* [PATCH 2/2] M-Audio USB Documentation
@ 2006-11-24 20:26 Pavel Polischouk
2006-11-27 11:17 ` Takashi Iwai
0 siblings, 1 reply; 4+ messages in thread
From: Pavel Polischouk @ 2006-11-24 20:26 UTC (permalink / raw)
To: alsa-devel
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Re-send, "hg export" format
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# HG changeset patch
# User Pavel Polischouk <pavelvp@inbox.ru>
# Date 1164387029 18000
# Node ID 6f81f7397f82b392e40c582e48a02a7c1cbd7c3e
# Parent 9a4c87387be903c78c07a018f30457370c7e54a3
Documentation update
diff -r 9a4c87387be9 -r 6f81f7397f82 Documentation/M-Audio-Usb.txt
--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/Documentation/M-Audio-Usb.txt Fri Nov 24 11:50:29 2006 -0500
@@ -0,0 +1,461 @@
+ Guide to using M-Audio USB devices with ALSA and Jack v1.4
+ =====================================================
+
+ Thibault Le Meur <Thibault.LeMeur@supelec.fr>
+ Pavel Polischouk <pavelvp@inbox.ru>
+
+This document is a guide to using the M-Audio Audiophile USB (tm) device with
+ALSA and JACK.
+
+1 - M-Audio USB Specs and correct usage
+==========================================
+This part is a reminder of important facts about the functions and limitations
+of the device.
+
+M-Audio Quattro has 4 audio interfaces and 2 MIDI ports:
+ * 2 Analog Stereo Inputs (Ai1, Ai2, both: Ai),
+ * 2 Analog Stereo Outputs (Ao1, Ao2, both: Ao)
+ * Midi In (Mi)
+ * Midi Out (Mo)
+
+M-Audio Audiophile USB 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 of all M-Audio USB devices has the following characteristics:
+* sample depth of 16 or 24 bits
+* sample rate from 8kHz to 96kHz
+* Two ports can't use different sample depths at the same time. Moreover, the
+M-Audio 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 for Quattro
+ - Ai+Ao+Di+Do for Audiophile
+ * 24-bit/48kHz ==> 4 channels in/2 channels out,
+ or 2 channels in/4 channels out
+ - Ai+Ao1, Ai+Ao2, Ai1+Ao, Ai2+Ao
+ - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do for Audiophile
+ * 24-bit/96kHz ==> 2 channels in, or 2 channels out (half duplex only)
+ - Ai1 or Ai2 or Ao1 or Ao2 for Quattro
+ - Ai or Ao or Di or Do for Audiophile
+
+Important facts about the Audiophile 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 - M-Audio USB support in ALSA
+==================================
+
+2.1 - MIDI ports
+----------------
+The M-Audio (formerly Midiman) 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.
+
+2.2 - Audio ports
+-----------------
+
+Audio functions of the M-Audio USB devices 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".
+
+2.2.1 - Default Alsa driver mode
+
+The default behavior of the snd-usb-audio driver is to parse the device
+capabilities at startup and enable all functions inside the device (including
+all ports at any supported sample rates and sample depths). This approach
+has the advantage to let the driver easily switch from sample rates/depths
+automatically according to the need of the application claiming the device.
+
+In this case the Audiophile ports 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
+Quattro device mapping:
+ * hw:1,0 is Ao1 and Ai1
+ * hw:1,1 is Ao2 and Ai2
+
+You must note as well that 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. An exception for Audiophile is the hw:1,2 port which is
+Little Endian compliant and thus uses S16_LE.
+
+Examples:
+ * playing a S24_3BE encoded raw file to the Ao port of Audiophile
+ % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw
+ * recording a S24_3BE encoded raw file from the Ai port of Audiophile
+ % arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw
+ * playing a S16_BE encoded raw file to the Do port of Audiophile
+ % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw
+
+ * playing S24_3LE encoded raw file to the Ao1 port of Quattro
+ % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw
+ * recording a S24_3BE encoded raw file from the Ai2 port of Quattro
+ % arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw
+
+If you're happy with the default Alsa driver setup and don't experience any
+issue with this mode, then you can skip the following chapter. It is very
+unlikely that this setup will give useful results for Quattro though.
+
+2.2.2 - 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 port 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".
