BugLink: https://launchpad.net/bugs/580749
Symptom: on the original reporter's VIA VT1708-based board, the
PulseAudio daemon dies shortly after the user attempts to play an audio
file.
Test case: boot from Ubuntu 10.04 LTS live cd; attempt to play an audio
file.
Resolution: add SSID for the original reporter's hardware to the
position_fix quirk table, explicitly specifying the LPIB method.
Reported-and-Tested-By: Harald
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/542550
Symptom: On the reporter's iMac, in Ubuntu 10.04 LTS neither playback
nor capture appear audible out-of-the-box.
Test case: Boot from an Ubuntu 10.04 LTS live cd or from an installed
configuration and attempt to play or capture audio.
Resolution: Specify the mb31 quirk for this machine in the codec SSID
table.
Reported-and-Tested-By: f3a97
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/465942
Symptom: On the reporter's ASUS device, using PulseAudio in Ubuntu 10.04
LTS results in the PA daemon crashing shortly after attempting to select
capture or to configure the audio hardware profile.
Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or
Linux 2.6.34, adjust the HDA device's capture volume with PulseAudio.
Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.
Reported-and-Tested-By: Irihapeti
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This defines the 24bps and 40bps (8khz sample rate) G.723 codec
formats. They are going to be used once I submit the driver for
an mpeg4/g723 compression card.
I've updated the signed value to -1 as per Takashi's comments
since these are non-linear formats.
Signed-off-by: Ben Collins <bcollins@bluecherry.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch registers the WM8750 codec on a proper place on the SPITZ machine
after the WM8750 driver was converted to new API.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (26 commits)
ALSA: snd-usb-caiaq: Bump version number to 1.3.21
ALSA: Revert "ALSA: snd-usb-caiaq: Set default input mode of A4DJ"
ALSA: snd-usb-caiaq: Simplify single case to an 'if'
ALSA: snd-usb-caiaq: Restore 'Control vinyl' input mode on A4DJ
ALSA: hda: Use LPIB for a Shuttle device
ALSA: hda: Add support for another Lenovo ThinkPad Edge in conexant codec
ALSA: hda: Use LPIB for Sony VPCS11V9E
ALSA: usb-audio: fix feature unit parser for UAC2
ALSA: asihpi - Minor code cleanup
ALSA: asihpi - Add support for new ASI8800 family
ALSA: asihpi - Fix bug preventing outstream_write preload from happening
ALSA: asihpi - Fix imbalanced lock path in hw_message
ALSA: asihpi - Remove support for old ASI8800 family
ALSA: asihpi - Add hd radio blend functions
ALSA: asihpi - Remove unused io map functions
ALSA: usb-audio: add support for UAC2 pitch control
ALSA: usb-audio: parse UAC2 endpoint descriptors correctly
ALSA: usb-audio: fix return values
ALSA: usb-audio: parse more format descriptors with structs
sound: Add missing spin_unlock
...
Do not explicity set the default input mode. Use the hardware default
of mode 0 ('Control vinyl'), which is now available.
This reverts commit e3ca4c9.
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After removing code, only one case remains. So use an 'if' instead.
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This feature was undocumented on early A4DJ units. It is indicated
by lighting both the 'line' and 'phono' lamps at the same time.
Newer units document this and the newer Windows drivers enable this
for all units, so restore the functionality.
This patch simplifies the code and changes the mode mapping to match
the A8DJ, favouring simpler code and consistency over keeping the
existing mapping.
Both 'Control vinyl' and 'Phono' input modes enable the hardware
preamp. The difference is the input impedance.
This reverts commit 9a9527e.
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/551949
Symptom: On the reporter's Shuttle device, using PulseAudio in Ubuntu
10.04 LTS results in "popping clicking" audio with the PA crashing
shortly thereafter.
Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or
Linux 2.6.34, adjust the HDA device's volume with PulseAudio.
Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.
Reported-and-Tested-By: Christian Mehlis <mehlis@inf.fu-berlin.de>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On a Thinkpad Edge 13 "01972NG" I had the problem that speakers played
sound although headphones were plugged in. Using model=ideapad with
latest alsa-git kernel fixed this. So adding this quirk to use ideapad
for another Thinkpad Edge variant seems sensible.
Cc: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Andreas Herrmann <andreas.herrmann3@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/586347
Symptom: On the Sony VPCS11V9E, using GStreamer-based applications with
PulseAudio in Ubuntu 10.04 LTS results in stuttering audio. It appears
to worsen with increased I/O.
Test case: use Rhythmbox under increased I/O pressure. This symptom is
reproducible in the current daily stable alsa-driver snapshots (at least
up until 21 May 2010; later snapshots fail to build from source due to
missing preprocessor directives when compiled against 2.6.32).
Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.
Reported-and-Tested-By: Lauri Kainulainen <lauri@sokkelo.net>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a small off-by-one bug which causes the feature unit to announce a
wrong number of channels. This leads to illegal requests sent to the
firmware eventually.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The S5PV210 and S5PC110 has the AC97 controller same as S3C6410.
Simply enable the options to build the drivers for S5PC110 and
S5PV210 also.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The S5PC100 has the AC97 controller same as S3C6410.
Simply enable the options to build the drivers for
S5PC100 also.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add hd radio blend functions. HPI version inc to 4.03.25.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This request is again handled differently in comparison to UAC1.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC2 devices have their information about pitch control stored in a
different field. Parse it, and emulate the bits for a v1 device.
A new struct uac2_iso_endpoint_descriptor is added.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
-1 is not a good return value as it means -EPERM, "not permitted".
Choose -ENOTSUPP instead, which is what the code really wants to tell
its callers.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a spin_unlock missing on the error path.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression E1;
@@
* spin_lock(E1,...);
<+... when != E1
if (...) {
... when != E1
* return ...;
}
...+>
* spin_unlock(E1,...);
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These should be regular, not linear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
These scales should be regular, not linear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
These should be regular rather than linear scales.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
This updates the i.MX SSI driver to make it compatible with the ASoC tree
following the move of DMA parameters from the DAI to the audio substream
object.
Signed-off-by: Stuart Longland <redhatter@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Add a module option to allow the GPR mixer controls to have the full
resolution of the hardware, i.e., 0...2^31-1 instead of 0...100.
Because of bugs in userspace tools like alsactl and alsamixer, this is
not yet enabled by default.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the cleanup of the hw_ptr update functions in 2.6.33, the calculation
of the delta value was changed to use the modulo operator to protect
against a negative difference due to the pointer wrapping around at the
boundary.
However, the ptr variables are unsigned, so a negative difference would
result in the two complement's value which has no relation to the actual
difference relative to the boundary; the result is typically some value
near LONG_MAX-boundary. Furthermore, even if the modulo operation would
be done with signed types, the result of a negative dividend could be
negative.
The invalid delta value is then caught by the following checks, but this
means that the pointer update is ignored.
To fix this, use a range check as in the other pointer calculations.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check that the interrupt raised for a stream is actually a buffer
completion interrupt before handling it as one. Otherwise, memory
errors or FIFO xruns would be interpreted as a pointer update and could
break the stream timing.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/576160
Symptom: Currently (2.6.32.12) the Dell M1730 uses the 3stack model
quirk. Unfortunately this means that capture is not functional out-
of-the-box despite ensuring that capture settings are unmuted and
raised fully.
Test case: boot from Ubuntu 10.04 LTS live cd; capture does not
work.
Resolution: Correct the model quirk for Dell M1730 to rely on the
BIOS configuration.
This patch also trivially sorts the quirk into the correct section
based on the comments.
Reported-and-Tested-By: <picdragon99@msn.com>
Tested-By: Daren Hayward
Tested-By: Tobias Krais
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
First issue:
With the original patch, I've noticed by unmuting the mic
(and even having it muted), there is a distorted("Noise")
coming from the internal speakers, even when the headphones are plugged in.
What my finding's revealed is:
/* Mic (rear) pin: input vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
From the original patch. Looking at codec#0 0x18/0x1a is listed as:
Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0000373c: IN OUT HP Detect
Vref caps: HIZ 50 GRD 80 100
Pin Default 0x90100141: [Fixed] Speaker at Int N/A
Conn = Unknown, Color = Unknown
DefAssociation = 0x4, Sequence = 0x1
Misc = NO_PRESENCE
Pin-ctls: 0x41: OUT VREF_50
Unsolicited: tag=00, enabled=0
Connection: 5
0x0c* 0x0d 0x0e 0x0f 0x26
seems this Node is listed as: [Fixed] Speaker while 0x15
Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x80 0x80]
Pincap 0x0000373c: IN OUT HP Detect
Vref caps: HIZ 50 GRD 80 100
Pin Default 0x018b3020: [Jack] Line In at Ext Rear
Conn = Comb, Color = Blue
DefAssociation = 0x2, Sequence = 0x0
Pin-ctls: 0x01: VREF_50
Unsolicited: tag=00, enabled=0
Connection: 5
0x0c 0x0d* 0x0e 0x0f 0x26
is [Jack] Line In at Ext Rear.
(looking at the other apple products as examples
I came up with the fix below).
Second issue:
alc885_mbp_4ch_modes
The original patch does a good job with the
HP pin automute function, but from what I noticed is I would have to manually
change the channel form 2 to 4 after plugging the headphones in.
And not to mention having odd moments to where I was jamming out
with the headphones on, then later realized I had sound blasting out
of the speakers as well. My findings revealed that changing
alc885_mbp_4ch_modes to alc885_mba21_ch_modes and setting
- spec->autocfg.speaker_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x18;
gets the automute function when the headphones plugged in working
flawlessly(and the no need to manually change the channel number
afterwards).
Third issue:
alc885_imac91_mixer
There probably doesnt need to be anything changed with this
(esspecially if your one to like lots of sliders),but my findings
revealed that mac osx only has a master on the top right,
another switch on itunes, and then a slider for the mic.
So the changes I did below try and mimic osx as much as possible
(only thing I had an issue with is just having one mute switch
on the master, instead of having two(still investigating)).
fourth issue:
alc882_capture_source
I endeded up creating alc889A_imac91_capture_source()
only because looking at alc882_capture_source I see
that the mic is set to 0x1 while this works, I also noticed
that adding 0x1 and 0x01 and testing that 0x1 somehow
stops working, and 0x01 works(so I figured 0x01 was more
of the alpha of the numbers(still need to figure out
where that valuse is)). In any case the microphone
does work with the original, and with the below patch, but both
still record not as clean(lots of "Noise", which I would like to
look into too).
Note: using alsamixer -Va reveals the capture switches.
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/549560
Symptom: on a significant number of hardware, booting from a live cd
results in capture working correctly, but once the distribution is
installed, booting from the install results in capture not working.
Test case: boot from Ubuntu 10.04 LTS live cd; capture works correctly.
Install to HD and reboot; capture does not work. Reproduced with 2.6.32
mainline build (vanilla kernel.org compile)
Resolution: add SSID for Toshiba A100-259 to the position_fix quirk
table, explicitly specifying the LPIB method.
I'll be sending additional patches for these SSIDs as bug reports are
confirmed.
This patch also trivially sorts the quirk table in ascending order by
subsystem vendor.
Reported-and-Tested-by: <davide.molteni@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/583983
Symptom: on a significant number of hardware, booting from a live cd
results in capture working correctly, but once the distribution is
installed, booting from the install results in capture not working.
Test case: boot from Ubuntu 10.04 LTS live cd; capture works correctly.
Install to HD and reboot; capture does not work. Reproduced with 2.6.32
mainline build (vanilla kernel.org compile).
Resolution: add SSID for Acer Aspire 5110 to the position_fix quirk
table, explicitly specifying the LPIB method.
I'll be sending additional patches for these SSIDs as bug reports are
confirmed.
Reported-and-Tested-By: Leo
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the VENDOR/DEVICE ids provided in pci_ids.h instead of creating
local ids of the same values.
Also, fix the following checkpatch.pl warnings:
WARNING: Use #include <linux/io.h> instead of <asm/io.h>
WARNING: unnecessary whitespace before a quoted newline
Signed-off-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* git://git.kernel.org/pub/scm/linux/kernel/git/lethal/sh-2.6: (25 commits)
sh: fix up sh7785lcr_32bit_defconfig.
arch/sh/lib/strlen.S: Checkpatch cleanup
sh: fix up sh7786 dmaengine build.
sh: guard cookie consistency across termination in the DMA driver
sh: prevent the DMA driver from unloading, while in use
sh: fix Oops in the serial SCI driver
sh: allow platforms to specify SD-card supported voltages
mmc: let MFD's provide supported Vdd card voltages to tmio_mmc
sh: disable SD-card write-protection detection on kfr2r09
mfd: pass platform flags down to the tmio_mmc driver
tmio: add a platform flag to disable card write-protection detection
sh: Add SDHI DMA support to migor
sh: Add SDHI DMA support to kfr2r09
sh: Add SDHI DMA support to ms7724se
sh: Add SDHI DMA support to ecovec
mmc: add DMA support to tmio_mmc driver, when used on SuperH
sh: prepare the SDHI MFD driver to pass DMA configuration to tmio_mmc.c
mmc: prepare tmio_mmc for passing of DMA configuration from the MFD cell
sh: add DMA slave definitions to sh7724
sh: add DMA slaves for two SDHI controllers to sh7722
...
Now that DMA slave IDs are only used used in platform specific code and have
become opaque cookies for the rest of the code, we can make the, CPU specific
too.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Merging in current state of Linus' tree to deal with merge conflicts and
build failures in vio.c after merge.
Conflicts:
drivers/i2c/busses/i2c-cpm.c
drivers/i2c/busses/i2c-mpc.c
drivers/net/gianfar.c
Also fixed up one line in arch/powerpc/kernel/vio.c to use the
correct node pointer.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
.name, .match_table and .owner are duplicated in both of_platform_driver
and device_driver. This patch is a removes the extra copies from struct
of_platform_driver and converts all users to the device_driver members.
This patch is a pretty mechanical change. The usage model doesn't change
and if any drivers have been missed, or if anything has been fixed up
incorrectly, then it will fail with a compile time error, and the fixup
will be trivial. This patch looks big and scary because it touches so
many files, but it should be pretty safe.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Sean MacLennan <smaclennan@pikatech.com>
By moving dma_mask into pdev_archdata, and adding archdata to
struct of_device, it makes it possible to substitute of_device
with struct platform_device, which is a stepping stone to
removing the of_platform bus entirely.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
* 'next' of git://git.kernel.org/pub/scm/linux/kernel/git/djbw/async_tx:
DMAENGINE: extend the control command to include an arg
async_tx: trim dma_async_tx_descriptor in 'no channel switch' case
DMAENGINE: DMA40 fix for allocation of logical channel 0
DMAENGINE: DMA40 support paused channel status
dmaengine: mpc512x: Use resource_size
DMA ENGINE: Do not reset 'private' of channel
ioat: Remove duplicated devm_kzalloc() calls for ioatdma_device
ioat3: disable cacheline-unaligned transfers for raid operations
ioat2,3: convert to producer/consumer locking
ioat: convert to circ_buf
DMAENGINE: Support for ST-Ericssons DMA40 block v3
async_tx: use of kzalloc/kfree requires the include of slab.h
dmaengine: provide helper for setting txstate
DMAENGINE: generic channel status v2
DMAENGINE: generic slave control v2
dma: timb-dma: Update comment and fix compiler warning
dma: Add timb-dma
DMAENGINE: COH 901 318 fix bytesleft
DMAENGINE: COH 901 318 rename confusing vars
* 'next' of git://git.kernel.org/pub/scm/linux/kernel/git/benh/powerpc: (92 commits)
powerpc: Remove unused 'protect4gb' boot parameter
powerpc: Build-in e1000e for pseries & ppc64_defconfig
powerpc/pseries: Make request_ras_irqs() available to other pseries code
powerpc/numa: Use ibm,architecture-vec-5 to detect form 1 affinity
powerpc/numa: Set a smaller value for RECLAIM_DISTANCE to enable zone reclaim
powerpc: Use smt_snooze_delay=-1 to always busy loop
powerpc: Remove check of ibm,smt-snooze-delay OF property
powerpc/kdump: Fix race in kdump shutdown
powerpc/kexec: Fix race in kexec shutdown
powerpc/kexec: Speedup kexec hash PTE tear down
powerpc/pseries: Add hcall to read 4 ptes at a time in real mode
powerpc: Use more accurate limit for first segment memory allocations
powerpc/kdump: Use chip->shutdown to disable IRQs
powerpc/kdump: CPUs assume the context of the oopsing CPU
powerpc/crashdump: Do not fail on NULL pointer dereferencing
powerpc/eeh: Fix oops when probing in early boot
powerpc/pci: Check devices status property when scanning OF tree
powerpc/vio: Switch VIO Bus PM to use generic helpers
powerpc: Avoid bad relocations in iSeries code
powerpc: Use common cpu_die (fixes SMP+SUSPEND build)
...
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The decoding/encoding is based on own reverse-engineering. Both control and
data ports are handled. Writing to control port supports SysEx events only,
as this is the only type of messages that MPD16 recognizes.
Signed-off-by: Krzysztof Foltman <wdev@foltman.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 7910b4a1db in 2.6.34 changed the
runtime->boundary calculation to make this value a multiple of both the
buffer_size and the period_size, because the latter is assumed by the
runtime->hw_ptr_interrupt calculation.
However, due to the lack of a ioctl that could read the software
parameters before they are set, the kernel requires that alsa-lib
calculates the boundary value, too. The changed algorithm leads to
a different boundary value used by alsa-lib, which makes, e.g., mplayer
fail to play a 44.1 kHz file because the silence_size parameter is now
invalid; bug report:
<https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5015>.
This patch reverts the change to the boundary calculation, and instead
fixes the hw_ptr_interrupt calculation to be period-aligned regardless
of the boundary value.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When using MCLK is configured for 19.2 Mhz, clock slicer should be
enabled and HPPLL should be bypassed in clock path.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This is needed before the USB merge.
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
For more clearance what the functions actually do,
usb_buffer_alloc() is renamed to usb_alloc_coherent()
usb_buffer_free() is renamed to usb_free_coherent()
They should only be used in code which really needs DMA coherency.
All call sites have been changed accordingly, except for staging
drivers.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Alan Stern <stern@rowland.harvard.edu>
Cc: Pedro Ribeiro <pedrib@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (250 commits)
ALSA: hda: Storage class should be before const qualifier
ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies
ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT
ASoC: sdp4430 - add sdp4430 pcm ops to DAI.
ASoC: TWL6040: Enable earphone path in codec
ASoC: SDP4430: Add support for Earphone speaker
ASoC: SDP4430: Add sdp4430 machine driver
ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF
ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function
ALSA: sound/pci/asihpi: Use kzalloc
ALSA: hdmi - dont fail on extra nodes
ALSA: intelhdmi - add id for the CougarPoint chipset
ALSA: intelhdmi - user friendly codec name
ALSA: intelhdmi - add dependency on SND_DYNAMIC_MINORS
ALSA: asihpi: incorrect range check
ALSA: asihpi: testing the wrong variable
ALSA: es1688: add pedantic range checks
ARM: McBSP: Add support for omap4 in McBSP driver
ARM: McBSP: Fix request for irq in OMAP4
OMAP: McBSP: Add 32-bit mode support
...
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (44 commits)
vlynq: make whole Kconfig-menu dependant on architecture
add descriptive comment for TIF_MEMDIE task flag declaration.
EEPROM: max6875: Header file cleanup
EEPROM: 93cx6: Header file cleanup
EEPROM: Header file cleanup
agp: use NULL instead of 0 when pointer is needed
rtc-v3020: make bitfield unsigned
PCI: make bitfield unsigned
jbd2: use NULL instead of 0 when pointer is needed
cciss: fix shadows sparse warning
doc: inode uses a mutex instead of a semaphore.
uml: i386: Avoid redefinition of NR_syscalls
fix "seperate" typos in comments
cocbalt_lcdfb: correct sections
doc: Change urls for sparse
Powerpc: wii: Fix typo in comment
i2o: cleanup some exit paths
Documentation/: it's -> its where appropriate
UML: Fix compiler warning due to missing task_struct declaration
UML: add kernel.h include to signal.c
...
The C99 specification states in section 6.11.5:
The placement of a storage-class specifier other than at the beginning
of the declaration specifiers in a declaration is an obsolescent
feature.
Signed-off-by: Tobias Klauser <tklauser@distanz.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These pins are for decoupling capacitors for the internal charge pumps
in TPA6130A2 and TPA6140A2 and not for connecting external supply.
Thanks to Eduardo Valentin <eduardo.valentin@nokia.com> for pointing out the
issue with TPA6130A2 and Ilkka Koskinen <ilkka.koskinen@nokia.com> with
TPA6140A2.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Reviewed-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (224 commits)
ARM: remove 'select GENERIC_TIME'
ARM: 6136/1: ARCH_REQUIRE_GPIOLIB selects GENERIC_GPIO
ARM: 6074/1: oprofile: convert from sysdev to platform device
ARM: 6073/1: oprofile: remove old files and update KConfig
ARM: 6072/1: oprofile: use perf-events framework as backend
ARM: 6071/1: perf-events: allow modules to query the number of hardware counters
ARM: 6070/1: perf-events: add support for xscale PMUs
ARM: 6069/1: perf-events: use numeric ID to identify PMU
ARM: 6064/1: pmu: register IRQs at runtime
ARM: Optionally allow ARMv6 to use 'normal, bufferable' memory for DMA
ARM: 6134/1: Handle instruction cache maintenance fault properly
ARM: nwfpe: allow debugging output to be configured at runtime
ARM: rename mach_cpu_disable() to platform_cpu_disable()
ARM: 6132/1: PL330: Add common core driver
ARM: 6094/1: Extend cache-l2x0 to support the 16-way PL310
ARM: Move memory mapping into mmu.c
ARM: Ensure meminfo is sorted prior to sanity_check_meminfo
ARM: Remove useless linux/bootmem.h includes
ARM: convert /proc/cpu/aligment to seq_file
arm: use asm-generic/scatterlist.h
...
Codec output pin should be defined with SND_SOC_DAPM_OUTPUT as otherwise
external widgets doesn't alter the output state.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add support for NUC900 AC97
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add control to enable earphone driver in TWL6040 codec. This driver
is connected to HSDAC Left.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The following structure elements duplicate the information in
'struct device.of_node' and so are being eliminated. This patch
makes all readers of these elements use device.of_node instead.
(struct of_device *)->node
(struct dev_archdata *)->prom_node (sparc)
(struct dev_archdata *)->of_node (powerpc & microblaze)
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
* 'core-locking-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip:
lockdep: Reduce stack_trace usage
lockdep: No need to disable preemption in debug atomic ops
lockdep: Actually _dec_ in debug_atomic_dec
lockdep: Provide off case for redundant_hardirqs_on increment
lockdep: Simplify debug atomic ops
lockdep: Fix redundant_hardirqs_on incremented with irqs enabled
lockstat: Make lockstat counting per cpu
i8253: Convert i8253_lock to raw_spinlock
This adds an argument to the DMAengine control function, so that
we can later provide control commands that need some external data
passed in through an argument akin to the ioctl() operation
prototype.
[dan.j.williams@intel.com: fix up some missed conversions]
Signed-off-by: Linus Walleij <linus.walleij@stericsson.com>
Signed-off-by: Dan Williams <dan.j.williams@intel.com>
Avoid calling the dac33_hard_power when the codec was
already in BIAS_OFF state.
This could happen in device suspend and module removal
time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the cases when the same power state would be set again
handled gracefully, we do not need to use dev_warn.
Signed-off-by: Felipe Balbi <felipe.balbi@nokia.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Use kzalloc rather than the combination of kmalloc and memset.
The semantic patch that makes this change is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x,size,flags;
statement S;
@@
-x = kmalloc(size,flags);
+x = kzalloc(size,flags);
if (x == NULL) S
-memset(x, 0, size);
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The number of HDMI nodes is expected to go up in future.
So don't fail hard on seeing extra converter/pin nodes.
We can still operate safely on the nodes within
MAX_HDMI_CVTS/MAX_HDMI_PINS.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the full chipset codename as codec name.
They are more user friendly than the spec abbrs.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is necessary to support >=3 HDMI playback devices
starting from the CougarPoint codec.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The entity_type_to_size[] array has LAST_ENTITY_TYPE (11) number of elements,
not LAST_ENTITY_ROLE (17). This only affects the debug output.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Smatch complains that if (dev == SNDRV_CARDS) we're one past the end of
the array. That's unlikely to happen in real life, I suppose.
Also smatch complains about "strcpy(card->shortname, pcm->name);"
The "pcm->name" buffer is 80 characters and "card->shortname" is 32
characters. If you follow the call paths it turns out we never actually
use more than 16 characters so it's not a problem. But anyway, let's
make it easy for people auditing this in the future.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is adding a new control which has the following capabilities:
- tlv
- variable data size (for instance, 7 ou 8 bit)
- double mixer
- data range centered around 0
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patchs should allow to use 32-bit samples on e.g. TLV320AIC3x codec,
or others.
Signed-off-by: Sergey Lapin <slapin@ossfans.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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>
The codec has support for swapping the left and right
channels in the digimic interface.
New kcontrol to handle this bit.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
MIPS non-coherent archs need the noncached pgprot in mmap of PCM buffers.
But, since the coherency needs to be checked dynamically via
plat_device_is_coherent(), we need an ugly check dependent on MIPS
in ALSA core code.
This should be cleaned up in MIPS arch side (e.g. creating
dma_mmap_coherent()) in near future.
