Documentation/sound/alsa/Audiophile-Usb.txt - LeafOS-Devices/android_kernel_samsung_gta4xl - Gitiles

 	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.1
 	========================================================

 	    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)
  * 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 caracteristics:
 * 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 additionnaly 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 syncrhonize 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
  * snd-seq-midi

 No additionnal 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 behaviour 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 sample rates and any sample depths supported). 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 Audiohile USB

 As far as the Audiohile 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_ intialize 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 trun off the device
    - de-register the snd-usb-audio module
    - change the device_setup parameter (by either manually reprobing the module
      or changing modprobe.conf)
    - turn on the device

 2.2.2.3 - Setting and switching configurations with the device_setup parameter

 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 thi 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 -  Technical Details for Audiophile Usb

 You may safely skip this section if you're not interrested in driver
 development.

 This section describes some internals aspect 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 intialization of the device, the driver
 _must_know_ how the device will be used:
  * if DTS is choosen, 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_enpoint 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:
 # /usr/local/bin/jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
	Guide to using M-Audio Audiophile USB with ALSA and Jack v1.1
	========================================================

	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)
	* 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 caracteristics:
	* 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 additionnaly 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 syncrhonize 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
	* snd-seq-midi

	No additionnal 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 behaviour 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 sample rates and any sample depths supported). 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 Audiohile USB

	As far as the Audiohile 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_ intialize 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 trun off the device
	- de-register the snd-usb-audio module
	- change the device_setup parameter (by either manually reprobing the module
	or changing modprobe.conf)
	- turn on the device

	2.2.2.3 - Setting and switching configurations with the device_setup parameter

	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 thi 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 - Technical Details for Audiophile Usb

	You may safely skip this section if you're not interrested in driver
	development.

	This section describes some internals aspect 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 intialization of the device, the driver
	_must_know_ how the device will be used:
	* if DTS is choosen, 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_enpoint 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:
	# /usr/local/bin/jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1