These are the final conversions for the ioctl file operation so we can remove
it in the next merge window.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This moves the lock_kernel() call from soundcore_open
to the individual OSS device drivers, where we can deal
with it one driver at a time if needed, or just kill
off the drivers.
All core components in ALSA already provide
adequate locking in their open()-functions
and do not require the big kernel lock, so
there is no need to add the BKL there.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correctly configure bidirectional pins when resuming; do not power down
widgets when they are needed for Smart5.1 output; and on 3-jack boards,
create the streams and controls needed for six channels.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Viliam Kubis <viliam.kubis@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As per-stream volume controls, the DXS controls are not intended to
adjust the overall sound level and so are initialized every time
a stream is opened. However, there are special situations where one
wants to reduce the overall volume in the digital domain, i.e., before
the AC'97 codec's PCM volume control. To allow this, add a module
parameter that sets the initial DXS volume.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Soeren D. Schulze <soeren.d.schulze@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It is not advisable to print a warning when a device does not support
setting the sample rate because this is perfectly valid for devices with
a single rate or where rates are implicitly changed by selecting another
alternate setting.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the amp max value at put callbacks and set the upper limit
so that the driver won't write any invalid value over the defined
range.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the beep mixer controls to Conexant codecs.
They simply control the digital beep generator widget.
For cx5047, I couldn't find any beep generator, so it's not implemented
there.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many codecs now clear the pin controls at suspend via snd_hda_shutup_pins()
for reducing the click noise at power-off. But this leaves some pins
uninitialized, and they'll be never recovered after resume.
This patch adds the proper recovery of cleared pin controls on resume.
Also it adds a check of bus->shutdown so that pins won't be cleared at
module unloading.
Reference: Kernel bug 16339
http://bugzilla.kernel.org/show_bug.cgi?id=16339
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8994 can output a clock derived from its internal SYSCLK, called
OPCLK. The rate can be selected as a sysclk, with a division from the
SYSCLK rate specified (multiplied by 10 since a division of 5.5 is
supported) and the clock can be disabled by specifying a divisor of
zero.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
i2s_accurate_sck switch can be used to have a better approximate
sampling frequency.
The clock is an externally visible bit clock and it is named
i2s continuous serial clock (I2S_SCK).
The trade off is between more accurate clock (fast clock)
and less accurate clock (slow clock).
The waveform will be not symmetric.
Probably it is possible to get a better algorithm for calculating
the divider, trying to keep a slower clock as possible.
This patch has been developed against the
http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
git tree and has been tested on bmx board (similar to dm365 evm, but using
uda1345 as external audio codec).
Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When McBSP peripheral gets the clock from an external pin,
there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR
and MCBSP_CLKS.
evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different
hardware connection and I use MCBSP_CLKS, so I have added
this possibility.
This patch has been developed against the:
http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
git tree and has been tested on bmx board (similar to dm365 evm)
Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added two clocking options for dm365 McBSP peripheral when used
with I2S timings, that are SND_SOC_DAIFMT_CBS_CFS (the cpu generates
clock and frame sync) and SND_SOC_DAIFMT_CBS_CFM (the cpu gets clock
from external pin and generates frame sync).
A slave clock management can be important when the external codec needs
the system clock and the bit clock synchronized (tested with uda1345).
This patch has been developed against the:
http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
git tree and has been tested on bmx board (similar to dm365 evm, but using
uda1345 as external audio codec).
Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The speaker was enabled when the headphone was plugged in, which isn't the
wanted behaviour so correct this.
Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Compander API changed to one function per parameter.
Factor out some common code for stereo log value reading.
Make some more entity functions static.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some deprecated items.
Change compander api to one function per parameter.
Add a version string define.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of this function is protected by the sound_loader_lock.
We can push down the BKL to this call out err = file->f_op->open(inode,file);
In order to build the sound core without the BKL, we
will need to push the lock_kernel() call into the ~20
device drivers that register their file operations.
Signed-off-by: John Kacur <jkacur@redhat.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Alan Cox <alan@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the PCI SSID gives an overriding SKU assno, PC-beep bit isn't
detected (since it's located over 16bit), resulting in no PC beep.
Also, many devices seem ignoring the requirement by Realtek's spec
for SSID numbers, and it also confuses the PC beep detection.
