audiophile-usb.txt

linux 内核源代码

	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.5
	========================================================

	    Thibault Le Meur <[e-mail]>

This document is a guide to using the M-Audio Audiophile USB (tm) device with 
ALSA and JACK.

History
=======
* v1.4 - Thibault Le Meur (2007-07-11)
 - Added Low Endianness nature of 16bits-modes
   found by Hakan Lennestal <[e-mail]>
 - Modifying document structure
* v1.5 - Thibault Le Meur (2007-07-12)
 - Added AC3/DTS passthru info


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 interfaces 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 MIDI support in ALSA
=======================================

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.


3 - Audiophile USB Audio support in ALSA
========================================

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".

3.1 - Default Alsa driver mode
------------------------------

The default behavior of the snd-usb-audio driver is to list the device 
capabilities at startup and activate the required mode when required 
by the applications: for instance if the user is recording in a 
24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode,
the snd-usb-audio module will reconfigure the device on the fly.

This approach has the advantage to let the driver automatically switch from sample 
rates/depths automatically according to the user's needs. However, those who 
are using the device under windows know that this is not how the device is meant to
work: under windows applications must be closed before using the m-audio control
panel to switch the device working mode. Thus as we'll see in next section, this 
Default Alsa driver mode can lead to device misconfigurations.

Let's get back to the Default Alsa driver mode for now.  In this case the 
Audiophile interfaces 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

In this mode, 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 was reported to be Little Endian 
compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams.
This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface 
is reported to be big endian in this default driver mode.

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
 * playing an ac3 sample file to the Do port
   % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw

If you're happy with the default Alsa driver mode and don't experience any 
issue with this mode, then you can skip the following chapter.

3.2 - 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 interface 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" (this parameter was introduced in kernel
release 2.6.17)

3.2.1 - 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 

When initialized with "device_setup=0x00", the snd-usb-audio module has
the same behaviour as when the parameter is omitted (see paragraph "Default 
Alsa driver mode" above)

Others modes are described in the following subsections.

3.2.1.1 - 16-bit modes

The two supported modes are:

 * 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

In this modes the device operates only at 16bits-modes. Before kernel 2.6.23,
the devices where reported to be Big-Endian when in fact they were Little-Endian
so that playing a file was a matter of using:
   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw
where "test_S16_LE.raw" was in fact a little-endian sample file.

Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in
these modes) a fix has been committed (expected in kernel 2.6.23) and
Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as
using:
   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw

3.2.1.2 - 24-bit modes

The three supported modes are:

 * 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

In these modes the device is only Big-Endian compliant (see "Default Alsa driver 
mode" above for an aplay command example)

3.2.1.3 - AC3 w/ DTS passthru mode

Thanks to Hakan Lennestal, I now have a report saying that this mode works.

 * device_setup=0x03
   - 16bits 48kHz mode with only the Do port enabled 
   - AC3 with DTS passthru
   - Caution with this setup the Do port is mapped to the pcm device hw:1,0

The command line used to playback the AC3/DTS encoded .wav-files in this mode: