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Fixes generated by 'codespell' and manually reviewed.
Signed-off-by: Lucas De Marchi <lucas.demarchi@profusion.mobi>
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The driver was using an initial value for the clock on the SPI bus
which was read from ICE1712 EEPROM,
ice->eeprom.data[ICE_EEP1_GPIO_STATE] & ICE1712_DELTA_AP_CCLK (0x02)
It appears some cards have it default high, some cards
have it default low. On my Delta 66 rev. E:
$ cat /proc/asound/M66/ice1712 | grep 'GPIO state'
GPIO state : 0x70 /* ICE1712_DELTA_AP_CCLK bit is zero */
On my Audiophile 2496:
$ cat /proc/asound/M2496/ice1712 | grep 'GPIO state'
GPIO state : 0xfe /* ICE1712_DELTA_AP_CCLK bit is one */
It must be raised before the first SPI write happens, or the write will
fail, leading to:
[ 23.248721] invalid CS8427 signature 0x0: let me try again...
I theorize that 4eb4550ab37d351ab0973ccec921a5a2d8560ec7
is no longer needed, it was a different way to workaround
the problem.
[fixed variable decleration by tiwai]
Signed-off-by: Brian Bloniarz <brian.bloniarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Rev. E of the M-Audio Delta 66 is partially supported (commit
ef2cd2ccad66b4aba518eca7514eface267ee0f3), but the layout of the GPIO
pins was still unclear. This patch adds the GPIO definitions so that
communication to the CS8247 & 2x AK4524 works correctly.
ALSA bug#3327 has more details; users cap & jhunt report there that the
GPIO wiring is similar to the Digigram VX442 (chip select: pin 4 =
CS8427, pin 5 = AK4524 #0, pin 6 = AK4524 #1). There has been a lot of
conflicting information in the bug, but given these definitions, my
Delta 66E works; I tested analog in&out at 44.1kHz & 96kHz, analog gain
settings, S/PDIF clock sync, and S/PDIF in&out at 44.1kHz.
Signed-off-by: Brian Bloniarz <brian.bloniarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This device is similar to the M-Audio Delta 1010LT in that it uses the
AK4524VF ADC/DAC, but it does not use the CS8427 for SPDIF.
The SPDIF appears to be set up correctly, but I am not able to test it
as I do not have any devices that use it.
This patch makes the ADC/DAC's and the hardware mixer visible to apps
such as alsamixer and envy24control.
Signed-off-by: Garnet MacPhee <dhubsith@comcast.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The capture source control of maya44 was wrongly coded with the bit
shift instead of the bit mask. Also, the slot for line-in was
wrongly assigned (slot 5 instead of 4).
Reported-by: Alex Chernyshoff <alexdsp@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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implicit slab.h inclusion from percpu.h
percpu.h is included by sched.h and module.h and thus ends up being
included when building most .c files. percpu.h includes slab.h which
in turn includes gfp.h making everything defined by the two files
universally available and complicating inclusion dependencies.
percpu.h -> slab.h dependency is about to be removed. Prepare for
this change by updating users of gfp and slab facilities include those
headers directly instead of assuming availability. As this conversion
needs to touch large number of source files, the following script is
used as the basis of conversion.
http://userweb.kernel.org/~tj/misc/slabh-sweep.py
The script does the followings.
* Scan files for gfp and slab usages and update includes such that
only the necessary includes are there. ie. if only gfp is used,
gfp.h, if slab is used, slab.h.
* When the script inserts a new include, it looks at the include
blocks and try to put the new include such that its order conforms
to its surrounding. It's put in the include block which contains
core kernel includes, in the same order that the rest are ordered -
alphabetical, Christmas tree, rev-Xmas-tree or at the end if there
doesn't seem to be any matching order.
* If the script can't find a place to put a new include (mostly
because the file doesn't have fitting include block), it prints out
an error message indicating which .h file needs to be added to the
file.
The conversion was done in the following steps.
1. The initial automatic conversion of all .c files updated slightly
over 4000 files, deleting around 700 includes and adding ~480 gfp.h
and ~3000 slab.h inclusions. The script emitted errors for ~400
files.
2. Each error was manually checked. Some didn't need the inclusion,
some needed manual addition while adding it to implementation .h or
embedding .c file was more appropriate for others. This step added
inclusions to around 150 files.
