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[ALSA] usb-audio - Fix audiophile-USB quirk for little-endian 

Audiophile-usb fix (corrects little-endianness in 16bit
modes, resets interfaces at device initialization, and updates the
documentation).

Signed-off-by: Thibault Le Meur <Thibault.LeMeur@supelec.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
wifi-calibration
Thibault Le Meur 2007-07-12 11:26:35 +02:00 committed by Jaroslav Kysela
parent be38114a49
commit f8c78b82b9
2 changed files with 157 additions and 78 deletions
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226
Documentation/sound/alsa/Audiophile-Usb.txt
View File

@ -1,4 +1,4 @@
Guide to using M-Audio Audiophile USB with ALSA and Jack v1.3
Guide to using M-Audio Audiophile USB with ALSA and Jack v1.4
========================================================
Thibault Le Meur <Thibault.LeMeur@supelec.fr>
@ -6,8 +6,17 @@
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 <Hakan.Lennestal@brfsodrahamn.se>
- Modifying document structure
1 - Audiophile USB Specs and correct usage
==========================================
This part is a reminder of important facts about the functions and limitations
of the device.
@ -25,18 +34,18 @@ The device has 4 audio interfaces, and 2 MIDI ports:
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"
* 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
* 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
* 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)
* 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:
@ -52,44 +61,53 @@ source is connected
synchronization error (for instance sound played at an odd sample rate)
2 - Audiophile USB support in ALSA
==================================
2 - Audiophile USB MIDI support in ALSA
=======================================
2.1 - MIDI ports
----------------
The Audiophile USB MIDI ports will be automatically supported once the
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
-----------------
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".
2.2.1 - Default Alsa driver mode
3.1 - 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.
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.
In this case the Audiophile ports are mapped to alsa pcm devices in the
following way (I suppose the device's index is 1):
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
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.
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 is reported
to be Little Endian compliant (supposedly supporting S16_LE) but processes
in fact only S16_BE streams.
Examples:
* playing a S24_3BE encoded raw file to the Ao port
@ -99,21 +117,23 @@ Examples:
* 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
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.
2.2.2 - Advanced module setup
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 port sounds distorted (as if boosted with an excessive high volume
gain).
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".
parameter called "device_setup" (this parameter was introduced in kernel
release 2.6.17)
2.2.2.1 - Initializing the working mode of the Audiophile USB
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:
@ -121,33 +141,57 @@ specify:
* 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 described earlier
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
@ -155,34 +199,61 @@ Here is a list of supported device_setup values for this device:
- 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
This mode is untested, I have no AC3 compliant device to test it. I uses:
* 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
3.2.2 - How to use 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:
CAUTION when initializaing the device
-------------------------------------
* Correct initialization on the device requires that device_setup is given to
the module BEFORE the device is turned on. So, if you use the "manual probing"
method described above, take care to power-on the device AFTER this initialization.
* Failing to respect this will lead in a misconfiguration of the device. In this case
turn off the device, unproble the snd-usb-audio module, then probe it again with
correct device_setup parameter and then (and only then) turn on the device again.
* If you've correctly initialized the device in a valid mode and then want to switch
to another mode (possibly with another sample-depth), please use also the following
procedure:
- 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
* A workaround for this last issue has been applied to kernel 2.6.23, but it may not
be enough to ensure the 'stability' of the device initialization.
2.2.2.3 - Audiophile USB's device_setup structure
3.2.3 - Technical details for hackers
-------------------------------------
This section is for hackers, wanting to understand details about the device
internals and how Alsa supports it.
3.2.3.1 - 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,
@ -228,12 +299,12 @@ Caution:
- 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
3.2.3.2 - USB implementation details for this device
You may safely skip this section if you're not interested in driver
development.
hacking.
This section describes some internal aspects of the device and summarize the
This section describes some internal aspects of the device and summarizes the
data I got by usb-snooping the windows and Linux drivers.
The M-Audio Audiophile USB has 7 USB Interfaces:
@ -293,43 +364,45 @@ 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
4 - 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
--------------------------------
There are 2 main potential issues when using Jackd with the device:
* support for Big-Endian devices in 24-bit modes
* support for 4-in / 4-out channels
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.
4.1 - Direct support in Jackd
-----------------------------
Jack supports big endian devices only in recent versions (thanks to
Andreas Steinmetz for his first big-endian patch). I can't remember
extacly when this support was released into jackd, let's just say that
with jackd version 0.103.0 it's almost ok (just a small bug is affecting
16bits Big-Endian devices, but since you've read carefully the above
paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices
are now Little Endians ;-) ).
You can run jackd with the following command for playback with Ao and
record with Ai:
% jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
4.2 - Using Alsa plughw
-----------------------
If you don't have a recent Jackd installed, you can downgrade to using
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
4.3 - Getting 2 input and/or output interfaces in Jack
------------------------------------------------------
As you can see, starting the Jack server this way will only enable 1 stereo
@ -339,6 +412,7 @@ 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.
@ -348,13 +422,11 @@ 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)
* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page)
* (maybe) 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.
I had no success in testing this for now, if you have any success with this kind
of setup, please drop me an email.

9
sound/usb/usbaudio.c
View File

@ -2350,7 +2350,9 @@ static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *
return 1;
break;
case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
return 1;
if (device_setup[chip->index] == 0x00 ||
fp->altsetting==1 || fp->altsetting==2 || fp->altsetting==3)
return 1;
}
return 0;
}
@ -3251,6 +3253,11 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
int iface, int altno)
{
/* Reset ALL ifaces to 0 altsetting.
* Call it for every possible altsetting of every interface.
*/
usb_set_interface(chip->dev, iface, 0);
if (device_setup[chip->index] & AUDIOPHILE_SET) {
if ((device_setup[chip->index] & AUDIOPHILE_SET_DTS)
&& altno != 6)