+
+2.2.2.1 - Initializing the working mode of the M-Audio USB devices
+
+As far as the M-Audio USB devices are concerned, this value let the user
+specify:
+ * the sample depth
+ * the sample rate
+ * whether the Di port is used or not (except Quattro)
+
+A list of supported device_setup values for Audiophile USB:
+ * device_setup=0x00 (or omitted)
+ - Alsa driver default mode
+ - maintains backward compatibility with setups that do not use this
+ parameter by not introducing any change
+ - results sometimes in corrupted sound as described earlier
+ * 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
+ * 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
+ * device_setup=0x03
+ - 16bits 48kHz mode with only the Do port enabled
+ - AC3 with DTS passthru (not tested)
+ - Caution with this setup the Do port is mapped to the pcm device hw:1,0
+
+A list of supported device_setup values for Quattro USB:
+ * device_setup=0x00 (or omitted)
+ - Alsa driver default mode
+ - maintains backward compatibility with setups that do not use this
+ parameter by not introducing any change
+ - results sometimes in corrupted sound as decribed earlier
+ * device_setup=0x01
+ - 16bits 48kHz
+ - Ai,Ao can be used at the same time
+ * device_setup=0x09
+ - 24bits 48kHz mode
+ - 3 ports from {Ai1,Ao1,Ai2,Ao2} can be used at the same time
+ * device_setup=0x05
+ - 24bits 96kHz mode
+ - Only 1 port from {Ai1,Ao1,Ai2,Ao2} can be used at the same time
+ * device_setup=0x81
+ - Windows "usbaudio.sys" compatibility mode
+ - 16bits up to 44.1kHz
+ - Ai1,Ao1 can be used at the same time
+ - Ai2,Ao2 not available
+
+2.2.2.2 - Setting and switching configurations with 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
+ - For Fedora distributions, edit the /etc/modprobe.conf file:
+ alias snd-card-1 snd-usb-audio
+ options snd-usb-audio index=1 device_setup=0x09
+ - udev will automatically load the driver when the device is connected.
+
+IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
+-------------------------------------------
+To switch the sample rate setting between 8-48 and 88-96, or to switch between
+bit sample width configurations (16->24) you MUST unload the driver with
+`rmmod' and re-load it with new parameters as outlined above. You don't need
+to turn off the device, but in some cases when the device seems to get "stuck"
+with previous configuration and refuses to switch to the new one the following
+sequence may be used:
+ * You may need to _first_ initialize the module with the correct device_setup
+ parameter and _only_after_ turn on the Audiophile USB device
+ * This is especially true when switching the sample depth:
+ - 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.conf
+ - turn on the device
+
+2.2.2.3 - M-Audio USB's device_setup structure
+
+If you want to understand the device_setup magic numbers for the M-Audio
+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|
+ +---+---+---+---+---+---+---+---+
+ |CMP| 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 b6 are reserved for future uses, and must be set to "0"
+ * b7 is the Windows "usbaudio.sys" compatibility mode. Generally not useful
+ because it's just a sub-set of 16/48 configuration, limited to 44.1 kHz and
+ one stereo device. Kept mostly for debugging purpose.
+
+Caution:
+ * there is no check on the value you will give to device_setup
+ - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x0d 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
+
+2.2.3 - USB implementation details for this device
+
+You may safely skip this section if you're not interested in driver
+development.
+
+This section describes some internal aspects of the device and summarize the
+data I got by usb-snooping the windows and Linux drivers.
+
+2.2.3.1. M-Audio Audiophile
+
+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.
+
+2.2.3.1. M-Audio Quattro
+
+The M-Audio Quattro USB has 10 USB Interfaces:
+a "USB interface":
+ * USB Interface nb.0
+ - Audio Control function
+ * USB Interface nb.1
+ - Analog Output, usbaudio.sys compatible
+ * USB Interface nb.2
+ - Analog Input, usbaudio.sys compatible
+ * USB Interface nb.3
+ - Audio Control function
+ * USB Interface nb.4
+ - Analog Output #1
+ * USB Interface nb.5
+ - Analog Input #1
+ * USB Interface nb.6
+ - Audio Control function
+ * USB Interface nb.7
+ - Analog Output #2
+ * USB Interface nb.8
+ - Analog Input #2
+ * USB Interface nb.9
+ - MIDI interface compliant with the MIDIMAN quirk
+
+Each interface has 3 altsettings (AltSet 1,3,4) except interfaces 1 and 2 (usbaudio.sys compatible) that have only one Alset nb.4
+
+Here is a short description of the AltSettings capabilities:
+ * AltSettings 1 corresponds to
+ - 24-bit depth, 88.2-96kHz sample mode
+ - Adaptive playback (Ao), Synch capture (Ai)
+ * AltSettings 3 corresponds to
+ - 24-bit depth, 8-48kHz sample mode
+ - Adaptive playback (Ao), Synch capture (Ai)
+ * AltSettings 4 corresponds to
+ - 16-bit depth, 8-48kHz sample mode
+ - Adaptive playback (Ao), Synch capture (Ai)
+
+In order to ensure a correct initialization of the device, the driver
+_must_know_ how the device will be used:
+ * if 96KHz only AltSets nb.1 of each interface must be selected
+ * if samples are using 24bits/48KHz then AltSet nb.3 must be used
+ * if samples are using 16bits/48KHz then AltSet 4 must be used.
