--- /dev/null
+ Guide to using M-Audio Audiophile USB with ALSA and Jack v1.1
+ ========================================================
+
+ 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)
+ * 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 caracteristics:
+* 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 additionnaly 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 syncrhonize 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
+ * snd-seq-midi
+
+No additionnal 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 behaviour 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 sample rates and any sample depths supported). 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 Audiohile USB
+
+As far as the Audiohile 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 decribed 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_ intialize 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 trun off the device
+ - de-register the snd-usb-audio module
+ - change the device_setup parameter (by either manually reprobing the module
+ or changing modprobe.conf)
+ - turn on the device
+
+2.2.2.3 - Setting and switching configurations with the device_setup parameter
+
+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 thi 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 - Technical Details for Audiophile Usb
+
+You may safely skip this section if you're not interrested in driver
+development.
+
+This section describes some internals aspect 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 intialization of the device, the driver
+_must_know_ how the device will be used:
+ * if DTS is choosen, 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_enpoint 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:
+# /usr/local/bin/jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
+
static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; /* Product ID for this card */
static int nrpacks = 4; /* max. number of packets per urb */
static int async_unlink = 1;
+static int device_setup[SNDRV_CARDS]; /* device parameter for this card*/
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for the USB audio adapter.");
MODULE_PARM_DESC(nrpacks, "Max. number of packets per URB.");
module_param(async_unlink, bool, 0444);
MODULE_PARM_DESC(async_unlink, "Use async unlink mode.");
+module_param_array(device_setup, int, NULL, 0444);
+MODULE_PARM_DESC(device_setup, "Specific device setup (if needed).");
/*
return 0;
}
+static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
+ int iface, int altno);
static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
{
struct usb_device *dev;
stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ?
SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
altno = altsd->bAlternateSetting;
+
+ /* audiophile usb: skip altsets incompatible with device_setup
+ */
+ if (chip->usb_id == USB_ID(0x0763, 0x2003) &&
+ audiophile_skip_setting_quirk(chip, iface_no, altno))
+ continue;
/* get audio formats */
fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL);
continue;
}
- snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, i, fp->endpoint);
+ snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, altno, fp->endpoint);
err = add_audio_endpoint(chip, stream, fp);
if (err < 0) {
kfree(fp->rate_table);
return 0;
}
+/*
+ * Setup quirks
+ */
+#define AUDIOPHILE_SET 0x01 /* if set, parse device_setup */
+#define AUDIOPHILE_SET_DTS 0x02 /* if set, enable DTS Digital Output */
+#define AUDIOPHILE_SET_96K 0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */
+#define AUDIOPHILE_SET_24B 0x08 /* 24bits sample if set, 16bits otherwise */
+#define AUDIOPHILE_SET_DI 0x10 /* if set, enable Digital Input */
+#define AUDIOPHILE_SET_MASK 0x1F /* bit mask for setup value */
+#define AUDIOPHILE_SET_24B_48K_DI 0x19 /* value for 24bits+48KHz+Digital Input */
+#define AUDIOPHILE_SET_24B_48K_NOTDI 0x09 /* value for 24bits+48KHz+No Digital Input */
+#define AUDIOPHILE_SET_16B_48K_DI 0x11 /* value for 16bits+48KHz+Digital Input */
+#define AUDIOPHILE_SET_16B_48K_NOTDI 0x01 /* value for 16bits+48KHz+No Digital Input */
+
+static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
+ int iface, int altno)
+{
+ if (device_setup[chip->index] & AUDIOPHILE_SET) {
+ if ((device_setup[chip->index] & AUDIOPHILE_SET_DTS)
+ && altno != 6)
+ return 1; /* skip this altsetting */
+ if ((device_setup[chip->index] & AUDIOPHILE_SET_96K)
+ && altno != 1)
+ return 1; /* skip this altsetting */
+ if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
+ AUDIOPHILE_SET_24B_48K_DI && altno != 2)
+ return 1; /* skip this altsetting */
+ if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
+ AUDIOPHILE_SET_24B_48K_NOTDI && altno != 3)
+ return 1; /* skip this altsetting */
+ if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
+ AUDIOPHILE_SET_16B_48K_DI && altno != 4)
+ return 1; /* skip this altsetting */
+ if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
+ AUDIOPHILE_SET_16B_48K_NOTDI && altno != 5)
+ return 1; /* skip this altsetting */
+ }
+ return 0; /* keep this altsetting */
+}
/*
* audio-interface quirks
git.openpandora.org / pandora-kernel.git / commitdiff