Tested-by: Wu Zhangjin <wuzhangjin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 65c3ac885c in 2.6.33 accidentally
left out the initialization of the AC97 codec FMIC2MIC bit, which broke
recording from the front panel microphone.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that HP dv series have inconsistent the mute-LED GPIO
mapping among various models. dv4/7 seem to use GPIO 0 while dv 5/6
seem to use GPIO 3. The previous commit
26ebe0a289
ALSA: hda - Fix mute-LED GPIO pin for HP dv series
breaks dv5/6.
This patch adds the new quirk model, hp-dv4, to handle HP dv4/7
separately from HP dv5/6.
Tested-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com> (for dv6-1110ax)
Acked-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For both UAC1 and UAC2, interrupt endpoint messages are now parsed with
structs rather that with anonymous buffer array accesses.
For UAC2, only CUR interrupt notifications are supported for now.
snd_usb_mixer_status_complete() was renamed to
snd_usb_mixer_interrupt().
Fixed one indentation flaw on the way.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to prevent code ambiguous, add namespace on functions in ssp driver.
Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
PXA_SSP is actually used by drivers like drivers/spi/pxa2xx_spi.c and
sound/soc/pxa/pxa-ssp.c, not by boards. Remove those incorrect 'select'
from Kconfig and make SOC_PXA_SSP to select.
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
If the FLL is not configured attempting to resume it will produce a
warning message so skip the resume.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Disable the output stage prior to the delay stage rather than the
other way around. Fixes merge issue with previous headphone output
path corrections.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Log the values we're getting back from the DC servo and the values we
write to it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add back the zero return value (activate workqueue) when
bdl_pos_adj is nonzero for position check.
Do the position related check only for first next period
using wallclk counter.
Return -1 value (ignore interrupt) when period_bytes
variable is zero.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
If the register for the volume needs invert, than the inversion
need to be done from the chip maximum, and not from the platform
dependent limit.
Introduce soc_mixer_control.platform_max value, which initially
equals to chip maximum.
The snd_soc_limit_volume function only modify the platform_max,
all volsw_info call returns this as well.
The .max value holds the chip default (maximum), and it is used
for the inversion, if it is needed.
Additional check in the volsw_info call has been added to check
the validity of the platform_max in case, when custom macros
used by codec drivers are not initializing it correctly.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use 24Mhz WALLCLK register to ignore too early interrupts and
wrong interrupt status. The bad timing confuses the higher ALSA
layer and causes audio skipping. More information about behaviour
and debugging can be found in kernel bz#15912.
https://bugzilla.kernel.org/show_bug.cgi?id=15912
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Old HP dv series seem to use the GPIO pin 0 for controlling the mute LED
although the pin is a large package, where the newer models use GPIO 3
in such a case. For fixing the regression from the previous kernels,
set spec->gpio_led statically for these model quirks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HDA controller in US15W (Poulsbo) reports inaccurate position values
for capture streams when using the LPIB read method, resulting in
distorted recordings.
However, using the position buffer is broken for playback streams,
resulting in a fallback to the LPIB method with the current driver.
This patch works around the issue by independently detecting the read
position method for capture and playback streams.
The patch will not have any effect if the position fix method is
explicitly set.
[Code simplified by tiwai]
Signed-off-by: Shahin Ghazinouri <shahin.ghazinouri@pelagicore.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reference: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-May/027525.html
As reported on the mailing list, we also need to cap to the 0 dB offset
for Lenovo models, else the sound will be distorted.
Reported-and-Tested-by: Tim Starling <tstarling@wikimedia.org>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the string based list management to a handle base
implementation to help with the hot path use of pm-qos, it also renames
much of the API to use "request" as opposed to "requirement" that was
used in the initial implementation. I did this because request more
accurately represents what it actually does.
Also, I added a string based ABI for users wanting to use a string
interface. So if the user writes 0xDDDDDDDD formatted hex it will be
accepted by the interface. (someone asked me for it and I don't think
it hurts anything.)
This patch updates some documentation input I got from Randy.
Signed-off-by: markgross <mgross@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
This reverts commit 7aee674665.
As it doesn't seem to be universally valid for all mainboard revisions of
the D945GCLF2 and breaks snd-hda-intel/ snd-hda-codec-realtek on the Intel
Corporation "D945GCLF2" (LF94510J.86A.0229.2009.0729.0209) mainboard.
00:1b.0 Audio device [0403]: Intel Corporation N10/ICH 7 Family High Definition Audio Controller [8086:27d8] (rev 01)
Signed-off-by: Stefan Lippers-Hollmann <s.l-h@gmx.de>
Cc: <stable@kernel.org> [2.6.33]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables the SPDIF output pin by default. It also enables
it for quirks related to Levono docking stations (x200 and 25041,
identified with the same 17aa:20f2 ID). Even though not all Lenovo
docking stations have SPDIF connectors, enabling the pin by default
shouldn't be a problem for anyone.
Other quirks remain unmodified.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows more flexible integration with subsystem features.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
As well as allowing DAPM pins to be marked as ignoring suspend allow DAI
links to be similarly marked. This is primarily intended for digital
links between CODECs and non-CPU devices such as basebands in mobile
phones and will suppress all suspend calls for the DAI link. It is
likely that this will need to be revisited if used with devices which
are part of the SoC CPU.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some devices can usefully run audio while the Linux system is suspended.
One of the most common examples is smartphone systems, which are normally
designed to allow audio to be run between the baseband and the CODEC
without passing through the CPU and so can suspend the CPU when on a
voice call for additional power savings.
Support such systems by providing an API snd_soc_dapm_ignore_suspend().
This can be used to mark DAPM endpoints as not being sensitive to
system suspend. When the system is being suspended paths between
endpoints which are marked as ignoring suspend will be kept active.
Both source and sink must be marked, and there must already be an
active path between the two endpoints prior to suspend.
When paths are active over suspend the bias management will hold the
device bias in the ON state. This is used to avoid suspending the
CODEC while it is still in use.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of using stream events to handle power down during suspend
integrate the handling with the normal widget path checking by
replacing all cases where we report a connected endpoint in a path
with a function snd_soc_dapm_suspend_check() which looks at the ALSA
power state for the card and reports false if we are in a D3 state.
Since the core moves us into D3 prior to initating the suspend all
power checks during suspend will cause the widgets to be powered
down. In order to ensure that widgets are powered up on resume set
the card to D2 at the start of resume handling (ALSA API calls
require D0 so we are still protected against userspace access).
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We now manage suspend within the main power analysis rather than by
flipping the state of widgets.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core will ensure that the device is in either STANDBY or OFF bias
before suspending, restoring the bias in the driver is unneeded. Some
drivers doing slightly more roundabout things have been left alone
for now.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ideapad quirks working for my ThinkPad X100e (microphone is not tested).
Signed-off-by: Andrej Gelenberg <andrej.gelenberg@udo.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dev_node_t was only used to transport some minor/major numbers
from the PCMCIA device drivers to deprecated userspace helpers.
However, only a few drivers made use of it, and the userspace
helpers are deprecated anyways. Therefore, get rid of dev_node_t .
As a first step, remove any usage of dev_node_t from drivers which
only wrote to this typedef/struct, but did not make use of it.
CC: linux-bluetooth@vger.kernel.org
CC: Harald Welte <laforge@gnumonks.org>
CC: linux-mtd@lists.infradead.org
CC: linux-wireless@vger.kernel.org
CC: netdev@vger.kernel.org
CC: linux-serial@vger.kernel.org
CC: alsa-devel@alsa-project.org
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
Instead of the old pcmcia_request_irq() interface, drivers may now
choose between:
- calling request_irq/free_irq directly. Use the IRQ from *p_dev->irq.
- use pcmcia_request_irq(p_dev, handler_t); the PCMCIA core will
clean up automatically on calls to pcmcia_disable_device() or
device ejection.
- drivers still not capable of IRQF_SHARED (or not telling us so) may
use the deprecated pcmcia_request_exclusive_irq() for the time
being; they might receive a shared IRQ nonetheless.
CC: linux-bluetooth@vger.kernel.org
CC: netdev@vger.kernel.org
CC: linux-wireless@vger.kernel.org
CC: linux-serial@vger.kernel.org
CC: alsa-devel@alsa-project.org
CC: linux-usb@vger.kernel.org
CC: linux-ide@vger.kernel.org
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
As it's only used there it makes no sense relying on pcmcia_request_irq().
CC: alsa-devel@alsa-project.org
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
The commit 24e4a1211f
ALSA: info - Use standard types for info callbacks
introduced a wrong type to snd_opl4_mem_proc_write() for pos argument.
Fixed now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ESS ES968 chip is nothing more then a PnP companion
for a non-PnP audio chip. It was paired with non-PnP ESS' chips:
ES688 and ES1688. The ESS' audio chips are handled by the es1688
driver in native mode. The PnP cards are handled by the ES968
driver in SB compatible mode.