+
+
+When device_setup is given as a parameter to the snd-usb-audio module, the
+parse_audio_endpoints function uses a quirk called
+"quattro_skip_setting_quirk" in order to prevent AltSettings not
+corresponding to device_setup from being registered in the driver.
+
+3 - M-Audio USB and Jack support
+===================================
+
+This section deals with support of the Audiophile USB device in Jack.
+The main issue regarding this support is that the device is Big Endian
+compliant.
+
+3.1 - Using the plug alsa plugin
+--------------------------------
+
+Jack doesn't directly support big endian devices. Thus, one way to have support
+for this device with Alsa is to use 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."
+
+3.2 - Patching alsa to use direct pcm device
+--------------------------------------------
+A patch for Jack by Andreas Steinmetz adds support for Big Endian devices.
+However it has not been included in the CVS tree.
+
+You can find it at the following URL:
+http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687&
+atid=425939
+
+After having applied the patch you can run jackd with the following command
+line:
+ % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
+
+3.2 - 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 M-Audio USB interfaces for Jackd will certainly require:
+* patching Jack with the previously mentioned "Big Endian" patch
+* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page)
+* 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, but this may be due to my OS
+configuration. If you have any success with this kind of setup, please
+drop me an email.
diff -r 9a4c87387be9 -r 6f81f7397f82 Documentation/Audiophile-Usb.txt
--- a/Documentation/Audiophile-Usb.txt Fri Nov 24 15:49:39 2006 +0100
+++ /dev/null Thu Jan 01 00:00:00 1970 +0000
@@ -1,360 +0,0 @@
- Guide to using M-Audio Audiophile USB with ALSA and Jack v1.3
- ========================================================
-
- 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.
-
-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 ports 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 support in ALSA
-==================================
-
-2.1 - MIDI ports
-----------------
-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.
-
-2.2 - Audio ports
------------------
-
-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".
-
-2.2.1 - Default Alsa driver mode
-
-The default behavior of the snd-usb-audio driver is to parse the device
-capabilities at startup and enable all functions inside the device (including
-all ports at any supported sample rates and sample depths). This approach
-has the advantage to let the driver easily switch from sample rates/depths
-automatically according to the need of the application claiming the device.
-
-In this case the Audiophile ports 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
-
-You must note as well that 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 is Little Endian
-compliant and thus uses S16_LE.
-
-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
-
-If you're happy with the default Alsa driver setup and don't experience any
-issue with this mode, then you can skip the following chapter.
-
-2.2.2 - 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 port 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".
-
-2.2.2.1 - 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
-
-Here is a list of supported device_setup values for this device:
- * device_setup=0x00 (or omitted)
- - Alsa driver default mode
- - maintains backward compatibility with setups that do not use this
- parameter by not introducing any change
- - results sometimes in corrupted sound as described earlier
- * 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
- * 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
- * device_setup=0x03
- - 16bits 48kHz mode with only the Do port enabled
- - AC3 with DTS passthru (not tested)
- - Caution with this setup the Do port is mapped to the pcm device hw:1,0
-
-2.2.2.2 - Setting and switching configurations with 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
- - For Fedora distributions, edit the /etc/modprobe.conf file:
- alias snd-card-1 snd-usb-audio
- options snd-usb-audio index=1 device_setup=0x09
-
-IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
--------------------------------------------
- * You may need to _first_ initialize the module with the correct device_setup
- parameter and _only_after_ turn on the Audiophile USB device
- * This is especially true when switching the sample depth:
- - 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.conf
- - turn on the device
-
-2.2.2.3 - 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
-
-2.2.3 - USB implementation details for this device
-
-You may safely skip this section if you're not interested in driver
-development.
-
-This section describes some internal aspects of the device and summarize 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.
-
-3 - Audiophile USB and Jack support
-===================================
-
-This section deals with support of the Audiophile USB device in Jack.
-The main issue regarding this support is that the device is Big Endian
-compliant.
-
-3.1 - Using the plug alsa plugin
---------------------------------
-
-Jack doesn't directly support big endian devices. Thus, one way to have support
-for this device with Alsa is to use 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."
-
-3.2 - Patching alsa to use direct pcm device
---------------------------------------------
-A patch for Jack by Andreas Steinmetz adds support for Big Endian devices.
-However it has not been included in the CVS tree.