3. The script was run again and the output was compared to the edits
from #2 to make sure no file was left behind.
4. Several build tests were done and a couple of problems were fixed.
e.g. lib/decompress_*.c used malloc/free() wrappers around slab
APIs requiring slab.h to be added manually.
5. The script was run on all .h files but without automatically
editing them as sprinkling gfp.h and slab.h inclusions around .h
files could easily lead to inclusion dependency hell. Most gfp.h
inclusion directives were ignored as stuff from gfp.h was usually
wildly available and often used in preprocessor macros. Each
slab.h inclusion directive was examined and added manually as
necessary.
6. percpu.h was updated not to include slab.h.
7. Build test were done on the following configurations and failures
were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my
distributed build env didn't work with gcov compiles) and a few
more options had to be turned off depending on archs to make things
build (like ipr on powerpc/64 which failed due to missing writeq).
* x86 and x86_64 UP and SMP allmodconfig and a custom test config.
* powerpc and powerpc64 SMP allmodconfig
* sparc and sparc64 SMP allmodconfig
* ia64 SMP allmodconfig
* s390 SMP allmodconfig
* alpha SMP allmodconfig
* um on x86_64 SMP allmodconfig
8. percpu.h modifications were reverted so that it could be applied as
a separate patch and serve as bisection point.
Given the fact that I had only a couple of failures from tests on step
6, I'm fairly confident about the coverage of this conversion patch.
If there is a breakage, it's likely to be something in one of the arch
headers which should be easily discoverable easily on most builds of
the specific arch.
Signed-off-by: Tejun Heo <tj@kernel.org>
Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org>
Cc: Ingo Molnar <mingo@redhat.com>
Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
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Add proper suspend/resume code for Terratec Aureon cards.
Based on ice1724 suspend/resume work of Igor Chernyshev.
Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4944
Tested on linux-2.6.32.9
Signed-off-by: Bernhard Urban <lewurm@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into devel
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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>
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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>
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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>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: ice1724 - aureon - modify WM8770 Master & DAC volume
ALSA: hda/realtek: quirk for D945GCLF2 mainboard
ALSA: hda - Terradici HDA controllers does not support 64-bit mode
ALSA: document: Add direct git link to grub hda-analyzer
ALSA: radio/sound/miro: fix build, cleanup depends/selects
ALSA: hda - Generalize EAPD inversion check in patch_analog.c
ASoC: Wrong variable returned on error
ALSA: snd-usb-us122l: add product IDs of US-122MKII and US-144MKII
ALSA: hda - Exclude unusable ADCs for ALC88x
ALSA: hda - Add missing Line-Out and PCM switches as slave
ALSA: hda - iMac 9,1 sound patch.
ALSA: opti93x: set MC indirect registers base from PnP data
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git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (42 commits)
tree-wide: fix misspelling of "definition" in comments
reiserfs: fix misspelling of "journaled"
doc: Fix a typo in slub.txt.
inotify: remove superfluous return code check
hdlc: spelling fix in find_pvc() comment
doc: fix regulator docs cut-and-pasteism
mtd: Fix comment in Kconfig
doc: Fix IRQ chip docs
tree-wide: fix assorted typos all over the place
drivers/ata/libata-sff.c: comment spelling fixes
fix typos/grammos in Documentation/edac.txt
sysctl: add missing comments
fs/debugfs/inode.c: fix comment typos
sgivwfb: Make use of ARRAY_SIZE.
sky2: fix sky2_link_down copy/paste comment error
tree-wide: fix typos "couter" -> "counter"
tree-wide: fix typos "offest" -> "offset"
fix kerneldoc for set_irq_msi()
spidev: fix double "of of" in comment
comment typo fix: sybsystem -> subsystem
...
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The volume levels in original implementation are incorrect and does
not match the dB scale. The real range is linear (in the sense of
the dB scale) from 0dB to -100dB. Remove logaritmic table and make
all volumes from range 0dB..100dB.
The tests are in RedHat's bugzilla #540817.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Conflicts:
kernel/irq/chip.c
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This is a clean up and doesn't change the behavior.
Bit fields should always be unsigned. Otherwise pm_suspend_enabled will
be -1 when you want it to be 1. The other bad thing is that the sparse
checker will complain 36 times if they aren't unsigned.