Move the ES968 chip handling to the es1688 driver so the driver
can handle both PnP and non-PnP cards. The es968 is removed.
Also, a new PnP id is added for the card I acquired (the change
was tested on this card).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allocate the snd_es1688 during the snd_card allocation.
This allows to remove the card pointer from the snd_es1688 structure.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 23caaf19b ("ALSA: usb-mixer: Add support for Audio Class v2.0")
broke support for Class1 devices due to two faulty changes. This patch
fixes it.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-and-Tested-by: The Source <thesourcehim@gmail.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Switch the MACHINE driver to use IISv4 CPU dai.
Remove BROKEN dependency now that we have proper CPU driver available.
Also, disable build for SMDK6400, since the S3C6400 doesn't have IISv4
controller.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the CPU driver for the IISv4 block found on S3C6410.
For now, the driver is almost a copy of s3c64xx-i2s.c but
it should diverge as more IISv4 specific stuff is added.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the functions arre only used for volume register,
change their name, and also fix them to properly
handle the cases, when via soc core the volume is
limited.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This reverts commit 6f3991152f.
Since core has now support for limiting the volume on controls this
patch is not needed. Furthermore, this patch actually prevents the core
to set new volume on the TPA.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the core to limit the maximum volume on an
existing control.
The function will modify the soc_mixer_control.max value
of the given control.
The new value must be lower than the original one (chip maximum)
If there is a need for limiting a gain on a given control,
than machine drivers can do the following in their
snd_soc_dai_link.init function:
snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21);
This will modify the original 31 (chip maximum) to 21, so user
space will not be able to set the gain higher than this.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Do not change the codec defaults for the following registers:
0x40, 0x41: Line output gains, do not use amplification
0x42: LOM/LOP Voltage hold, and selection
0x44: LOM inversion control
It has been found, that the values configured to these registers
can cause amplification, which can make the output of DAC33
distorted.
The codec reset values are considered safe in all environmnts.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add support for platform dependent gain limiting on the
tpa6130a2 (and tpa6140a2) Headset amplifier.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Handle the reset GPIO within the codec driver in order to follow
the startup protocol for the tlv320aic3x codecs.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for integrated stereo speakers and digital
microphone found on Nokia RX-51 hardware. This is a cut down version based
on Maemo kernel sources and earlier patchset by Eduardo Valentin et al.
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.html
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Eduardo Valentin <eduardo.valentin@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
We had a fixed external amp setup enabled for ALC888, but this seems
unnecessary. The amps are controlled rather by GPIOs.
Let's remove it now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda: Fix 0 dB for Packard Bell models using Conexant CX20549 (Venice)
ALSA: hda - Add quirk for Dell Inspiron 19T using a Conexant CX20582
ALSA: take tu->qlock with irqs disabled
ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite P500-PSPGSC-01800T
ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite Pro T130-15F
ALSA: hda - fix array indexing while creating inputs for Cirrus codecs
ALSA: es968: fix wrong PnP dma index
Now that we can specify feature of a particular controller, we can
avoid multiple copies of same code by defining the CDCLKCON bit
feature in controller specific code and detecting that flag in the
code common to all controllers.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to make s3c-i2s-v2.c manage controllers with minor
quirks and variation in features, we define a per-block flag
that indicates the availability/lack of a particular feature
to the s3c-i2s-v2.c
While adding support for new SoCs' I2S, check for the blocks
of older SoCs that have similar feature and set the flag for
that feature.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that the fields are defined for s3c2412, use them and avoid having
multiple copies of same defines.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we have two callbacks s3c2412_i2s_get_clock & s3c64xx_i2s_get_clock
doing exactly the same thing, we can define one generic s3c_i2sv2_get_clock
and discard other two copies. Also, switch the users to make calls to the
newly defined and generic s3c_i2sv2_get_clock
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No need to keep redundant field iis_clk in s3c_i2sv2_info.
iis_cclk and iis_pclk is all we need.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Until now, s3c2412_get_iisclk would return NULL since iis_clk was never
initialized.
Return appropriate pointer as per the selection made for source clock.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The IMS field of s3c2412/13 is essentially the same as that of s3c64xx.
That is, the IISMOD[11] bit decides Master/Slave mode and IISMOD[10] bit
selects source clock for signal generation.
For that reason, remove improper defines for IISMOD[11:10] field mask
and define two 1bit fields that can be set independent of each other.
As a consequence, corresponding fields for PLAT_S3C64XX too get to use
these new defines.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Define more bit definitions in the order of mainline
support for the SoC.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The header for I2Sv2
linux/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
contains only controller specific definitions and nothing
SoC specific. So, it could be moved to sound/soc/s3c24xx/
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: https://launchpad.net/bugs/541802
The OR's hardware distorts at PCM 100% because it does not correspond to
0 dB. Fix this in patch_cxt5045() for all Packard Bell models.
Reported-by: Valombre
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk for all-in-one computer Dell Inspiron One 19 Touch to have proper
HP and Mic support.
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should disable irqs when we take the tu->qlock because it is used in
the irq handler. The only place that doesn't is
snd_timer_user_ccallback(). Most of the time snd_timer_user_ccallback()
is called with interrupts disabled but the the first ti->ccallback()
call in snd_timer_notify1() has interrupts enabled.
This was caught by lockdep which generates the following message:
> =================================
> [ INFO: inconsistent lock state ]
> 2.6.34-rc5 #5
> ---------------------------------
> inconsistent {HARDIRQ-ON-W} -> {IN-HARDIRQ-W} usage.
> dolphin/4003 [HC1[1]:SC0[0]:HE0:SE1] takes:
> (&(&tu->qlock)->rlock){?.+...}, at: [<f84ec472>] snd_timer_user_tinterrupt+0x28/0x132 [snd_timer]
> {HARDIRQ-ON-W} state was registered at:
> [<c1048de9>] __lock_acquire+0x654/0x1482
> [<c1049c73>] lock_acquire+0x5c/0x73
> [<c125ac3e>] _raw_spin_lock+0x25/0x34
> [<f84ec370>] snd_timer_user_ccallback+0x55/0x95 [snd_timer]
> [<f84ecc4b>] snd_timer_notify1+0x53/0xca [snd_timer]
Reported-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/549267
The OR verified that using the olpc-xo-1_5 model quirk allows the
headphones to be audible when inserted into the jack. Capture was
also verified to work correctly.
Reported-by: Richard Gagne
Tested-by: Richard Gagne
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/573284
The OR verified that using the olpc-xo-1_5 model quirk allows the
headphones to be audible when inserted into the jack. Capture was
also verified to work correctly.
Reported-by: Andy Couldrake <acouldrake@googlemail.com>
Tested-by: Andy Couldrake <acouldrake@googlemail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fixes a problem where cards show up as only having a single mixer
element, suppressing all sound output.
Signed-off-by: Brian J. Tarricone <brian@tarricone.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both tpa6130a2, and tpa6140a2 is supported by the
same driver, but the gain dB scaling is different on
the amplifiers.
Provide different mixer control for the chips with correct
TLV mapping.
User space will see:
"TPA6130A2 Headphone Playback Volume" in case of 6130
"TPA6140A2 Headphone Playback Volume" in case of 6140
The way machine drivers are using this amplifier remained
the same.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Let the codec to hit OFF instead of STANDBY, when there is no activity.
When the codec is off, than the associated regulator can be also turned
off (if the number of users on the regulator is 0).
After initialization, the codec remains in power off, it is only turned
on for reading the ID registers (also testing the regulators).
The codec power is enabled, when the codec is moving from BIAS_OFF
to BIAS_STANDBY.
The codec is turned off, when it hits BIAS_OFF.
There are few scenarios, which has to be taken care::
1. Analog bypass caused BIAS_OFF -> BIAS_ON
We need to power on the codec, and do the chip init, but we does not
need to execute the playback related configuration
2. Playback caused BIAS_OFF -> BIAS_ON
We need to power on the codec, and do the chip init, and also we need
to execute the playback related configuration.
3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON)
We need to execute the playback related configuration. The codec is
already on.
4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON)
Nothing need to be done.
5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON)
We need to execute the playback related configuration. The codec is
still on.
Since the power up, and the codec init is optimized, the added overhead
in stream start is minimal.
Withing this patch, the hard_power function is now only doing what it
supposed to: only handle the powers, and GPIO reset line.
The codec initialization and state restore has been moved out.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
As a preparation for supporting codec to be turned off,
when we are in BIAS_STANDBY.
The substream must be easily available in other places than
pcm_* callbacks.
Manage a pointer in _startup, and _shutdown for this.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Optimize the way how tlv320dac33 is powered uppon module and
soc initialization.
Also read the DAC33 ID registers, and update the reg_cache
to reflect it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
On power up we only need to initialize the codec, and
restore only registers, which are not in either in DAPM
nor in the playback start sequence.
