mirror of https://gitee.com/openkylin/linux.git
361 lines
14 KiB
Plaintext
361 lines
14 KiB
Plaintext
Guide to using M-Audio Audiophile USB with ALSA and Jack v1.3
|
|
========================================================
|
|
|
|
Thibault Le Meur <Thibault.LeMeur@supelec.fr>
|
|
|
|
This document is a guide to using the M-Audio Audiophile USB (tm) device with
|
|
ALSA and JACK.
|
|
|
|
1 - Audiophile USB Specs and correct usage
|
|
==========================================
|
|
This part is a reminder of important facts about the functions and limitations
|
|
of the device.
|
|
|
|
The device has 4 audio interfaces, and 2 MIDI ports:
|
|
* Analog Stereo Input (Ai)
|
|
- This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA)
|
|
- When the 1/4" TS (jack) connectors are connected, the RCA connectors
|
|
are disabled
|
|
* Analog Stereo Output (Ao)
|
|
* Digital Stereo Input (Di)
|
|
* Digital Stereo Output (Do)
|
|
* Midi In (Mi)
|
|
* Midi Out (Mo)
|
|
|
|
The internal DAC/ADC has the following characteristics:
|
|
* sample depth of 16 or 24 bits
|
|
* sample rate from 8kHz to 96kHz
|
|
* Two ports can't use different sample depths at the same time. Moreover, the
|
|
Audiophile USB documentation gives the following Warning: "Please exit any
|
|
audio application running before switching between bit depths"
|
|
|
|
Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be
|
|
activated at the same time depending on the audio mode selected:
|
|
* 16-bit/48kHz ==> 4 channels in/4 channels out
|
|
- Ai+Ao+Di+Do
|
|
* 24-bit/48kHz ==> 4 channels in/2 channels out,
|
|
or 2 channels in/4 channels out
|
|
- Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do
|
|
* 24-bit/96kHz ==> 2 channels in, or 2 channels out (half duplex only)
|
|
- Ai or Ao or Di or Do
|
|
|
|
Important facts about the Digital interface:
|
|
--------------------------------------------
|
|
* The Do port additionally supports surround-encoded AC-3 and DTS passthrough,
|
|
though I haven't tested it under Linux
|
|
- Note that in this setup only the Do interface can be enabled
|
|
* Apart from recording an audio digital stream, enabling the Di port is a way
|
|
to synchronize the device to an external sample clock
|
|
- As a consequence, the Di port must be enable only if an active Digital
|
|
source is connected
|
|
- Enabling Di when no digital source is connected can result in a
|
|
synchronization error (for instance sound played at an odd sample rate)
|
|
|
|
|
|
2 - Audiophile USB support in ALSA
|
|
==================================
|
|
|
|
2.1 - MIDI ports
|
|
----------------
|
|
The Audiophile USB MIDI ports will be automatically supported once the
|
|
following modules have been loaded:
|
|
* snd-usb-audio
|
|
* snd-seq-midi
|
|
|
|
No additional setting is required.
|
|
|
|
2.2 - Audio ports
|
|
-----------------
|
|
|
|
Audio functions of the Audiophile USB device are handled by the snd-usb-audio
|
|
module. This module can work in a default mode (without any device-specific
|
|
parameter), or in an "advanced" mode with the device-specific parameter called
|
|
"device_setup".
|
|
|
|
2.2.1 - Default Alsa driver mode
|
|
|
|
The default behavior of the snd-usb-audio driver is to parse the device
|
|
capabilities at startup and enable all functions inside the device (including
|
|
all ports at any supported sample rates and sample depths). This approach
|
|
has the advantage to let the driver easily switch from sample rates/depths
|
|
automatically according to the need of the application claiming the device.
|
|
|
|
In this case the Audiophile ports are mapped to alsa pcm devices in the
|
|
following way (I suppose the device's index is 1):
|
|
* hw:1,0 is Ao in playback and Di in capture
|
|
* hw:1,1 is Do in playback and Ai in capture
|
|
* hw:1,2 is Do in AC3/DTS passthrough mode
|
|
|
|
You must note as well that the device uses Big Endian byte encoding so that
|
|
supported audio format are S16_BE for 16-bit depth modes and S24_3BE for
|
|
24-bits depth mode. One exception is the hw:1,2 port which is Little Endian
|
|
compliant and thus uses S16_LE.
