff(vend.) Sub=02 Prot=01 Driver=(none)
I:  If#= 0 Alt= 1 #EPs= 1 Cls=ff(vend.) Sub=02 Prot=01 Driver=(none)
E:  Ad=82(I) Atr=05(Isoc) MxPS= 608 Ivl=1ms

Next I checked the UA-25 Device Information section and my usb listing seems close enough.

You don't need to patch your driver or kernel if you have at least alsa-driver-1.0.22.1 installed (version check: /proc/asound/version).

Looking at /usr/src/linux-2.6.27.19/sound/usb/usbaudio.c and usbquirks.h I made the following patch, tested on linux-2.6.27.19 (distribution kernel), based on the newer alsa-driver-1.0.19:

Add Alsa support for Roland Cakewalk UA-1G in Advanced mode
(for sample rates of 48 kHz and 96 kHz)
usbquirks.h
===================================================================
diff -uNr alsa-kernel/usb/usbquirks.h sound/usb/usbquirks.h
--- alsa-kernel/usb/usbquirks.h~        2009-01-19 12:08:58.000000000 +0100
+++ alsa-kernel/usb/usbquirks.h 2009-03-29 14:30:39.000000000 +0200
@@ -1528,6 +1528,37 @@
                }
        }
 },
+{
+       /* Only needed in "Advanced Driver" mode 
+        * For the standard mode, Cakewalk UA-1G has ID 0582:00ea, which
+        * offers only 16-bit PCM at set SAMPLE RATE (switches).
+        * No mixers available with this quirk in Advanced Driver mode!
+        */
+       USB_DEVICE(0x0582, 0x00e9),
+       .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+               .vendor_name = "Roland",
+               .product_name = "Cakewalk UA-1G",
+               .ifnum = QUIRK_ANY_INTERFACE,
+               .type = QUIRK_COMPOSITE,
+               .data = (const struct snd_usb_audio_quirk[]) {
+                       {
+                               .ifnum = 0,
+                               .type = QUIRK_AUDIO_EDIROL_UAXX
+                       },
+                       {
+                               .ifnum = 1,
+                               .type = QUIRK_AUDIO_EDIROL_UAXX
+                       },
+                       {
+                               .ifnum = -1,
+                               .type = QUIRK_AUDIO_EDIROL_UAXX
+                       },
+                       {
+                               .ifnum = -1
+                       }
+               }
+       }
+},

 /* Guillemot devices */
 {

When I plug in the card, I get the following in the kernel log: kernel: usb 3-1: USB disconnect, address 4 kernel: usb 3-1: new full speed USB device using uhci_hcd and address 5 kernel: usb 3-1: configuration #1 chosen from 1 choice kernel: usb 3-1: New USB device found, idVendor=0582, idProduct=00e9 kernel: usb 3-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 kernel: usb 3-1: Product: UA-1G kernel: usb 3-1: Manufacturer: Roland

The snd_usb_audio driver correctly detects the playback and capture sample rate:

$cat /proc/asound/cards
 0 [UA1G           ]: USB-Audio - Cakewalk UA-1G
                      Roland Cakewalk UA-1G at usb-0000:00:0b.0-6, full speed
$cat /proc/asound/card0/stream0 # correctly detects 32000, 44100, 48000 based on position of SAMPLE RATE switches
Roland Cakewalk UA-1G at usb-0000:00:0b.0-6, full speed : USB Audio

Playback:
  Status: Stop
  Interface 0
    Altset 1
    Format: 0x20
    Channels: 2
    Endpoint: 1 OUT (ADAPTIVE)
    Rates: 48000 - 48000 (continuous)

Capture:
  Status: Stop
  Interface 1
    Altset 1
    Format: 0x20
    Channels: 2
    Endpoint: 2 IN (ASYNC)
    Rates: 48000 - 48000 (continuous)

$cat /proc/asound/card0/pcm0p/sub0/hw_params # playing a 44.1 kHz .wav, card at 44.1 kHz
access: MMAP_INTERLEAVED
format: S24_3LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 6000
buffer_size: 240001
$cat /proc/asound/card0/pcm0p/sub0/hw_params # playing a 44.1 kHz .wav, card at 48 kHz
access: MMAP_INTERLEAVED
format: S24_3LE
subformat: STD
channels: 2
rate: 48000 (48000/1)
period_size: 6000
buffer_size: 240001
$cat /proc/asound/card0/pcm0c/sub0/hw_params # recording a .wav, card at 96 kHz RECORD
access: MMAP_INTERLEAVED
format: S24_3LE
subformat: STD
channels: 2
rate: 96000 (96000/1)
period_size: 6000
buffer_size: 240001
$cat /proc/asound/card0/pcm0p/sub0/hw_params # playing a 44.1 kHz .wav, card at 96 kHz PLAY
access: MMAP_INTERLEAVED
format: S24_3LE
subformat: STD
channels: 2
rate: 96000 (96000/1)
period_size: 6000
buffer_size: 240001

Switching to 48 kHz playback/recording, /proc/bus/usb/devices has 1 more endpoint:

C:* #Ifs= 2 Cfg#= 1 Atr=80 MxPwr=200mA
I:* If#= 0 Alt= 0 #EPs= 0 Cls=ff(vend.) Sub=02 Prot=02 Driver=snd-usb-audio
I:  If#= 0 Alt= 1 #EPs= 1 Cls=ff(vend.) Sub=02 Prot=02 Driver=snd-usb-audio
E:  Ad=01(O) Atr=09(Isoc) MxPS= 320 Ivl=1ms
I:* If#= 1 Alt= 0 #EPs= 0 Cls=ff(vend.) Sub=02 Prot=01 Driver=snd-usb-audio
I:  If#= 1 Alt= 1 #EPs= 1 Cls=ff(vend.) Sub=02 Prot=01 Driver=snd-usb-audio
E:  Ad=82(I) Atr=05(Isoc) MxPS= 320 Ivl=1ms

Obviously, at 96 kHz the interfaces list is quite different. But everything is detected fine, just like the UA-25.