These are mostly gain related registers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
OUTL/R are leftovers from the original driver, and they
are no longer needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch orders the APLL and AIF power sequence in
case of HiFi (audio in TWL4030 terms) playback/capture.
We also need to make sure that the AIF is running during
playback/capture, when there is no valid DAPM route
available. For this purpose I introduce these virtual
widgets:
/* To have complete playback route all the time */
DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */
/* To have complete capture route all the time */
DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */
/* To have complete playback route for the voice module */
DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */
The DAPM_SUPPLY widgets for APLL and AIF are placed in a way,
that during any audio activity the needed configuration of AIF
and APLL will be enabled (playback, capture, analog loopback,
digital loopback, and voice activity).
The apll reference counting code has been lifted,
and modified from Liam Girdwood's earlier patch.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This enables autoloading of the TXx9 sound driver on RBTX4927.
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
To: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Cc: Linux MIPS Mailing List <linux-mips@linux-mips.org>
Patchwork: http://patchwork.linux-mips.org/patch/1101/
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
The WM9090 is a high performance low power audio subsystem, including
headphone and class D speaker drivers.
Note that this driver is a standalone CODEC driver and so is only
immediately suitable for use with the WM9090 as a standalone sound card
taking line inputs, or with a DAC with no software control. The pending
ASoC multi-CODEC support will expand the range of systems that can use
the driver, or system-specific adaptations can be made.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The SYSCLK source is automatically managed when configuring the PLL.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the TLV320AIC3x supplies and enables all of them for the
entire lifetime of the device.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Move PLL enable from BIAS_ON state to BIAS_PREPARE to be pair with
BIAS_STANDBY where PLL is disabled. Remove also old comments about power
control.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
These ADC, DAC and output pin power off commands are needless in
aic3x_set_bias_level since they are not enabled in aic3x_init and they are
defined in aic3x_dapm_widgets so the ASoC DAPM will take care of them
anyway.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for Philips UDA1345 CODEC. The CODEC has only
volume control, de-emphasis, mute, DC filtering and power control features.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Anti Sullin <anti.sullin@artecdesign.ee>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Delay reporting for the three implemented DAC33 FIFO modes.
DAC33 has FIFO depth status register(s), but it can not be used, since
inside of pcm_pointer we can not send I2C commands.
Timestamp based estimation need to be used. The method of calculating
the delay depends on the active FIFO mode.
Bypass mode: FIFO is bypassed, report 0 as delay
Mode1: nSample fill mode. In this mode I need to use two timestamp
ts1: taken when the interrupt has been received
ts2: taken before writing to nSample register.
Interrupts are coming when DAC33 FIFO depth goes under alarm threshold.
Phase1: when we received the alarm threshold, but our workqueue has
not been executed (safeguard phase). Just count the played out
samples since ts1 and subtract it from the alarm threshold
value.
Phase2: During nSample burst (after writing to nSample register), count
the played out samples since ts1, count the samples received
since ts2 (in a burst). Estimate the FIFO depth using these and
alarm threshold value.
Phase3: Draining phase (after the burst read), count the played out
samples since ts1. Estimate the FIFO depth using the nSample
configuration and the alarm threshold value.
Mode7: Threshold based fill mode. In this mode one timestamp is enough.
ts1: taken when the interrupt has been received
Interrupts are coming when DAC33 FIFO depth reaches upper threshold.
Phase1: Draining phase (after the burst), counting the played out
samples since ts1, and subtract it from the upper threshold
value.
Phase2: During burst operation. Using the pre calculated time needed to
play out samples from the buffer during the drain period (from
upper to lower threshold), move the time window to cover the
estimated time from the burst start to the current time.
Calculate the samples played out since lower threshold and also
the samples received during the same time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
When the DAC33 FIFO is in use the dai interface is running in
much higher speed than the sampling frequency.
Calculate the rate based on the internal base frequency and
the bclk divider.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Upper and Lower threshold values are used as magic
numbers. Replace them with defines for later use.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
There is no need for calculations for FIFO bypass mode.
Just in case set the nsample maximum limit, which
has been done in the calculation phase.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Alarm threshold interrupt is triggered right after the
playback start.
This interrupt is recieved during the first burst period,
and caused the state machine to write additional nSample
command, which has to be avoided.
To fix this issue move the DAC33 interrupt unmasking
after we configured the PREFILL register with a small
delay.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
There is only one dma for the ESS ES968 based board.
Its index is 0 and not 1.
This make the es968 card working.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Follow the core jack implementation and allow reporting on the status
of NULL jacks, avoiding the need to check in detection implementations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The hardware volume handling code in essence just detects key presses, and
then does some hardcoded modification of the master volume based on which key
is pressed.
Clearly the right thing to do here is just report these keypresses to
userspace and let userspace decide what to with them.
This patch adds a Kconfig option which when enabled reports the volume
buttons as keypresses using an input device. When enabled this option
also gets rid of the ugly direct ac97 writes from the tasklet, the ac97lock
and the need for using a tasklet in general.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While working on the sound suspend / resume problems with my laptop
I noticed that the hardware volume handling code in essence just detects
key presses, and then does some hardcoded modification of the master volume
based on which key is pressed.
This made me think that clearly the right thing to do here is just report
these keypresses to userspace and let userspace decide what to with them.
This patch adds a Kconfig option which when enabled reports the volume
buttons as keypresses using an input device. When enabled this option
also gets rid of the ugly direct ac97 writes from the tasklet, the ac97lock
and the need for using a tasklet in general.
As an added bonus the keys now work identical to volume keys on a (usb)
keyboard with multimedia keys, providing visual feedback of the volume
level change, and a better range of the volume control (with a properly
configured desktop environment).
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/568600
The OR has verified that the dell-m6 model quirk is necessary for audio
to be audible by default on the Dell Studio XPS 1645.
This change is necessary for 2.6.32.11 and 2.6.33.2 alike.
Reported-by: Andy Ross <andy@plausible.org>
Tested-by: Andy Ross <andy@plausible.org>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/459083
The OR has verified with 2.6.32.11 and the latest alsa-driver stable
daily snapshot that position_fix=1 is necessary for the external mic
to work and for PulseAudio not to crash constantly.
This patch is necessary also for 2.6.32.11 and 2.6.33.2.
Reported-by: <imwithid@yahoo.com>
Tested-by: <imwithid@yahoo.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While working on a fix for the volume being muted on the allegro in my
Compaq EVO N600C after suspend, I've learned a few things about the hardware
volume control worth documenting. The actual fix for the suspend / resume
issue is in the next patch in this set.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ignore spurious HV interrupts during suspend / resume, this avoids
mistaking them for a mute button press. This is not very pretty but
it seems the only way to fix the master volume control gets muted
after suspend issue I'm seeing. Note that the es1968 driver is doing
exactly the same.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this quirk sound stops working after suspend resume. With this quirk,
one still needs to manually unmute the master volume control after a suspend /
/ resume cycle. That is fixed in another patch in this set.
Note that this patch was submitted to the alsa bug tracker a long time ago:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4319
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/567494
The OR has verified that the existing model quirk, ALC880_UNIWILL,
is insufficient for audible playback and capture by default. Instead,
the ALC880_F1734 model quirk needs to be used.
This change is necessary for both 2.6.32.11 and 2.6.33.2.
Reported-by: Arnaud Malpeyre <amalpeyre@gmail.com>
Tested-by: Arnaud Malpeyre <amalpeyre@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/553002
The OR has verified that the dell-m6 model quirk is necessary for audio
to be audible by default on the Dell Studio XPS 1645.
This change is necessary for 2.6.32.11 and 2.6.33.2 alike.
Reported-by: Robert Chambers
Tested-by: Robert Chambers
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of AudioScience ASI boards.
The driver has been tested for years on alsa-driver external tree,
now finally got merged to the kernel.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use all the available Fratio values when configuring the WM8994 FLL, not
just 0 and 3, following more complete characterisation of the device
performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8994 FLL can be clocked from one of four inputs, the two MCLKs and
the LRCLK and BCLK of the AIF associated with the FLL. Allow all four
inputs to be used rather than defaulting to MCLK1.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
An index equal to the array size may not be accessed.
Signed-off-by: Phil Carmody <ext-phil.2.carmody@nokia.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When HP laptop with AD1981 codec is suspended and the docking-station
is connected before the resume, the outputs get confused, and wrongly
routed still to the speaker. This is because of a change in 2.6.34-rc1
ea52bf260e
ALSA: hda: Add powerdown for Analog Devices HDA codecs
The problem was the added resume callback that doesn't consider the
modified init hook. The fix is simply remove the resume callback here
and make the resume normally. This doesn't change any behavior intended
in the commit above (for shutting down the sound at suspend) but only
fixes the resume.
