Commit graph

6809 commits

Author SHA1 Message Date
Andrea Gelmini
76b53774c5 sound/oss/v_midi.h: Checkpatch cleanup
sound/oss/v_midi.h:5: ERROR: code indent should use tabs where possible
sound/oss/v_midi.h:7: ERROR: trailing whitespace

Signed-off-by: Andrea Gelmini <andrea.gelmini@gelma.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-02 11:22:08 +01:00
Daniel Mack
de48c7bc6f ALSA: usbaudio: consolidate header files
Use the definitions from linux/usb/audio.h all over the ALSA USB audio
driver and add some missing definitions there as well.

Use the endpoint attribute macros from linux/usb/ch9 and remove the own
things from sound/usb/usbaudio.h.

Now things are also nicely prefixed which makes understanding the code
easier.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:51:56 +01:00
Daniel Mack
7b8a043f26 ALSA: usbmixer: bail out early when parsing audio class v2 descriptors
This is just a quick hack that needs to be removed once the new units
defined by the audio class v2.0 standard are supported.

However, it allows using these devices for now, without mixer support.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:40:26 +01:00
Daniel Mack
53ee98fe8a ALSA: usbaudio: implement basic set of class v2.0 parser
This adds a number of parsers for audio class v2.0. In particular, the
following internals are different and now handled by the code:

* the number of streaming interfaces is now reported by an interface
  association descriptor. The old approach using a proprietary
  descriptor is deprecated.

* The number of channels per interface is now stored in the AS_GENERAL
  descriptor (used to be part of the FORMAT_TYPE descriptor).

* The list of supported sample rates is no longer stored in a variable
  length appendix of the format_type descriptor but is retrieved from
  the device using a class specific GET_RANGE command.

* Supported sample formats are now reported as 32bit bitmap rather than
  a fixed value. For now, this is worked around by choosing just one of
  them.

* A devices needs to have at least one CLOCK_SOURCE descriptor which
  denotes a clockID that is needed im the class request command.

* Many descriptors (format_type, ...) have changed their layout. Handle
  this by casting the descriptors to the appropriate structs.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:40:24 +01:00
Daniel Mack
8fee4aff8c ALSA: usbaudio: introduce new types for audio class v2
This patch adds some definitions for audio class v2.

Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have
different numerical representations in both standards, so there is need
for a _V1 add-on now. usbmixer.c is changed accordingly.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:40:20 +01:00
Daniel Mack
28e1b77308 ALSA: usbaudio: parse USB descriptors with structs
In preparation of support for v2.0 audio class, use the structs from
linux/usb/audio.h and add some new ones to describe the fields that are
actually parsed by the descriptor decoders.

Also, factor out code from usb_create_streams(). This makes it easier to
adopt the new iteration logic needed for v2.0.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:40:12 +01:00
Chris J Arges
40717382e0 ALSA: usbaudio Mbox support, output only
Signed-off-by: Chris J Arges <christopherarges@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 09:56:26 +01:00
Florian Zumbiehl
04510a74bf ALSA: cs46xx - fix some typos
Signed-off-by: Florian Zumbiehl <florz@florz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-18 08:12:30 +01:00
Florian Zumbiehl
7fb2d723e6 ALSA: cs46xx - Do test writes to register AC97_REC_GAIN in
snd_cs46xx_codec_reset() bypassing the register cache, so as to not
clobber the cached register value during resume.

Signed-off-by: Florian Zumbiehl <florz@florz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-18 08:10:54 +01:00
Takashi Iwai
7fb3a069bc Merge branch 'fix/misc' into topic/misc
Conflicts:
	sound/pci/hda/patch_realtek.c
2010-02-17 14:24:46 +01:00
Takashi Iwai
9d3415a8cc Merge remote branch 'alsa/fixes' into fix/misc 2010-02-17 14:22:21 +01:00
Giuliano Pochini
b721e68bdc ALSA: Echoaudio, fix Guru Meditation #00000005.48454C50
This patch fixes a division by zero error in the irq handler.

