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Another simple conversion from a plain text file. Put to cards directory. Signed-off-by: Takashi Iwai <tiwai@suse.de>
551 lines
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========================================================
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Guide to using M-Audio Audiophile USB with ALSA and Jack
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========================================================
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v1.5
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Thibault Le Meur <Thibault.LeMeur@supelec.fr>
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This document is a guide to using the M-Audio Audiophile USB (tm) device with
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ALSA and JACK.
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History
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=======
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* v1.4 - Thibault Le Meur (2007-07-11)
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- Added Low Endianness nature of 16bits-modes
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found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se>
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- Modifying document structure
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* v1.5 - Thibault Le Meur (2007-07-12)
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- Added AC3/DTS passthru info
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Audiophile USB Specs and correct usage
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======================================
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This part is a reminder of important facts about the functions and limitations
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of the device.
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The device has 4 audio interfaces, and 2 MIDI ports:
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* Analog Stereo Input (Ai)
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- This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA)
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- When the 1/4" TS (jack) connectors are connected, the RCA connectors
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are disabled
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* Analog Stereo Output (Ao)
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* Digital Stereo Input (Di)
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* Digital Stereo Output (Do)
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* Midi In (Mi)
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* Midi Out (Mo)
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The internal DAC/ADC has the following characteristics:
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* sample depth of 16 or 24 bits
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* sample rate from 8kHz to 96kHz
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* Two interfaces can't use different sample depths at the same time.
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Moreover, the Audiophile USB documentation gives the following Warning:
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Please exit any audio application running before switching between bit depths
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Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be
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activated at the same time depending on the audio mode selected:
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* 16-bit/48kHz ==> 4 channels in + 4 channels out
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- Ai+Ao+Di+Do
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* 24-bit/48kHz ==> 4 channels in + 2 channels out,
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or 2 channels in + 4 channels out
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- Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do
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* 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only)
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- Ai or Ao or Di or Do
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Important facts about the Digital interface:
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--------------------------------------------
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* The Do port additionally supports surround-encoded AC-3 and DTS passthrough,
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though I haven't tested it under Linux
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- Note that in this setup only the Do interface can be enabled
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* Apart from recording an audio digital stream, enabling the Di port is a way
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to synchronize the device to an external sample clock
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- As a consequence, the Di port must be enable only if an active Digital
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source is connected
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- Enabling Di when no digital source is connected can result in a
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synchronization error (for instance sound played at an odd sample rate)
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Audiophile USB MIDI support in ALSA
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===================================
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The Audiophile USB MIDI ports will be automatically supported once the
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following modules have been loaded:
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* snd-usb-audio
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* snd-seq-midi
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No additional setting is required.
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Audiophile USB Audio support in ALSA
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====================================
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Audio functions of the Audiophile USB device are handled by the snd-usb-audio
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module. This module can work in a default mode (without any device-specific
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parameter), or in an "advanced" mode with the device-specific parameter called
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``device_setup``.
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Default Alsa driver mode
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------------------------
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The default behavior of the snd-usb-audio driver is to list the device
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capabilities at startup and activate the required mode when required
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by the applications: for instance if the user is recording in a
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24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode,
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the snd-usb-audio module will reconfigure the device on the fly.
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This approach has the advantage to let the driver automatically switch from sample
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rates/depths automatically according to the user's needs. However, those who
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are using the device under windows know that this is not how the device is meant to
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work: under windows applications must be closed before using the m-audio control
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panel to switch the device working mode. Thus as we'll see in next section, this
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Default Alsa driver mode can lead to device misconfigurations.
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Let's get back to the Default Alsa driver mode for now. In this case the
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Audiophile interfaces are mapped to alsa pcm devices in the following
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way (I suppose the device's index is 1):
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* hw:1,0 is Ao in playback and Di in capture
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* hw:1,1 is Do in playback and Ai in capture
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* hw:1,2 is Do in AC3/DTS passthrough mode
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In this mode, the device uses Big Endian byte-encoding so that
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supported audio format are S16_BE for 16-bit depth modes and S24_3BE for
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24-bits depth mode.