Reported-and-tested-by: Frans Pop <elendil@planet.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.
To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Doing so causes a deadlock, so just signal the timer to stop
using an atomic variable.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixed 8 checkpatch errors (ERROR: do not use assignment in if condition)
in sound/i2c/i2c.c.
Signed-off-by: Brian Waters <brianmwaters@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
attached please find a patch that adds support for at least the T410s
HDA codec. Most likely it will also add support for the T410 and T510
based models.
The patch was derived from Ideapad support. Support for the laptop's and
docking-station output connectors as well as the docking-station microphone
connector and the laptops internal devices has been tested. Since it has been
developed without a data-sheet available, support for digital outputs and the
laptop's microphone input may well be incorrect.
Microphone mute functionality is not included:
The microphone mute button seems to be reported through thinkpad_acpi key
0000101b. The mute button LED seems to be wired to thinkpad_acpi led
number 15.
Signed-off-by: Jens Taprogge <jens.taprogge@taprogge.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Biostar mobo seems to give a wrong DMA position, resulting in
stuttering or skipping sounds on 2.6.34. Since the commit
7b3a177b0d, "ALSA: pcm_lib: fix "something
must be really wrong" condition", makes the position check more strictly,
the DMA position problem is revealed more clearly now.
The fix is to use only LPIB for obtaining the position, i.e. passing
position_fix=1. This patch adds a static quirk to achieve it as default.
Reported-by: Frank Griffin <ftg@roadrunner.com>
Cc: Eric Piel <Eric.Piel@tremplin-utc.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the matching model for Clevo laptop M570U.
Signed-off-by: Joerg Schirottke <master@kanotix.com>
Tested-by: Maximilian Gerhard <maxbox@directbox.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the notification of elapsed periods is not very exact.
Increase minimum periods to 4 as suggested by Liam Girdwood.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some VIA codecs have no multiple source selection for headphone pins,
thus it's useless (and wrong) to create "Independent HP" control on them.
This patch adds the check of connections to skip the control in such a
case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 5b0cb1d850
ALSA: hda - add more NID->Control mapping
breaks the control element allocation by returning a wrong value.
Let's fix it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set no_llseek to llseek file ops of each sound component (but for hwdep).
This avoids the implicit BKL invocation via generic_file_llseek() used
as default when fops.llseek is NULL.
Also call nonseekable_open() at each open ops to ensure the file flags
have no seek bit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The llseek implementation is identical for existing driver implementations,
so let's merge to the common layer. The same code for the text proc file
can be used even for the binary proc file.
The driver can provide its own llseek method if needed. Then the common
code will be skipped.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the validity of the file position in the common info layer before
calling read or write callbacks in assumption that entry->size is set up
properly to indicate the max file size.
Removed the redundant checks from the callbacks as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 29a4f2d3 used writel() at offset 0x26 which is
half-word aligned causing unaligned exceptions on a
Cortex-A8. The original patch solved the "aaci-pl041 fpga:04:
ac97 read back fail" issue on a soft reset. Reading from any
arbitrary aaci register seems to solve this issue.
Signed-off-by: Philby John <pjohn@mvista.com>
Acked-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for sound through the WM8750 codec on Zipit Z2.
Also, this patch incorporates support for detecting headset jack
insertion through the jack detection API.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Bill Gatliff <bgat@billgatliff.com>
Acked-by: Richard Purdie <rpurdie@rpsys.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sony VAIO models with ALC269 need to initialize the pin 0x19 to VREF
ground or Hi-Z to make the headphone working. Other than that, model=auto
works fine, so let's use model=auto with a specific fix-up table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few enhancement / fixes for fix-up table of some Realtek codecs:
- Apply fix-ups only for the auto model
- Apply additional verbs after normal init verbs
- Add a debug print to show the fix-up application
This is basically a preliminary work for the next fix for Sony VAIO.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb-midi causes sometimes Oops at snd_usbmidi_output_drain() after
disconnection. This is due to the access to the endpoints which have
been already released at disconnection while the files are still alive.
This patch fixes the problem by checking disconnection state at
snd_usbmidi_output_drain() and by releasing urbs but keeping the
endpoint instances until really all freed.
Tested-by: Tvrtko Ursulin <tvrtko@ursulin.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The widget connections of ADC of ALC880 and ALC2260 aren't initialized,
thus it might point to invalid pin. This can be a problem when mode=auto
and there is only one input pin. Then user can't change the connection
at all.
This patch adds the code to initialize the input pin connection of these
codecs.
Reference: Novell bnc#594363
https://bugzilla.novell.com/show_bug.cgi?id=594363
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code
around. Hugely inspired by WM8753 which was already converted.
Also, this patch fixes the Jive and Spitz machine.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Corrected HP and mic pins for ALC269vb amic and dmic models.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269vb has an alternative HP pin 0x21 in addition.
Fix the parser to recognize it.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a local mutex instead of BKL. This should suffice since each device
type has also its open_mutex.
Also, a bit of clean-up of the legacy device auto-loading code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using a regular timer results in poll times < 1 jiffie with small
buffers, so we loaded the timer with the actual jiffie value. We can
be more accurate using a hrtimer. Also, we have to call
snd_pcm_period_elapsed after playing period_bytes and not
runtime->period_size (which is in samples and not in bytes).
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When checking if we are DMA capable we have to check for the
IMX_SSI_DMA flag which is already set from platform_data instead
of setting it again when we want to do DMA.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@Slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: mixart: range checking proc file
ALSA: hda - Fix a wrong array range check in patch_realtek.c
ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
ALSA: hda - Enable amplifiers on Acer Inspire 6530G
ASoC: Only do WM8994 bias off transition from standby
ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices
ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction
ASoC: Support second DC servo readback method for wm_hubs
ASoC: Avoid wraparound in wm_hubs DC servo correction
ALSA: echoaudio - Eliminate use after free
ALSA: i2c: cleanup: change parameter to pointer
ALSA: hda - Add MSI blacklist for Aopen MZ915-M
ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
ALSA: hda - Update document about MSI and interrupts
ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
ALSA: hda - Add missing printk argument in previous patch
ASoC: Fix passing platform_data to ac97 bus users and fix a leak
ALSA: hda - Fix ADC/MUX assignment of ALC269 codec
ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo()
ASoC: wm8994: playback => capture
This adds support for the Medion WIM2160 soundcard.
There's no PCI quirk added because it has the same PCI id as the
Medion MD2.
Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Assign DACs properly to each output. Currently, the front output is bound
to HP/speaker outputs blindly, but they should be assigned to individual
DACs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The original code doesn't take into consideration that the value of
MIXART_BA0_SIZE - pos can be less than zero which would lead to a large
unsigned value for "count".
Also I moved the check that read size is a multiple of 4 bytes below
the code that adjusts "count".
Signed-off-by: Dan Carpenter <error27@gmail.com>
Cc: <stable@kernel.org>
Acked-by: Linus Torvalds <torvalds@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 6a4f2ccb46 introduced a wrong
comparision for the array range check, which effectively skips the whole
initialization of DAC connections. Fixed now.
Reference: bko#15689
https://bugzilla.kernel.org/show_bug.cgi?id=15689
Reported-by: Adrian Ulrich <kernel@blinkenlights.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support interrupt based microphone bias detection. The WM8994 has two
microphone bias supplies, with detection supported on both. Detection
using GPIOs together with the standard GPIO based jack framework is
already supported via the platform data for the WM8994 core driver.
Note that as well as the microphone bias itself the system clock and
whichever AIF clock is supplying the system clock will need to be
enabled for detection to function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
[Note that this is a backported version for 2.6.34.
Upstream commit is fd23b7dee]
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After more tests it appears that EAPD needs to be enabled
on both the 0x14 and 0x15 NIDs to enable the main speaker
and headphone amplifiers. The maximum volume setting is
now equal to what the machine achieves under other operating
systems.
Disabling Front or LFE playback triggers EAPD and disables
the amplifier. As such, these two playback switches have
been removed from the mixer.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise we may try to power down multiple times when the using
idle bias off and the driver is removed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The DCS_DATAPATH_BUSY bit used to monitor the completion of DC servo
operations has been deprecated and with some more recente revisions
may perform incorrectly, especially when only analogue bypass paths
are in use. Switch to using readback from the DC servo command
register instead, which is supported for all devices. Without this
unacceptably long timeouts may be observed in some circumstances.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
If we need to offset correct the DC servo then don't use runtime
recalibration since that is likely to introduce further offsets
which will be evident on powerdown.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
More recent Wolfson hubs devices add the ability to read back the DC
servo calibration information from the register used to write offsets,
and later still ones remove the old readback registers. Add support
for the new scheme, and use it for WM8994 device revisions that
support it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
If the correction wraps around then a substantial offset would be
introduced.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Reverse headphone detection bit on PowerMac G4 Digital Audio (Tumbler).
Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the call to snd_card_free in the error handling code at the end of the
function, as in the other error cases.
A simplified version of the semantic patch that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression E,E2;
@@
snd_card_free(E)
...
(
E = E2
|
* E
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We actually pass an array of 7 chars not 5.
This silences a smatch warning.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device needs MSI disablement. Added to the quirk list.
Reported-by: Harald Dunkel <harri@afaics.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With recent (2.6.34) chnages in PCM handling, capture stopped working on my
OMAP1510 based Amstrad Delta videophone.
Using 2.6.34-rc2, I was able to correct the problem in 3 different ways:
1. reverting commit 7b3a177b0d,
2. enabling additional jiffies check with
echo 4 >/proc/asound/card0/pcm0c0/xrun_debug
3. applying the patch below.
Since I wasn't able to reproduce the problem on my i686 PC, I guess the
problem is probably machine specific.
The patch reuses the method for software emulation of missing hardware
pointer, already implemented for playback on OMAP1510. It's possible that
event if a hardware pointer is available for capture on this machine, its
behaviour may be not compatible with what upper layer expects.
If you think the problem may be more general and should be solved differently,
on a higher level, I can try to work more on it if you give me a hint.
If the patch gets accepted, I suggest it goes as a fix in the current release
cycle.
Created and tested against linux-2.6.34-rc2.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: https://launchpad.net/bugs/551606
The OR's hardware distorts at PCM 100% because it does not correspond to
0 dB. Fix this in patch_ad1981() for all models using the Thinkpad
quirk.
Reported-by: Jane Silber
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new helper, snd_hda_codec_update_cache(), for reducing the unneeded
verbs. This function checks the cached value and skips if it's identical
with the given one. Otherwise it works like snd_hda_codec_write_cache().
The alc269 code uses this function as an example.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some HP laptops have a mute LED that is controlled over the unused
MIC2 VREF pin. Implement the LED updater like patch_sigmatel.c for this
model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
The way i've factored out the bus probe and removal functions so
that there's no code in the individual I2C and SPI functions means
that the register() and unregister() functions could just be squashed
into the bus_probe() and bus_remove() functions.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
[The issue is an attempt to write the pdata without the AC97 device
allocated when using ac97.c - also added a comment in soc-core.c for the
special case for ac97. -- broonie]
Signed-off-by: Graham Gower <graham.gower@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Implicit slab.h inclusion via percpu.h is about to go away. Make sure
gfp.h or slab.h is included as necessary.
Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implicit slab.h inclusion via percpu.h is about to go away. Make sure
gfp.h or slab.h is included as necessary.
Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269 codec has a few different variants, and each of them may have
different ADC and MUX widgets. For example, one model has ADC 0x08
with MUX 0x23 while others has ADC 0x09 or ADC 0x07 with MUX 022 or
0x24. The difference of ADC appears usually as the capability of
the digital mic pin (0x12), and the current driver sometimes misses
the internal mic pin due to the mismatching ADC.
This patch adds a bit more clever way to find the matching ADC instead
of the static list. Now the driver checks all active input pins and
fills only the ADC/MUX's that contain all of them.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mask and value parameters passed to snd_hda_codec_amp_stereo()
should be 8-bit values for mute and volume. Passing AMP_IN_MUTE() is
wrong, which is found in many places in patch_realtek.c as a left-over
from the conversion to snd_hda_codec_amp_stereo().
Reported-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=575669
The OR states that position_fix=1 is necessary to work around glitching
during volume adjustments using PulseAudio.
Reported-by: Carlos Laviola <claviola@debian.org>
Tested-by: Carlos Laviola <claviola@debian.org>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/481058
The OR has verified that both 'Headphone Jack Sense' and 'Line Jack Sense'
need to be muted for sound to be audible, so just add the machine's SSID
to the ac97 jack sense blacklist.
Reported-by: Richard Gagne
Tested-by: Richard Gagne
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert the device_terminate_all() operation on the
DMA engine to a generic device_control() operation
which can now optionally support also pausing and
resuming DMA on a certain channel. Implemented for the
COH 901 318 DMAC as an example.
[dan.j.williams@intel.com: update for timberdale]
Signed-off-by: Linus Walleij <linus.walleij@stericsson.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Maciej Sosnowski <maciej.sosnowski@intel.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Pavel Machek <pavel@ucw.cz>
Cc: Li Yang <leoli@freescale.com>
Cc: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Cc: Paul Mundt <lethal@linux-sh.org>
Cc: Ralf Baechle <ralf@linux-mips.org>
Cc: Haavard Skinnemoen <haavard.skinnemoen@atmel.com>
Cc: Magnus Damm <damm@opensource.se>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: Joe Perches <joe@perches.com>
Cc: Roland Dreier <rdreier@cisco.com>
Signed-off-by: Dan Williams <dan.j.williams@intel.com>
The commit 4d96eb255c broke the interrupt
time xrun functionality (stream stop etc.) if the CONFIG_SND_PCM_XRUN_DEBUG
is not set. This is because the xrun() is null defined without it.
Fix this by letting the function xrun() to be always defined as it was
before.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ARM-SHMOBILE series have FIFO-buffered serial interface 2 (FSI2)
device which is advanced version of FSI.
This patch add simple support for it.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The probe_only module parameter skips the codec initialization, too.
Remove the model=hwio code and use second bit in probe_only to
skip the HDA codec reset procedure.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
For Lenovo Thinkpad T61/X61, the analog beep input is connected
to node 0x20, index 3. Move the digital beep mute/volume controls
as "Digital Beep" and create analog beep controls for mentioned node.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Using the 'model=hwio' option, the driver bypasses any codec
initialization and the reset procedure for codecs is also
bypassed. This mode is usefull to enable direct access using
hwdep interface (using hdaverb or hda-analyzer tools) and
retain codec setup from BIOS.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
BugLink: https://launchpad.net/bugs/303789
This model needs both 'Headphone Jack Sense' and 'Line Jack Sense'
muted for audible audio, so just add its SSID to the blacklist and
don't enumerate the controls.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add full duplex support on AT91 and AVR.
It was a bug: we needed to check first if there are some chips opened so we
could enable both reception and sending of the data.
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add AC97 support for ATMEL AT91, using the AVR32 code.
While AVR is using a DMA, the AT91 chips are using a Peripheral Data
Controller.
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Skip initialization of connections of DAC widgets that aren't used,
which resulted in invalid verb parameters.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds basic support for M-Audio's Fast Track Ultra series of USB
audio interfaces. It is a refactored version of the patch Clemens
Ladisch posted some time ago. Neither playback nor capturing work
properly at 44100 Hz (don't know why).
The other sampling rates work properly. There's no support for the DSP
mixer, yet.
Signed-off-by: Felix Homann <fexpop@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sparse caught that initialize "playback" two times instead of
initializing "capture".
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown
Daniel T Chen
Ben Collins
Marek Vasut
Linus Torvalds
Takashi Iwai
Mark Hills
Andreas Herrmann
Daniel Mack
Jassi Brar
Eliot Blennerhassett
Julia Lawall
Jerone Young
Stuart Longland
Guennadi Liakhovetski
Clemens Ladisch
Justin P. Mattock
H Hartley Sweeten
Grant Likely
Barry Song
Krzysztof Foltman
Jorge Eduardo Candelaria
Stephen Rothwell
Tobias Klauser
Jarkko Nikula
Wan ZongShun
Liam Girdwood
Misael Lopez Cruz
Dan Williams
Linus Walleij
Geert Uytterhoeven
Peter Ujfalusi
Felipe Balbi
Wu Fengguang
Dan Carpenter
apatard@mandriva.com
Sergey Lapin
Haojian Zhuang
Eric Miao
Jaroslav Kysela
Shahin Ghazinouri
Mark Gross
Stefan Lippers-Hollmann
Pierre-Louis Bossart
Andrej Gelenberg
Dominik Brodowski
Krzysztof Helt
Ville Syrjälä
Benjamin Herrenschmidt
Anisse Astier
Brian J. Tarricone
Ingo Molnar
Vladimir Zapolskiy
Hans de Goede
Jiri Kosina
Phil Carmody
Sascha Hauer
Brian Waters
Jens Taprogge
Joerg Schirottke
Philby John
Bill Gatliff
Kailang Yang
Huang Weiyi
Maurus Cuelenaere
d binderman
Tony Vroon
Risto Suominen
Janusz Krzysztofik
Tejun Heo
Graham Gower
Daniel Chen
Kuninori Morimoto
Sedji Gaouaou
Felix Homann