The other bitfields in that struct are unsigned already.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Remove invlid __devinit prefix from the suspend callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add proper suspend/resume code for Juli@ cards. Based on ice1724
suspend/resume work of Igor Chernyshev.
Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413
Tested on linux-2.6.31.6
Signed-off-by: Aleksey Kunitskiy <alexey.kv@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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something-bility is spelled as something-blity
so a grep for 'blit' would find these lines
this is so trivial that I didn't split it by subsystem / copy
additional maintainers - all changes are to comments
The only purpose is to get fewer false positives when grepping
around the kernel sources.
Signed-off-by: Dirk Hohndel <hohndel@infradead.org>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
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If two streams are started immediately after one another (such as a
playback and a recording stream), the call to set hw params fails with
EBUSY. This patch makes the call succeed, so playback and recording will
work properly.
Signed-off-by: David Henningsson <launchpad.web@epost.diwic.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix the num_total_dacs setting for Chaintech AV710. The existing comment
that only PSDOUT0 is connected is correct, but since the card is using
packed AC97 mode to send 6 channels to the codec, num_total_dacs should be
set to 6 and not 2. This allows 6-channel surround to work. Also clarify
a comment regarding the additional WM8728 codec on this card (it's connected
to the SPDIF output and always receives the same data).
Signed-off-by: Robert Hancock <hancockrwd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Increase the default and maximum PCM buffer prellocation size for ice1724's
SPDIF and independent stereo pair outputs to 256K, which is the hardware's
maximum supported size. This allows a reduction in interrupt rate and
potentially power usage when an application is not latency-critical.
Signed-off-by: Robert Hancock <hancockrwd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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MIXER to PCM type
* PLEASE NOTE - this change requires the corresponding update of
envy24control for ice1712 - kind of an ABI change.
* The "Multi Track Peak" control is read-only level meters indicator.
* The control is VERY confusing to most users since it is currently displayed
in regular mixers. E.g. alsamixer ignores its read-only status
and allows changing the levels with keys which makes no sense.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* three external clock types
* all controls supported
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* Support for customization of the external clock names
* Adding hooks to playback_pro_open and capture_pro_open, allowing e.g.
limiting available stream rates to a single value when the external
clock rate is detected
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* pro-rate-locking applies to internal clock mode only
* required rate and current rate are compared for internal clock mode only
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* get/set routines for GPIO mask and direction
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* complete support for ak4620
* codec regs listed in proc for all codecs/chips
* adding total regs for each codec
* fixing nb. of steps in input attenuation controls
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/ice1724-pm:
ALSA: ice1724 - Fix section mismatch
ALSA: ice1724 - Patch for suspend/resume for Audiotrak Prodigy HD2
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Now snd_vt1724_chip_reset() is used in the resume callback, thus
it cannot be __devinit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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I've built a small HTPC and had to add suspend/resume support in ice1724
driver. There seem to be 3 existing bugs related to that:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3748
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2314
Due to hardware (un)availability, I only enabled the fix for Audiotrak
Prodigy HD2 card, which is installed in my HTPC. However, most of my code
should be reusable in the future on other ice1724-based cards as well (as
long as people add card-specific peices of code). The fix is currently based
on ALSA 1.0.20 and works on my MythBuntu 9.04 HTPC (using 2.6.28-11 kernel).
Signed-off-by: Igor Chernyshev <igor.ch75+alsa at gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Added the support for ESI Maya44 board to ice1724 driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Added a new flag, own_routing, to allow spec drivers to create own
routing controls. Also, the basic get/put calls are changed to be
external for later use by maya44 driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add the PCI posting to ensure the reset sequence in snd_vt1724_chip_reset().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Rename some vt1724_pcm_reg records to more generic and consistent ones.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The set_rate might return error but the current code doesn't check it.
This patch adds a proper error check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Replace all DMA_28BIT_MASK macro with DMA_BIT_MASK(28)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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Signed-off-by: Takashi Iwai <tiwai@suse.de>
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MIDI_TX IRQ seems always pending when any bytes on FIFO is available.
Thus, it's better to enable MPU_TX only when any bytres are really
stored in the substream, and disables immediately when the queue
becomes empty.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Convert from snd_card_new() to the new snd_card_create() function
in sound/pci/*.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix trivial name string typo as reported in bug 2552.
Signed-off-by: Alan Horstmann <gineera@aspect135.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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