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[ALSA] Fixes audiophile usb analog capture with the new device_setup parameter

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Modules: Documentation,USB generic driver

The patch adds the 'device_setup' module parameter and a specific
quirk to correctly initialize the audiophile usb device: this fixes
the distorted sound bug on the Analog capture port. Backward
compatibility is achieved by simply omitting the new parameter.

Signed-off-by: Thibault LE MEUR <Thibault.LeMeur@supelec.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is contained in:
Thibault LE MEUR 2006-03-14 11:44:53 +01:00 committed by Jaroslav Kysela
parent ecefb19252
commit e311334ee6
3 changed files with 384 additions and 1 deletions
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3
Documentation/sound/alsa/ALSA-Configuration.txt
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@ -1411,6 +1411,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
vid - Vendor ID for the device (optional)
pid - Product ID for the device (optional)
device_setup - Device specific magic number (optional)
- Influence depends on the device
- Default: 0x0000
This module supports multiple devices, autoprobe and hotplugging.

330
Documentation/sound/alsa/Audiophile-Usb.txt Normal file
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@ -0,0 +1,330 @@
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

52
sound/usb/usbaudio.c
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@ -70,6 +70,7 @@ static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; /* Vendor ID for
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.");
@ -85,6 +86,8 @@ module_param(nrpacks, int, 0644);
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).");
/*
@ -2547,6 +2550,8 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp
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;
@ -2581,6 +2586,12 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
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);
@ -2675,7 +2686,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
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);
@ -3083,6 +3094,45 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev)
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