|
|
|
|
Examples:
|
|
* playing a S24_3BE encoded raw file to the Ao port
|
|
% aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw
|
|
* recording a S24_3BE encoded raw file from the Ai port
|
|
% arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw
|
|
* playing a S16_BE encoded raw file to the Do port
|
|
% aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw
|
|
|
|
If you're happy with the default Alsa driver setup and don't experience any
|
|
issue with this mode, then you can skip the following chapter.
|
|
|
|
2.2.2 - Advanced module setup
|
|
|
|
Due to the hardware constraints described above, the device initialization made
|
|
by the Alsa driver in default mode may result in a corrupted state of the
|
|
device. For instance, a particularly annoying issue is that the sound captured
|
|
from the Ai port sounds distorted (as if boosted with an excessive high volume
|
|
gain).
|
|
|
|
For people having this problem, the snd-usb-audio module has a new module
|
|
parameter called "device_setup".
|
|
|
|
2.2.2.1 - Initializing the working mode of the Audiophile USB
|
|
|
|
As far as the Audiophile USB device is concerned, this value let the user
|
|
specify:
|
|
* the sample depth
|
|
* the sample rate
|
|
* whether the Di port is used or not
|
|
|
|
Here is a list of supported device_setup values for this device:
|
|
* device_setup=0x00 (or omitted)
|
|
- Alsa driver default mode
|
|
- maintains backward compatibility with setups that do not use this
|
|
parameter by not introducing any change
|
|
- results sometimes in corrupted sound as decribed earlier
|
|
* device_setup=0x01
|
|
- 16bits 48kHz mode with Di disabled
|
|
- Ai,Ao,Do can be used at the same time
|
|
- hw:1,0 is not available in capture mode
|
|
- hw:1,2 is not available
|
|
* device_setup=0x11
|
|
- 16bits 48kHz mode with Di enabled
|
|
- Ai,Ao,Di,Do can be used at the same time
|
|
- hw:1,0 is available in capture mode
|
|
- hw:1,2 is not available
|
|
* device_setup=0x09
|
|
- 24bits 48kHz mode with Di disabled
|
|
- Ai,Ao,Do can be used at the same time
|
|
- hw:1,0 is not available in capture mode
|
|
- hw:1,2 is not available
|
|
* device_setup=0x19
|
|
- 24bits 48kHz mode with Di enabled
|
|
- 3 ports from {Ai,Ao,Di,Do} can be used at the same time
|
|
- hw:1,0 is available in capture mode and an active digital source must be
|
|
connected to Di
|
|
- hw:1,2 is not available
|
|
* device_setup=0x0D or 0x10
|
|
- 24bits 96kHz mode
|
|
- Di is enabled by default for this mode but does not need to be connected
|
|
to an active source
|
|
- Only 1 port from {Ai,Ao,Di,Do} can be used at the same time
|
|
- hw:1,0 is available in captured mode
|
|
- hw:1,2 is not available
|
|
* device_setup=0x03
|
|
- 16bits 48kHz mode with only the Do port enabled
|
|
- AC3 with DTS passthru (not tested)
|
|
- Caution with this setup the Do port is mapped to the pcm device hw:1,0
|
|
|
|
2.2.2.2 - Setting and switching configurations with the device_setup parameter
|
|
|
|
The parameter can be given:
|
|
* By manually probing the device (as root):
|
|
# modprobe -r snd-usb-audio
|
|
# modprobe snd-usb-audio index=1 device_setup=0x09
|
|
* Or while configuring the modules options in your modules configuration file
|
|
- For Fedora distributions, edit the /etc/modprobe.conf file:
|
|
alias snd-card-1 snd-usb-audio
|
|
options snd-usb-audio index=1 device_setup=0x09
|
|
|
|
IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
|
|
-------------------------------------------
|
|
* You may need to _first_ initialize the module with the correct device_setup
|
|
parameter and _only_after_ turn on the Audiophile USB device
|
|
* This is especially true when switching the sample depth:
|
|
- first turn off the device
|
|
- de-register the snd-usb-audio module (modprobe -r)
|
|
- change the device_setup parameter by changing the device_setup
|
|
option in /etc/modprobe.conf
|
|
- turn on the device
|
|
|
|
2.2.2.3 - Audiophile USB's device_setup structure
|
|
|
|
If you want to understand the device_setup magic numbers for the Audiophile
|
|
USB, you need some very basic understanding of binary computation. However,
|
|
this is not required to use the parameter and you may skip this section.