There is a small window between the hw_params() callback and when
runtime->frame_bits is set by ALSA middle layer. When another substream is
already running, if an interrupt is delivered during that window the irq
handler calls pcm_pointer() which does a division by zero. The patch below
makes the irq handler skip substreams that are initialized but not started
yet. Cc to Clemens Ladisch because he proposed an alternate fix.

For more information, please read the original thread in the linux-kernel
mailing list: http://lkml.org/lkml/2010/2/2/187

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-17 13:02:29 +01:00
Jaroslav Kysela
291186e049 ALSA: usbmixer - use MAX_ID_ELEMS where possible
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16 12:00:45 +01:00
Jaroslav Kysela
7affdc17d4 ALSA: usbmixer - add usb_id value to usbmixer proc file
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16 12:00:42 +01:00
Jaroslav Kysela
3be522a951 ALSA: pcm core - fix fifo_size channels interval check
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@kernel.org>
2010-02-16 12:00:20 +01:00
Jaroslav Kysela
ebfdeea3df ALSA: usbmixer - introduce /proc/asound/card#/usbmixer file
The usbmixer proc file contains mapping between ALSA control API and
USB mixer control units. The purpose of this file is for debugging
and a problem diagnostics.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16 11:25:55 +01:00
Jaroslav Kysela
b8f1f5983f Merge branch 'topic/misc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into devel 2010-02-16 11:25:03 +01:00
Jaroslav Kysela
ba9341dfef Merge branch 'fixes' into devel 2010-02-16 11:19:18 +01:00
Sebastien Alaiwan
d39e82db73 ALSA: USB MIDI support for Access Music VirusTI
Here's a patch that adds MIDI support through USB for one of the Access
Music synths, the VirusTI.

The synth uses standard USBMIDI protocol on its USB interface 3, although
it does signal "vendor specific" class. A magic string has to be sent on
interface 3 to enable the sending of MIDI from the synth (this string was
found by sniffing usb communication of the Windows driver). This is all
my patch does, and it works on my computer.

Please note that the synth can also do standard usb audio I/O on its
interfaces 2&3, which already works with the current snd-usb-audio driver,
except for the audio input from the synth. I'm going to work on it when I
have some time.

Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> (cosmetics, list terminator)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16 09:34:56 +01:00
Clemens Ladisch
f167e1d073 ALSA: usb-audio: reduce MIDI packet size to work around broken firmware
Extend the list of devices whose firmware does not expect more than one
USB MIDI packet in one USB packet.

bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3752

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16 08:08:01 +01:00
Giuliano Pochini
47b5d028fd ALSA: Echoaudio - Add suspend support #2
This patch adds rearranges parts of the initialization code and adds
suspend and resume callbacks.

This patch adds suspend and resume callbacks.
It also rearranges parts of the initialization code so it can be
used in both the first initialization (when the module is loaded we
also have to load default settings) and the resume callback (where
we have to restore the previous settings).

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-15 10:40:15 +01:00
Giuliano Pochini
ad3499f466 ALSA: Echoaudio - Add suspend support #1
Move the controls init code outside the init_hw() function because is must
not be called during resume.

This patch moves the code that initializes the card's controls with
default valued from the init_hw() function into a separated
set_mixer_defaults() function (one for each of the 16 supported
cards). This change is necessary because during resume we must
resurrect the hardware without losing the previous
settings. set_mixer_defaults() must be called only once when the
module is loaded.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-15 10:39:22 +01:00
Giuliano Pochini
4f8ada444c ALSA: Echoaudio - Add firmware cache #2
This patch implements a simple cache for the firmware files when CONFIG_PM is defined.

