forked from Minki/linux
[ALSA] Fix typos and add information about Jack support to Audiophile-Usb.txt
Signed-off-by: Charis Kouzinopoulos <kouzinopoulos@gmail.com> Signed-off-by: Thibault Le Meur <Thibault.LeMeur@supelec.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Charis Kouzinopoulos
Jaroslav Kysela
parent
c128b82cf4
commit
5732e7a2ce
@ -1,4 +1,4 @@
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Guide to using M-Audio Audiophile USB with ALSA and Jack v1.2
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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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@ -22,16 +22,16 @@ The device has 4 audio interfaces, and 2 MIDI ports:
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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 caracteristics:
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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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* 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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* 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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@ -41,8 +41,8 @@ activated at the same time depending on the audio mode selected:
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Important facts about the Digital interface:
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--------------------------------------------
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* The Do port additionnaly supports surround-encoded AC-3 and DTS passthrough,
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though I haven't tested it under linux
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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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@ -60,24 +60,23 @@ synchronization error (for instance sound played at an odd sample rate)
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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
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* snd-seq-midi
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No additionnal setting is required.
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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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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 behaviour of the snd-usb-audio driver is to parse the device
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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 sample rates and any sample depths supported). This approach
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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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@ -114,9 +113,9 @@ 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 Audiohile USB
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2.2.2.1 - Initializing the working mode of the Audiophile USB
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As far as the Audiohile USB device is concerned, this value let the user
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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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@ -174,20 +173,20 @@ The parameter can be given:
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IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
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-------------------------------------------
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* You may need to _first_ intialize the module with the correct device_setup
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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 trun off the device
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- de-register the snd-usb-audio module
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- change the device_setup parameter (by either manually reprobing the module
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or changing modprobe.conf)
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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 thi section.
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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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@ -231,11 +230,11 @@ Caution:
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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 interrested in driver
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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 internals aspect 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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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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@ -277,9 +276,9 @@ Here is a short description of the AltSettings capabilities:
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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 intialization of the device, the driver
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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 choosen, only Interface 2 with AltSet nb.6 must be
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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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@ -290,7 +289,7 @@ _must_know_ how the device will be used:
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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_enpoint function uses a quirk called
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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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@ -317,9 +316,8 @@ However you may see the following warning message:
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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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--------------------------------------------
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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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@ -331,3 +329,32 @@ 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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