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+ Guide to using M-Audio Audiophile USB with ALSA and Jack v1.5
+ ========================================================
+ Thibault Le Meur <Thibault.LeMeur@supelec.fr>
+This document is a guide to using the M-Audio Audiophile USB (tm) device with
+* 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
+* 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
+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.
+ * 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
+ * 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.
+ - 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
+ % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw
+ - 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)
+ - 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:
+ % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw
+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
+ (tipically a .conf file in /etc/modprobe.d/ directory:
+ alias snd-card-1 snd-usb-audio
+ options snd-usb-audio index=1 device_setup=0x09
+CAUTION when initializing 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 to a misconfiguration of the device. In this case
+ turn off the device, unprobe 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.d/*.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.
+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.
+ - 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|
+ +---+---+---+---+---+---+---+---+
+ * 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
+ * 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
+ - USB implementation details for this device
+You may safely skip this section if you're not interested in driver
+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:
+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.
+4 - Audiophile USB and Jack support
+This section deals with support of the Audiophile USB device in Jack.
+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
+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
+exactly 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."
+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
+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:
+* 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, if you have any success with this kind
+of setup, please drop me an email.