Commit 494b9aea authored by Linus Torvalds's avatar Linus Torvalds
Browse files

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
  ...
parents e0a515bc a769577b

Documentation/sound/alsa/Audiophile-Usb.txt

+54 −27
Original line number Diff line number Diff line 
	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>
@@ -22,7 +22,7 @@ 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
@@ -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.

Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl

+2 −2
Original line number Diff line number Diff line
@@ -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, },

          ....

drivers/media/video/cx88/cx88-alsa.c

+1 −1
Original line number Diff line number Diff line
@@ -616,7 +616,7 @@ static struct snd_kcontrol_new snd_cx88_capture_volume = { 
 * Only boards with eeprom and byte 1 at eeprom=1 have it

 */



static struct pci_device_id cx88_audio_pci_tbl[] = {

static struct pci_device_id cx88_audio_pci_tbl[] __devinitdata = {

	{0x14f1,0x8801,PCI_ANY_ID,PCI_ANY_ID,0,0,0},

	{0x14f1,0x8811,PCI_ANY_ID,PCI_ANY_ID,0,0,0},

	{0, }

include/sound/pcm.h

+5 −1
Original line number Diff line number Diff line
@@ -374,12 +374,14 @@ struct snd_pcm_substream { 
	/* -- OSS things -- */

	struct snd_pcm_oss_substream oss;

#endif

#ifdef CONFIG_SND_VERBOSE_PROCFS

	struct snd_info_entry *proc_root;

	struct snd_info_entry *proc_info_entry;

	struct snd_info_entry *proc_hw_params_entry;

	struct snd_info_entry *proc_sw_params_entry;

	struct snd_info_entry *proc_status_entry;

	struct snd_info_entry *proc_prealloc_entry;

#endif

	/* misc flags */

	unsigned int no_mmap_ctrl: 1;

	unsigned int hw_opened: 1;
@@ -400,12 +402,14 @@ struct snd_pcm_str { 
	struct snd_pcm_oss_stream oss;

#endif

	struct snd_pcm_file *files;

#ifdef CONFIG_SND_VERBOSE_PROCFS

	struct snd_info_entry *proc_root;

	struct snd_info_entry *proc_info_entry;

#ifdef CONFIG_SND_DEBUG

#ifdef CONFIG_SND_PCM_XRUN_DEBUG

	unsigned int xrun_debug;	/* 0 = disabled, 1 = verbose, 2 = stacktrace */

	struct snd_info_entry *proc_xrun_debug_entry;

#endif

#endif

};



struct snd_pcm {

include/sound/pcm_oss.h

+2 −0
Original line number Diff line number Diff line
@@ -75,7 +75,9 @@ struct snd_pcm_oss_substream { 
struct snd_pcm_oss_stream {

	struct snd_pcm_oss_setup *setup_list;	/* setup list */

	struct mutex setup_mutex;

#ifdef CONFIG_SND_VERBOSE_PROCFS

	struct snd_info_entry *proc_entry;

#endif

};



struct snd_pcm_oss {

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