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One exception is the hw:1,2 port which was reported to be Little Endian
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compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams.
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This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface
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is reported to be big endian in this default driver mode.
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Examples:
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* playing a S24_3BE encoded raw file to the Ao port::
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% aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw
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* recording a S24_3BE encoded raw file from the Ai port::
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% arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw
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* playing a S16_BE encoded raw file to the Do port::
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% aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw
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* playing an ac3 sample file to the Do port::
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% aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw
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If you're happy with the default Alsa driver mode and don't experience any
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issue with this mode, then you can skip the following chapter.
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Advanced module setup
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---------------------
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Due to the hardware constraints described above, the device initialization made
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by the Alsa driver in default mode may result in a corrupted state of the
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device. For instance, a particularly annoying issue is that the sound captured
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from the Ai interface sounds distorted (as if boosted with an excessive high
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volume gain).
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For people having this problem, the snd-usb-audio module has a new module
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parameter called ``device_setup`` (this parameter was introduced in kernel
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release 2.6.17)
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Initializing the working mode of the Audiophile USB
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~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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As far as the Audiophile USB device is concerned, this value let the user
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specify:
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* the sample depth
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* the sample rate
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* whether the Di port is used or not
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When initialized with ``device_setup=0x00``, the snd-usb-audio module has
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the same behaviour as when the parameter is omitted (see paragraph "Default
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Alsa driver mode" above)
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Others modes are described in the following subsections.
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16-bit modes
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~~~~~~~~~~~~
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The two supported modes are:
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* ``device_setup=0x01``
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- 16bits 48kHz mode with Di disabled
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- Ai,Ao,Do can be used at the same time
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- hw:1,0 is not available in capture mode
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- hw:1,2 is not available
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* ``device_setup=0x11``
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- 16bits 48kHz mode with Di enabled
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- Ai,Ao,Di,Do can be used at the same time
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- hw:1,0 is available in capture mode
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- hw:1,2 is not available
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In this modes the device operates only at 16bits-modes. Before kernel 2.6.23,
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the devices where reported to be Big-Endian when in fact they were Little-Endian
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so that playing a file was a matter of using:
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::
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% aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw
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where "test_S16_LE.raw" was in fact a little-endian sample file.
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Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in
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these modes) a fix has been committed (expected in kernel 2.6.23) and
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Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as
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using:
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::
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% aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw
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24-bit modes
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~~~~~~~~~~~~
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The three supported modes are:
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* ``device_setup=0x09``
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- 24bits 48kHz mode with Di disabled
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- Ai,Ao,Do can be used at the same time
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- hw:1,0 is not available in capture mode
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- hw:1,2 is not available
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* ``device_setup=0x19``
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- 24bits 48kHz mode with Di enabled
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- 3 ports from {Ai,Ao,Di,Do} can be used at the same time
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- hw:1,0 is available in capture mode and an active digital source must be
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connected to Di
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- hw:1,2 is not available
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* ``device_setup=0x0D`` or ``0x10``
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- 24bits 96kHz mode
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- Di is enabled by default for this mode but does not need to be connected
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to an active source
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- Only 1 port from {Ai,Ao,Di,Do} can be used at the same time
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- hw:1,0 is available in captured mode
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- hw:1,2 is not available
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In these modes the device is only Big-Endian compliant (see "Default Alsa driver
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mode" above for an aplay command example)
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AC3 w/ DTS passthru mode
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~~~~~~~~~~~~~~~~~~~~~~~~
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Thanks to Hakan Lennestal, I now have a report saying that this mode works.
