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diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt new file mode 100644 index 000000000..e7a5ed4dc --- /dev/null +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -0,0 +1,442 @@ + 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 + + +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 +"device_setup". + +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. + +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. + +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 +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. + +3.2.1.1 - 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 + +3.2.1.2 - 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) + +3.2.1.3 - 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 + (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. + +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. + +3.2.3.1 - 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 + +3.2.3.2 - 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. + +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. |