| ======================================================== |
| 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 |
| ALSA and JACK. |
| |
| History |
| ======= |
| |
| * v1.4 - Thibault Le Meur (2007-07-11) |
| |
| - Added Low Endianness nature of 16bits-modes |
| found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se> |
| - Modifying document structure |
| |
| * v1.5 - Thibault Le Meur (2007-07-12) |
| - Added AC3/DTS passthru info |
| |
| |
| 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) |
| |
| |
| 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. |
| |
| |
| Audiophile USB Audio support in ALSA |
| ==================================== |
| |
| Audio functions of the Audiophile USB device are handled by the snd-usb-audio |
| module. This module can work in a default mode (without any device-specific |
| parameter), or in an "advanced" mode with the device-specific parameter called |
| ``device_setup``. |
| |
| 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. |
| |
| 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 |
| |
| * 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. |
| |
| 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) |
| |
| Initializing the working mode of the Audiophile USB |
| ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ |
| |
| As far as the Audiophile USB device is concerned, this value let the user |
| specify: |
| |
| * 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 |
| using: |
| :: |
| |
| % 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 |
| |
| 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 |
| (typically 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. |
| |
| 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| |
| +---+---+---+---+---+---+---+---+ |
| |
| 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 |
| |
| USB implementation details for this device |
| ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ |
| |
| You may safely skip this section if you're not interested in driver |
| hacking. |
| |
| 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. |
| |
| 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 |
| |
| 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 |
| |
| 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. |
| |
| 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. |