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* ``device_setup=0x03``
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- 16bits 48kHz mode with only the Do port enabled
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- AC3 with DTS passthru
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- Caution with this setup the Do port is mapped to the pcm device hw:1,0
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The command line used to playback the AC3/DTS encoded .wav-files in this mode:
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::
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% aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw
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How to use the ``device_setup`` parameter
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~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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The parameter can be given:
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* By manually probing the device (as root):::
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# modprobe -r snd-usb-audio
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# modprobe snd-usb-audio index=1 device_setup=0x09
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* Or while configuring the modules options in your modules configuration file
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(typically a .conf file in /etc/modprobe.d/ directory:::
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alias snd-card-1 snd-usb-audio
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options snd-usb-audio index=1 device_setup=0x09
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CAUTION when initializing the device
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-------------------------------------
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* Correct initialization on the device requires that device_setup is given to
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the module BEFORE the device is turned on. So, if you use the "manual probing"
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method described above, take care to power-on the device AFTER this initialization.
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* Failing to respect this will lead to a misconfiguration of the device. In this case
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turn off the device, unprobe the snd-usb-audio module, then probe it again with
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correct device_setup parameter and then (and only then) turn on the device again.
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* If you've correctly initialized the device in a valid mode and then want to switch
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to another mode (possibly with another sample-depth), please use also the following
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procedure:
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- first turn off the device
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- de-register the snd-usb-audio module (modprobe -r)
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- change the device_setup parameter by changing the device_setup
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option in ``/etc/modprobe.d/*.conf``
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- turn on the device
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* A workaround for this last issue has been applied to kernel 2.6.23, but it may not
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be enough to ensure the 'stability' of the device initialization.
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Technical details for hackers
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-----------------------------
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This section is for hackers, wanting to understand details about the device
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internals and how Alsa supports it.
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Audiophile USB's ``device_setup`` structure
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~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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If you want to understand the device_setup magic numbers for the Audiophile
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USB, you need some very basic understanding of binary computation. However,
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this is not required to use the parameter and you may skip this section.
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The device_setup is one byte long and its structure is the following:
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::
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+---+---+---+---+---+---+---+---+
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| b7| b6| b5| b4| b3| b2| b1| b0|
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+---+---+---+---+---+---+---+---+
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| 0 | 0 | 0 | Di|24B|96K|DTS|SET|
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+---+---+---+---+---+---+---+---+
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Where:
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* b0 is the ``SET`` bit
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- it MUST be set if device_setup is initialized
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* b1 is the ``DTS`` bit
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- it is set only for Digital output with DTS/AC3
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- this setup is not tested
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* b2 is the Rate selection flag
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- When set to ``1`` the rate range is 48.1-96kHz
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- Otherwise the sample rate range is 8-48kHz
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* b3 is the bit depth selection flag
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- When set to ``1`` samples are 24bits long
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- Otherwise they are 16bits long
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- Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits
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samples
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* b4 is the Digital input flag
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- When set to ``1`` the device assumes that an active digital source is
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connected
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- You shouldn't enable Di if no source is seen on the port (this leads to
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synchronization issues)
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- b4 is implied by b2 (since only one port is enabled at a time no synch
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error can occur)
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* b5 to b7 are reserved for future uses, and must be set to ``0``
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- might become Ao, Do, Ai, for b7, b6, b4 respectively
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Caution:
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* there is no check on the value you will give to device_setup
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- for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since
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b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages
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* Hardware constraints due to the USB bus limitation aren't checked
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- choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
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only be able to use one at the same time
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USB implementation details for this device
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~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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You may safely skip this section if you're not interested in driver
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hacking.
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This section describes some internal aspects of the device and summarizes the
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data I got by usb-snooping the windows and Linux drivers.