This patch changes get_firmware(), free_firmware() and adds
free_firmware_cache(). The first two functions implement a very
simple cache and the latter is used to actually release all the stored
firmwares when the module is unloaded. 
When CONFIG_PM is not enabled those functions act as before, that is
free_firmware() releases the firmware immediately and
free_firmware_cache() does nothing.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-15 10:38:10 +01:00
Giuliano Pochini
19b5006378 ALSA: Echoaudio - Add firmware cache #1
Changes the way the firmware is passed through functions.

When CONFIG_PM is enabled the firmware cannot be released because the
driver will need it again to resume the card. 
With this patch the firmware is passed as an index of the struct
firmware card_fw[] in place of a pointer. That same index is then used
to locate the firmware in the firmware cache.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-15 10:36:51 +01:00
Linus Torvalds
e99cc290ca Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - use WARN_ON_ONCE() for zero-division detection
2010-02-12 10:12:28 -08:00
Takashi Iwai
d6d8bf5493 ALSA: hda - use WARN_ON_ONCE() for zero-division detection
Replace the zero-division warning message with WARN_ON_ONCE() per the
advice by Linus.  This shouldn't happen, but if it happens, it's
possible that the bug happens often due to buggy IRQs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-12 18:20:04 +01:00
Linus Torvalds
0e9695d9a4 Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda-intel: Avoid divide by zero crash
2010-02-12 08:48:47 -08:00
Takashi Iwai
a540e13386 Merge remote branch 'alsa/devel' into topic/misc 2010-02-12 10:42:38 +01:00
Jaroslav Kysela
c3a3e040f0 ALSA: usbmixer - add possibility to remap dB values
USB devices tends to represent dB ranges in different way than ALSA expects.
Add possibility to override these values and add guessed values for
SoundBlaster MP3+.

Also rename 'Capture Input Source' control to 'Capture Source' for
SoundBlaster MP3+ and Extigy.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-11 18:00:16 +01:00
Jody Bruchon
fed08d036f ALSA: hda-intel: Avoid divide by zero crash
On my AMD780V chipset, hda_intel.c can crash the kernel with a divide by
zero
for as-yet unknown reasons. A simple check for zero prevents it, though
the problem that causes it remains. Since the workaround is harmless and
won't affect anyone except victims of this bug, it should be safe;
moreover,
because this crash can be triggered by a user-mode application, there are
denial of service implications on the systems affected by the bug without
the patch.

Signed-off-by: Jody Bruchon <jody@nctritech.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-09 21:33:33 +01:00
Alexey Dobriyan
cebe41d4b8 sound: use DEFINE_PCI_DEVICE_TABLE
Use DEFINE_PCI_DEVICE_TABLE() to make PCI device ids go to
.devinit.rodata section, so they can be discarded in some cases,
and make them const.

Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-09 11:08:33 +01:00
Takashi Iwai
3e0b33f786 Merge remote branch 'alsa/fixes' into for-linus 2010-02-05 19:57:23 +01:00
Takashi Iwai
a26a408888 Merge branch 'fix/asoc' into for-linus 2010-02-05 19:57:16 +01:00
Takashi Iwai
db9256c003 Merge branch 'fix/hda' into for-linus 2010-02-05 19:56:55 +01:00
Grazvydas Ignotas
3b9447fb7f ASoC: pandora: Add APLL supply to fix audio output
Pandora's external DAC is using 256*Fs output from the TWL4030
codec, and TWL4030 needs to have APLL enabled for it's 256*Fs
output to function.

Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-05 12:35:35 +00:00
Jaroslav Kysela
9d4c746445 ALSA: ice1724 - aureon - fix wm8770 volume offset
The volume register is from 0..0x7f and 0..0x1a range is mute.
Also, fix mute combining in wm_vol_put(). The wrong behaviour was
noticed by Peter Christensen.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-05 10:24:25 +01:00
Maxim Levitsky
9492837a6f ALSA: cosmetic: make hda intel interrupt name consistent with others
This renames the interrupt name in /proc/interrupt.
HDA Intel -> hda_intel

This also eliminates space from the name, probably helping some
parsers.
Don't think anybody depends on this name in userspace

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-05 09:08:14 +01:00
Maxim Levitsky
1eb6dc7dab ALSA: hda - Delay switching to polling mode if an interrupt was missing
My sound codec seems sometimes (very rarely) to omit interrupts (ALC268)
However, interrupt mode still works.
Thus if we get timeout, poll the codec once.