|
|
|
|
The device_setup is one byte long and its structure is the following:
|
|
|
|
+---+---+---+---+---+---+---+---+
|
|
| b7| b6| b5| b4| b3| b2| b1| b0|
|
|
+---+---+---+---+---+---+---+---+
|
|
| 0 | 0 | 0 | Di|24B|96K|DTS|SET|
|
|
+---+---+---+---+---+---+---+---+
|
|
|
|
Where:
|
|
* b0 is the "SET" bit
|
|
- it MUST be set if device_setup is initialized
|
|
* b1 is the "DTS" bit
|
|
- it is set only for Digital output with DTS/AC3
|
|
- this setup is not tested
|
|
* b2 is the Rate selection flag
|
|
- When set to "1" the rate range is 48.1-96kHz
|
|
- Otherwise the sample rate range is 8-48kHz
|
|
* b3 is the bit depth selection flag
|
|
- When set to "1" samples are 24bits long
|
|
- Otherwise they are 16bits long
|
|
- Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits
|
|
samples
|
|
* b4 is the Digital input flag
|
|
- When set to "1" the device assumes that an active digital source is
|
|
connected
|
|
- You shouldn't enable Di if no source is seen on the port (this leads to
|
|
synchronization issues)
|
|
- b4 is implied by b2 (since only one port is enabled at a time no synch
|
|
error can occur)
|
|
* b5 to b7 are reserved for future uses, and must be set to "0"
|
|
- might become Ao, Do, Ai, for b7, b6, b4 respectively
|
|
|
|
Caution:
|
|
* there is no check on the value you will give to device_setup
|
|
- for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since
|
|
b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages
|
|
* Hardware constraints due to the USB bus limitation aren't checked
|
|
- choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
|
|
only be able to use one at the same time
|
|
|
|
2.2.3 - USB implementation details for this device
|
|
|
|
You may safely skip this section if you're not interested in driver
|
|
development.
|
|
|
|
This section describes some internal aspects of the device and summarize the
|
|
data I got by usb-snooping the windows and Linux drivers.
|
|
|
|
The M-Audio Audiophile USB has 7 USB Interfaces:
|
|
a "USB interface":
|
|
* USB Interface nb.0
|
|
* USB Interface nb.1
|
|
- Audio Control function
|
|
* USB Interface nb.2
|
|
- Analog Output
|
|
* USB Interface nb.3
|
|
- Digital Output
|
|
* USB Interface nb.4
|
|
- Analog Input
|
|
* USB Interface nb.5
|
|
- Digital Input
|
|
* USB Interface nb.6
|
|
- MIDI interface compliant with the MIDIMAN quirk
|
|
|
|
Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
|
|
* Interface 3 (Digital Out) has an extra Alset nb.6
|
|
* Interface 5 (Digital In) does not have Alset nb.3 and 5
|
|
|
|
Here is a short description of the AltSettings capabilities:
|
|
* AltSettings 1 corresponds to
|
|
- 24-bit depth, 48.1-96kHz sample mode
|
|
- Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di)
|
|
* AltSettings 2 corresponds to
|
|
- 24-bit depth, 8-48kHz sample mode
|
|
- Asynch capture and playback (Ao,Ai,Do,Di)
|
|
* AltSettings 3 corresponds to
|
|
- 24-bit depth, 8-48kHz sample mode
|
|
- Synch capture (Ai) and Adaptive playback (Ao,Do)
|
|
* AltSettings 4 corresponds to
|
|
- 16-bit depth, 8-48kHz sample mode
|
|
- Asynch capture and playback (Ao,Ai,Do,Di)
|
|
* AltSettings 5 corresponds to
|
|
- 16-bit depth, 8-48kHz sample mode
|
|
- Synch capture (Ai) and Adaptive playback (Ao,Do)
|
|
* AltSettings 6 corresponds to
|
|
- 16-bit depth, 8-48kHz sample mode
|
|
- Synch playback (Do), audio format type III IEC1937_AC-3
|
|
|
|
In order to ensure a correct initialization of the device, the driver
|
|
_must_know_ how the device will be used:
|
|
* if DTS is chosen, only Interface 2 with AltSet nb.6 must be