This patch assumes the PC beep is available on every machine with
PCI SSID override. It's a regression fix from 2.6.34.
Reference: Kernel bug 16251
http://bugzilla.kernel.org/show_bug.cgi?id=16251
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It will be replaced with automatic deemphasis rate configuration but since
we have an enumeration table in this driver this is done in a separate
commit to make the renumbering of the enumeration items clear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This adds sound support for the SmartQ board.
The hardware consists of a S3C6410 coupled with a WM8987 over I²S. The WM8750
driver is used for driving the WM8987, as they are register compatible.
Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8987 codec is register compatible with the WM8750, so just add it to the
SPI and I²C device table.
Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For UDA1341 codec power control is managed in STATUS1 register, and
for all other codecs in DATA011 register.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In UDA1340, UDA1344 and UDA1345 codecs there is one more functional
register in part of DATA0 tranfser. For UDA1341 this register
coincides with EA register.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch moves SPARC architecture specific data members out of
struct of_device and into the pdev_archdata structure. The reason
for this change is to unify the struct of_device definition amongst
all the architectures. It also remvoes the .sysdata, .slot, .portid
and .clock_freq properties because they aren't actually used by
anything.
A subsequent patch will replace struct of_device entirely with struct
platform_device and the of_platform support code will share common
routines with the platform bus (but the bus instances themselves can
remain separate).
This patch also adds 'struct resources *resource' and num_resources
to match the fields defined in struct platform_device. After this
change, 'struct platform_device' can be used as a drop-in replacement
for 'struct of_platform'.
This change is in preparation for merging the of_platform_bus_type
with the platform_bus_type.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: David S. Miller <davem@davemloft.net>
Cc: Stephen Rothwell <sfr@canb.auug.org.au>
A few boards using this controller are reported to need a little extra
time during their reset cycle.
Reported-by: Michael Goeke <michael.goeke@icachip.de>
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When using a timing voice to clock out periods during capture, the
driver would slowly loose synchronization and never catch up, eventually
reaching a point where it no longer generated interrupts. To avoid
this situation, the virtual period clocking was changed to shorten the
next timing period when our timing voice falls too far behind the
capture voice. In addition, the first virtual period for the timing
voice was slightly too short, causing the timing voice to initially be
ahead of the capture voice.
While tracking down this problem, I noticed that the expected sample
offset was being incorrectly initialized, causing an overrun to be
incorrectly reported when the timing voice happened to be perfectly
synchronized.
Reported-by: Hans Schou <linux@schou.dk>
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When using poll() to wait for the next period -- or avail_min samples --
one gets a consistent delay for each system call that is usually just a
little short of the selected period time. However, When using
snd_pcm_read/write(), one gets a jittery delay that alternates between
less than a millisecond and approximately two period times. This is
caused by snd_pcm_lib_{read,write}1() transferring any available samples
to the user's buffer and adjusting the application pointer prior to
sleeping to the end of the current period. When the next period
interrupt occurs, there is then less than avail_min samples remaining to
be transferred in the period, so we end up sleeping until a second
period occurs.
This is solved by using runtime->twake as the number of samples needed
for a wakeup in addition to selecting the proper wait queue to wake in
snd_pcm_update_state(). This requires twake to be non-zero when used
by snd_pcm_lib_{read,write}1() even if avail_min is zero.
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: usb/endpoint, fix dangling pointer use
ALSA: asihpi - Get rid of incorrect "long" types and casts.
ASoC: DaVinci: Fix McASP hardware FIFO configuration
ALSA: hda - Fix line-in for mb5 model MacBook (Pro) 5,1 / 5,2
ALSA: usb-audio: fix UAC2 control value queries
ALSA: usb-audio: parse UAC2 sample rate ranges correctly
ALSA: usb-audio: fix control messages for USB_RECIP_INTERFACE
ALSA: usb-audio: add check for faulty clock in parse_audio_format_rates_v2()
ALSA: hda - Don't check capture source mixer if no ADC is available
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On initialization ADC/DAC are enabled only for UDA1341, that's why
bias_level shall be set to off explicitly, otherwise dapm is
misinformed about bias_level on startup.
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>
This change wipes out a hardcoded macro, which enables codec bias
level control. Now is_powered_on_standby value shall be used instead.
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>