-
-You can find it at the following URL:
-http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687&
-atid=425939
-
-After having applied the patch you can run jackd with the following command
-line:
- % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
-
-3.2 - 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:
-* patching Jack with the previously mentioned "Big Endian" patch
-* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page)
-* 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, but this may be due to my OS
-configuration. If you have any success with this kind of setup, please
-drop me an email.
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^ permalink raw reply [flat|nested] 4+ messages in thread
* Re: [PATCH 2/2] M-Audio USB Documentation
2006-11-24 20:26 [PATCH 2/2] M-Audio USB Documentation Pavel Polischouk
@ 2006-11-27 11:17 ` Takashi Iwai
2006-11-27 11:47 ` RE : " Thibault Le Meur
0 siblings, 1 reply; 4+ messages in thread
From: Takashi Iwai @ 2006-11-27 11:17 UTC (permalink / raw)
To: Pavel Polischouk; +Cc: alsa-devel
At Fri, 24 Nov 2006 15:26:46 -0500,
Pavel Polischouk wrote:
>
> +3.2 - Patching alsa to use direct pcm device
> +--------------------------------------------
> +A patch for Jack by Andreas Steinmetz adds support for Big Endian devices.
> +However it has not been included in the CVS tree.
Is this true? If so, you should submit the patch to upstream...
Takashi
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^ permalink raw reply [flat|nested] 4+ messages in thread
* RE : [PATCH 2/2] M-Audio USB Documentation
2006-11-27 11:17 ` Takashi Iwai
@ 2006-11-27 11:47 ` Thibault Le Meur
2006-11-27 15:20 ` Pavel Polischouk
0 siblings, 1 reply; 4+ messages in thread
From: Thibault Le Meur @ 2006-11-27 11:47 UTC (permalink / raw)
To: 'Takashi Iwai', 'Pavel Polischouk'; +Cc: alsa-devel
> At Fri, 24 Nov 2006 15:26:46 -0500,
> Pavel Polischouk wrote:
> >
> > +3.2 - Patching alsa to use direct pcm device
> > +--------------------------------------------
> > +A patch for Jack by Andreas Steinmetz adds support for Big Endian
> > +devices.
> > +However it has not been included in the CVS tree.
>
> Is this true? If so, you should submit the patch to upstream...
Oups, my mistake:
* It is "Patching Jackd to use direct pcm device" and NOT "Patching alsa to
use direct pcm device"
* Moreover, this patch has now been adapted and integrated into the Jackd
main code (I can't remember which version had this patch, but the latest one
has it for sure)
And sorry I hadn't time to test the new M-Audio USB patch. I'll try to do
this for next week.
Regards,
Thibault
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^ permalink raw reply [flat|nested] 4+ messages in thread
* Re: [PATCH 2/2] M-Audio USB Documentation
2006-11-27 11:47 ` RE : " Thibault Le Meur
@ 2006-11-27 15:20 ` Pavel Polischouk
0 siblings, 0 replies; 4+ messages in thread
From: Pavel Polischouk @ 2006-11-27 15:20 UTC (permalink / raw)
To: alsa-devel
Thibault Le Meur wrote:
>> At Fri, 24 Nov 2006 15:26:46 -0500,
>> Pavel Polischouk wrote:
>>
>>> +3.2 - Patching alsa to use direct pcm device
>>> +--------------------------------------------
>>> +A patch for Jack by Andreas Steinmetz adds support for Big Endian
>>> +devices.
>>> +However it has not been included in the CVS tree.
>>>
>> Is this true? If so, you should submit the patch to upstream...
>>
>
> Oups, my mistake:
> * It is "Patching Jackd to use direct pcm device" and NOT "Patching alsa to
> use direct pcm device"
> * Moreover, this patch has now been adapted and integrated into the Jackd
> main code (I can't remember which version had this patch, but the latest one
> has it for sure)
>
This part of the documentation came from the original
Audiophile-Usb.txt, I didn't modify this part. Looks like it's outdated,
jack 0.102 already has big endian support integrated (sort of - it has
certain problems, like not supporting different endians for playback and
capture, I'm gonna submit the patch for this to Jack team). I can update
this part of documentation too.
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^ permalink raw reply [flat|nested] 4+ messages in thread
end of thread, other threads:[~2006-11-27 15:20 UTC | newest]
Thread overview: 4+ messages (download: mbox.gz follow: Atom feed
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2006-11-24 20:26 [PATCH 2/2] M-Audio USB Documentation Pavel Polischouk
2006-11-27 11:17 ` Takashi Iwai
2006-11-27 11:47 ` RE : " Thibault Le Meur
2006-11-27 15:20 ` Pavel Polischouk
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