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authorLinus Torvalds <torvalds@g5.osdl.org>2006-05-01 07:46:46 -0700
committerLinus Torvalds <torvalds@g5.osdl.org>2006-05-01 07:46:46 -0700
commit494b9aea6d451e1eaab5d52b65951d7dc6e81cb8 (patch)
treeea70b0d3934a3a7f468d285833029798be24d5e1 /Documentation
parentMerge branch 'merge' of git://git.kernel.org/pub/scm/linux/kernel/git/paulus/powerpc (diff)
parent[ALSA] via82xx - Use DXS_SRC as default for VIA8235/8237/8251 chips (diff)
downloadlinux-dev-494b9aea6d451e1eaab5d52b65951d7dc6e81cb8.tar.xz
linux-dev-494b9aea6d451e1eaab5d52b65951d7dc6e81cb8.zip
Merge git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa
* git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa: (22 commits) [ALSA] via82xx - Use DXS_SRC as default for VIA8235/8237/8251 chips [ALSA] hda-codec - Add model entry for ASUS Z62F [ALSA] PCMCIA sound devices shouldn't depend on ISA [ALSA] hda-codec - Fix capture from line-in on VAIO SZ/FE laptops [ALSA] Fix Oops at rmmod with CONFIG_SND_VERBOSE_PROCFS=n [ALSA] PCM core - introduce CONFIG_SND_PCM_XRUN_DEBUG [ALSA] adding __devinitdata to pci_device_id [ALSA] add __devinitdata to all pci_device_id [ALSA] hda-codec - Add codec id for AD1988B codec chip [ALSA] hda-codec - Add model entry for ASUS M9 laptop [ALSA] pcxhr - Fix a compiler warning on 64bit architectures [ALSA] via82xx: tweak VT8251 workaround [ALSA] intel8x0 - Disable ALI5455 SPDIF-input [ALSA] via82xx: add support for VIA VT8251 (AC'97) [ALSA] Fix typos and add information about Jack support to Audiophile-Usb.txt [ALSA] Fix double free in error path of miro driver [ALSA] hda-codec - Add entry for Epox EP-5LDA+ GLi [ALSA] sound/pci/: remove duplicate #include's [ALSA] hda-codec - Use model 'hp' for all HP laptops with AD1981HD [ALSA] continue on IS_ERR from platform device registration ...
Diffstat (limited to 'Documentation')
-rw-r--r--Documentation/sound/alsa/Audiophile-Usb.txt81
-rw-r--r--Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl4
2 files changed, 56 insertions, 29 deletions
diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
index 4692c8e77dc1..b535c2a198f8 100644
--- a/Documentation/sound/alsa/Audiophile-Usb.txt
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -1,4 +1,4 @@
- Guide to using M-Audio Audiophile USB with ALSA and Jack v1.2
+ Guide to using M-Audio Audiophile USB with ALSA and Jack v1.3
========================================================
Thibault Le Meur <Thibault.LeMeur@supelec.fr>
@@ -22,16 +22,16 @@ The device has 4 audio interfaces, and 2 MIDI ports:
* Midi In (Mi)
* Midi Out (Mo)
-The internal DAC/ADC has the following caracteristics:
+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
+* 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
+ * 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
@@ -41,8 +41,8 @@ activated at the same time depending on the audio mode selected:
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
+ * 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
@@ -60,24 +60,23 @@ synchronization error (for instance sound played at an odd sample rate)
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.
+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
+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
+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 sample rates and any sample depths supported). This approach
+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.
@@ -114,9 +113,9 @@ 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
+2.2.2.1 - Initializing the working mode of the Audiophile USB
-As far as the Audiohile USB device is concerned, this value let the user
+As far as the Audiophile USB device is concerned, this value let the user
specify:
* the sample depth
* the sample rate
@@ -174,20 +173,20 @@ The parameter can be given:
IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
-------------------------------------------
- * You may need to _first_ intialize the module with the correct device_setup
+ * 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 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)
+ - 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 thi section.
+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:
@@ -231,11 +230,11 @@ Caution:
2.2.3 - USB implementation details for this device
-You may safely skip this section if you're not interrested in driver
+You may safely skip this section if you're not interested 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.
+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":
@@ -277,9 +276,9 @@ Here is a short description of the AltSettings capabilities:
- 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
+In order to ensure a correct initialization 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
+ * 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
@@ -290,7 +289,7 @@ _must_know_ how the device will be used:
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
+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.
@@ -317,9 +316,8 @@ However you may see the following warning message:
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.
@@ -331,3 +329,32 @@ 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.
diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
index 68eeebc17ff4..1faf76383bab 100644
--- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -1172,7 +1172,7 @@
}
/* PCI IDs */
- static struct pci_device_id snd_mychip_ids[] = {
+ static struct pci_device_id snd_mychip_ids[] __devinitdata = {
{ PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR,
PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
....
@@ -1565,7 +1565,7 @@
<informalexample>
<programlisting>
<![CDATA[
- static struct pci_device_id snd_mychip_ids[] = {
+ static struct pci_device_id snd_mychip_ids[] __devinitdata = {
{ PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR,
PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
....