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The M-Audio Audiophile USB has 7 USB Interfaces:
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a "USB interface":
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* USB Interface nb.0
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* USB Interface nb.1
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- Audio Control function
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* USB Interface nb.2
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- Analog Output
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* USB Interface nb.3
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- Digital Output
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* USB Interface nb.4
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- Analog Input
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* USB Interface nb.5
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- Digital Input
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* USB Interface nb.6
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- MIDI interface compliant with the MIDIMAN quirk
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Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
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* Interface 3 (Digital Out) has an extra Alset nb.6
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* Interface 5 (Digital In) does not have Alset nb.3 and 5
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Here is a short description of the AltSettings capabilities:
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* AltSettings 1 corresponds to
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- 24-bit depth, 48.1-96kHz sample mode
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- Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di)
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* AltSettings 2 corresponds to
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- 24-bit depth, 8-48kHz sample mode
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- Asynch capture and playback (Ao,Ai,Do,Di)
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* AltSettings 3 corresponds to
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- 24-bit depth, 8-48kHz sample mode
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- Synch capture (Ai) and Adaptive playback (Ao,Do)
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* AltSettings 4 corresponds to
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- 16-bit depth, 8-48kHz sample mode
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- Asynch capture and playback (Ao,Ai,Do,Di)
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* AltSettings 5 corresponds to
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- 16-bit depth, 8-48kHz sample mode
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- Synch capture (Ai) and Adaptive playback (Ao,Do)
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* AltSettings 6 corresponds to
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- 16-bit depth, 8-48kHz sample mode
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- Synch playback (Do), audio format type III IEC1937_AC-3
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In order to ensure a correct initialization of the device, the driver
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*must* *know* how the device will be used:
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* if DTS is chosen, only Interface 2 with AltSet nb.6 must be
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registered
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* if 96KHz only AltSets nb.1 of each interface must be selected
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* if samples are using 24bits/48KHz then AltSet 2 must me used if
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Digital input is connected, and only AltSet nb.3 if Digital input
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is not connected
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* if samples are using 16bits/48KHz then AltSet 4 must me used if
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Digital input is connected, and only AltSet nb.5 if Digital input
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is not connected
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When device_setup is given as a parameter to the snd-usb-audio module, the
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parse_audio_endpoints function uses a quirk called
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``audiophile_skip_setting_quirk`` in order to prevent AltSettings not
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corresponding to device_setup from being registered in the driver.
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Audiophile USB and Jack support
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===============================
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This section deals with support of the Audiophile USB device in Jack.
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There are 2 main potential issues when using Jackd with the device:
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* support for Big-Endian devices in 24-bit modes
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* support for 4-in / 4-out channels
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Direct support in Jackd
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-----------------------
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Jack supports big endian devices only in recent versions (thanks to
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Andreas Steinmetz for his first big-endian patch). I can't remember
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exactly when this support was released into jackd, let's just say that
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with jackd version 0.103.0 it's almost ok (just a small bug is affecting
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16bits Big-Endian devices, but since you've read carefully the above
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paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices
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are now Little Endians ;-) ).
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You can run jackd with the following command for playback with Ao and
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record with Ai:
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::
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% jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
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Using Alsa plughw
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-----------------
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If you don't have a recent Jackd installed, you can downgrade to using
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the Alsa ``plug`` converter.
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For instance here is one way to run Jack with 2 playback channels on Ao and 2
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capture channels from Ai:
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::
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% jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
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However you may see the following warning message:
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You appear to be using the ALSA software "plug" layer, probably a result of
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using the "default" ALSA device. This is less efficient than it could be.
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Consider using a hardware device instead rather than using the plug layer.
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Getting 2 input and/or output interfaces in Jack
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------------------------------------------------
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As you can see, starting the Jack server this way will only enable 1 stereo
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input (Di or Ai) and 1 stereo output (Ao or Do).
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This is due to the following restrictions:
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* Jack can only open one capture device and one playback device at a time
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* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1
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(and optionally hw:1,2)
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If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to
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combine the Alsa devices into one logical "complex" device.
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If you want to give it a try, I recommend reading the information from
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this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html
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It is related to another device (ice1712) but can be adapted to suit
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the Audiophile USB.
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Enabling multiple Audiophile USB interfaces for Jackd will certainly require:
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* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page)
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* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
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* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc
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file
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* start jackd with this device
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I had no success in testing this for now, if you have any success with this kind
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of setup, please drop me an email.
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