If we get 3 such polls in a row, then switch to polling mode.

This patch is maybe an bandaid, but this might be a workaround for hardware bug.

Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-05 09:07:21 +01:00
Sebastien Alaiwan
350a514787 ALSA: ice1712: fix: lock samplerate when samplerate locking is enabled
I found that the sampling rate locking setting of the ice1712 sound driver
was only half-respected : when the driver was locked to, let's say, 44100Hz,
and a usermode app was requesting 48000Hz playback, the request was succesful
although the soundcard would continue to run at 44100Hz.

Here's a patch that will make those requests to fail.

Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-05 08:58:20 +01:00
Jaroslav Kysela
21956b61f5 ALSA: ctxfi - fix PTP address initialization
After hours of debugging, I finally found the reason why some source
and runtime combination does not work. The PTP (page table pages)
address must be aligned. I am not sure how much, but alignment to
PAGE_SIZE is sufficient. Also, use ALSA's page allocation routines
to ensure proper virtual -> physical address translation.

Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-04 21:48:00 +01:00
Jaroslav Kysela
d5e1ca05f7 ALSA: dummy driver - add model parameter
This is a cleanup for the dummy driver. The model kernel module parameter
is introduced to select the soundcard emulation.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-02 17:50:57 +01:00
Thadeu Lima de Souza Cascardo
c85a400499 ALSA: trivial: sound seq ioctl dbg: print hexadecimal value padded with 0s
Instead of padding with blanks and printing "number=0x a", print
"number=0x0a".

Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-02 00:27:47 +01:00
Takashi Iwai
30ede1b9f0 Merge remote branch 'alsa/devel' into topic/misc 2010-02-01 15:46:00 +01:00
Clemens Ladisch
6123637faf sound: control: fix minimum TLV length
Allow TLV blocks that do not have any values; the smallest possible TLV
is an empty container or one where the information is only in the tag.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-01 14:12:12 +01:00
Clemens Ladisch
a75d7a4cf5 sound: control: actually allow TLV command access
Creating a control with TLV_COMMAND access was not possible because
snd_ctl_new1() forgot to include it in the mask of allowable access
bits.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-01 14:11:52 +01:00
Takashi Iwai
f3f1e14ce9 Merge branch 'fix/asoc' into for-linus 2010-01-31 14:41:05 +01:00
Takashi Iwai
74ce25c0ee Merge branch 'fix/hda' into for-linus 2010-01-31 14:40:58 +01:00
Anuj Aggarwal
5bbd4953a4 ASoC: AM3517: ASoC driver not getting compiled
Commit 761c9d45 (ASoC: Fix build of OMAP sound drivers) changes
CONFIG_MACH_OMAP3517EVM -> CONFIG_SND_OMAP_SOC_OMAP3517EVM in the
Makefile. Whereas the config option defined in Kconfig is
SND_OMAP_SOC_AM3517EVM. Because of this, ASoC driver for AM3517
was not getting compiled.

Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-29 13:43:51 +00:00
Anuj Aggarwal
3e59aaa7ae ASoC: AIC23: Fixing writes to non-existing registers in resume function
Commit e9ff5eb2 (Fixing infinite loop in resume path) uses wrong AIC23
register in resume function because of which register writes happen
on some non-existing registers.

Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-29 13:42:37 +00:00
Takashi Iwai
8ce28d6abf ALSA: hda - Add an ASUS mobo to MSI blacklist
Sid Boyce reported that his machine locks up without enable_msi=0 option.
This looks like another ASUS mobo with Nvidia combo.

Reported-by: Sid Boyce <sboyce@blueyonder.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-27 20:26:08 +01:00