\

Getting Advanced mode to work (alternative way)

As I don't understand how to patch a kernel because I am more musician than a Unix programmer, I found another way to get the advanced mode to work. You can use the alternative driver from Michael Minn MMUSBAUDIO (http://michaelminn.com/linux/mmusbaudio/) which is dedicated to Roland and Edirol products. As I believed that the Cakewalk UA-1G was very close to the Edirol UA-1EX for the hardware, I guessed that modifying the source code a little bit could do the trick. It worked. So you have to modify the source to add support for the UA-1G. Nothing impossible, don't worry.

UPDATE : The driver from Michael Minn has been updated and integrated the following fix. You shouldn't have to modify anything now. Just compile the driver with "make" and install it with "make install".

\ Prerequisites : You have to know the manufacturer ID and device ID of your sound card. It can be a little different from mine. Mine is like that :

Manufacturer : 0x0582    DeviceID : 0x00EA (for basic mode)
Manufacturer : 0x0582    DeviceID : 0x00E9 (for advanced driver mode)

You can find those values by doing unplugging the soundcard, plugging it again in basic mode and type the following in the console :

$cat /proc/bus/usb/devices

Then you can find those values for the device whose manufacturer is Roland. Note the manufacturer id and device id. - Do the same thing but in advanced driver mode.

Once you have the manufacturer and device ID, you can continue :

​1) Download and extract the driver to a directory.

​2) Modify the following files :

\ mmusbaudio.c :

\ Near line 2035 :

else if ((device->descriptor.idVendor == 0x582) && (interface == 1) &&
 (device->descriptor.idProduct == 0xEA))
 driver_data = mmusbaudio_assign_audio_device(device, "UA-1G (Basic Mode)");

else if ((device->descriptor.idVendor == 0x582) && (interface == 1) &&
 (device->descriptor.idProduct == 0xE9))
 driver_data = mmusbaudio_assign_audio_device(device, "UA-1G (Advanced Mode)");

\ Near line 2120 :

static struct usb_device_id mmusbaudio_ids [] =
{
 { USB_DEVICE_VER(0x0582, 0x0000, 0x0, 0xffff) }, /* UA-100 */
 { USB_DEVICE_VER(0x0582, 0x0003, 0x0, 0xffff) }, /* SC-8850 */
 { USB_DEVICE_VER(0x0582, 0x0009, 0x0, 0xffff) }, /* UM-1 */
 { USB_DEVICE_VER(0x0582, 0x0010, 0x0, 0xffff) }, /* UA-5 (Advanced Mode) */
 { USB_DEVICE_VER(0x0582, 0x0011, 0x0, 0xffff) }, /* UA-5 (Basic Mode) */
 { USB_DEVICE_VER(0x0582, 0x0096, 0x0, 0xffff) }, /* UA-1EX */
 { USB_DEVICE_VER(0x0582, 0x00EA, 0x0, 0xffff) }, /* UA-1G basic */
 { USB_DEVICE_VER(0x0582, 0x00E9, 0x0, 0xffff) }, /* UA-1G advanced */
 { USB_DEVICE_VER(0x1210, 0x0011, 0x0, 0xffff) }, /* DigiTech GSP 1101 */
 { } /* Terminating entry */
};

\ mmusbaudio.mod.c (you maybe need to be root to modify that one) : After those lines :

MODULE_ALIAS("usb:v0582p0000d*dc*dsc*dp*ic*isc*ip*");
MODULE_ALIAS("usb:v0582p0003d*dc*dsc*dp*ic*isc*ip*");
MODULE_ALIAS("usb:v0582p0009d*dc*dsc*dp*ic*isc*ip*");
MODULE_ALIAS("usb:v0582p0010d*dc*dsc*dp*ic*isc*ip*");
MODULE_ALIAS("usb:v0582p0011d*dc*dsc*dp*ic*isc*ip*");
MODULE_ALIAS("usb:v0582p0096d*dc*dsc*dp*ic*isc*ip*");
MODULE_ALIAS("usb:v1210p0011d*dc*dsc*dp*ic*isc*ip*");

Add the following two lines :

MODULE_ALIAS("usb:v0582p00EAd*dc*dsc*dp*ic*isc*ip*"); /*UA-1G (basic mode)*/
MODULE_ALIAS("usb:v0582p00E9d*dc*dsc*dp*ic*isc*ip*"); /*UA-1G (advanced driver mode)*/

​3) Compile and install :

cd /home/gabriel/mmusbaudio/ (that's where I extracted the driver, will be different for you of course)
make
make install
make load

​4) Under Debian 5.01 I had to modify the following file to load the driver at each startup : /etc/modprobe.d/sound (can be /etc/modules.conf or /etc/modprobe.conf on other distributions, please replace with appropriate file...)

gedit /etc/modprobe.d/sound

I commented everything and added two lines. Example : I had a HDA chip on my laptop. I commented it. You can comment everything and add the two last lines.

#alias snd-card-0 snd-hda-intel
#options snd-hda-intel index=0
alias sound-slot-0 mmusbaudio
alias sound-service-0-0 off

​5) Restart your computer (really, don't try to cheat)

​6) Now choose OSS as your audio interface. It should work in advanced mode !

ToDo

See also

Retrieved from "http://alsa.opensrc.org/Cakewalk_UA-1G"

Category: Sound cards

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