361 строка
14 KiB
Plaintext
361 строка
14 KiB
Plaintext
Guide to using M-Audio Audiophile USB with ALSA and Jack v1.3
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========================================================
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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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1 - 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 ports can't use different sample depths at the same time. Moreover, the
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Audiophile USB documentation gives the following Warning: "Please exit any
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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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2 - Audiophile USB support in ALSA
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==================================
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2.1 - MIDI ports
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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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2.2 - Audio ports
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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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2.2.1 - Default Alsa driver mode
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The default behavior of the snd-usb-audio driver is to parse the device
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capabilities at startup and enable all functions inside the device (including
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all ports at any supported sample rates and sample depths). This approach
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has the advantage to let the driver easily switch from sample rates/depths
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automatically according to the need of the application claiming the device.
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In this case the Audiophile ports are mapped to alsa pcm devices in the
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following 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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You must note as well that 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. One exception is the hw:1,2 port which is Little Endian
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compliant and thus uses S16_LE.
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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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If you're happy with the default Alsa driver setup and don't experience any
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issue with this mode, then you can skip the following chapter.
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2.2.2 - Advanced module setup
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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 port sounds distorted (as if boosted with an excessive high volume
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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".
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2.2.2.1 - Initializing the working mode of the Audiophile USB
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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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Here is a list of supported device_setup values for this device:
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* device_setup=0x00 (or omitted)
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- Alsa driver default mode
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- maintains backward compatibility with setups that do not use this
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parameter by not introducing any change
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- results sometimes in corrupted sound as described earlier
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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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* 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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* device_setup=0x03
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- 16bits 48kHz mode with only the Do port enabled
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- AC3 with DTS passthru (not tested)
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- Caution with this setup the Do port is mapped to the pcm device hw:1,0
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2.2.2.2 - Setting and switching configurations with the device_setup parameter
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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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- For Fedora distributions, edit the /etc/modprobe.conf file:
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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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IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
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-------------------------------------------
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* You may need to _first_ initialize the module with the correct device_setup
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parameter and _only_after_ turn on the Audiophile USB device
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* This is especially true when switching the sample depth:
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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.conf
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- turn on the device
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2.2.2.3 - Audiophile USB's device_setup structure
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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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| 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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2.2.3 - USB implementation details for this device
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You may safely skip this section if you're not interested in driver
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development.
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This section describes some internal aspects of the device and summarize 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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3 - 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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The main issue regarding this support is that the device is Big Endian
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compliant.
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3.1 - Using the plug alsa plugin
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--------------------------------
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Jack doesn't directly support big endian devices. Thus, one way to have support
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for this device with Alsa is to use 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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% 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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3.2 - Patching alsa to use direct pcm device
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--------------------------------------------
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A patch for Jack by Andreas Steinmetz adds support for Big Endian devices.
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However it has not been included in the CVS tree.
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You can find it at the following URL:
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http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687&
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atid=425939
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After having applied the patch you can run jackd with the following command
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line:
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% jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
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3.2 - 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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* patching Jack with the previously mentioned "Big Endian" patch
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* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page)
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* 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, but this may be due to my OS
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configuration. If you have any success with this kind of setup, please
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drop me an email.
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