|
|
registered
|
|
* if 96KHz only AltSets nb.1 of each interface must be selected
|
|
* if samples are using 24bits/48KHz then AltSet 2 must me used if
|
|
Digital input is connected, and only AltSet nb.3 if Digital input
|
|
is not connected
|
|
* if samples are using 16bits/48KHz then AltSet 4 must me used if
|
|
Digital input is connected, and only AltSet nb.5 if Digital input
|
|
is not connected
|
|
|
|
When device_setup is given as a parameter to the snd-usb-audio module, the
|
|
parse_audio_endpoints function uses a quirk called
|
|
"audiophile_skip_setting_quirk" in order to prevent AltSettings not
|
|
corresponding to device_setup from being registered in the driver.
|
|
|
|
3 - Audiophile USB and Jack support
|
|
===================================
|
|
|
|
This section deals with support of the Audiophile USB device in Jack.
|
|
The main issue regarding this support is that the device is Big Endian
|
|
compliant.
|
|
|
|
3.1 - Using the plug alsa plugin
|
|
--------------------------------
|
|
|
|
Jack doesn't directly support big endian devices. Thus, one way to have support
|
|
for this device with Alsa is to use the Alsa "plug" converter.
|
|
|
|
For instance here is one way to run Jack with 2 playback channels on Ao and 2
|
|
capture channels from Ai:
|
|
% jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
|
|
|
|
|
|
However you may see the following warning message:
|
|
"You appear to be using the ALSA software "plug" layer, probably a result of
|
|
using the "default" ALSA device. This is less efficient than it could be.
|
|
Consider using a hardware device instead rather than using the plug layer."
|
|
|
|
3.2 - Patching alsa to use direct pcm device
|
|
--------------------------------------------
|
|
A patch for Jack by Andreas Steinmetz adds support for Big Endian devices.
|
|
However it has not been included in the CVS tree.
|
|
|
|
You can find it at the following URL:
|
|
http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687&
|
|
atid=425939
|
|
|
|
After having applied the patch you can run jackd with the following command
|
|
line:
|
|
% jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
|
|
|
|
3.2 - Getting 2 input and/or output interfaces in Jack
|
|
------------------------------------------------------
|
|
|
|
As you can see, starting the Jack server this way will only enable 1 stereo
|
|
input (Di or Ai) and 1 stereo output (Ao or Do).
|
|
|
|
This is due to the following restrictions:
|
|
* Jack can only open one capture device and one playback device at a time
|
|
* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1
|
|
(and optionally hw:1,2)
|
|
If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to
|
|
combine the Alsa devices into one logical "complex" device.
|
|
|
|
If you want to give it a try, I recommend reading the information from
|
|
this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html
|
|
It is related to another device (ice1712) but can be adapted to suit
|
|
the Audiophile USB.
|
|
|
|
Enabling multiple Audiophile USB interfaces for Jackd will certainly require:
|
|
* patching Jack with the previously mentioned "Big Endian" patch
|
|
* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page)
|
|
* patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
|
|
* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc
|
|
file
|
|
* start jackd with this device
|
|
|
|
I had no success in testing this for now, but this may be due to my OS
|
|
configuration. If you have any success with this kind of setup, please
|
|
drop me an email.
|