diff options
author | Linus Torvalds <torvalds@g5.osdl.org> | 2006-05-01 07:46:46 -0700 |
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committer | Linus Torvalds <torvalds@g5.osdl.org> | 2006-05-01 07:46:46 -0700 |
commit | 494b9aea6d451e1eaab5d52b65951d7dc6e81cb8 (patch) | |
tree | ea70b0d3934a3a7f468d285833029798be24d5e1 /Documentation | |
parent | e0a515bc6a2188f02916e976f419a8640312e32a (diff) | |
parent | a769577b3716c757e354a681aab3524ac6b651be (diff) |
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.txt | 81 | ||||
-rw-r--r-- | Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 4 |
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, }, .... |