This patch adds the common machine driver for SMDKs that
have a WM9713 codec attched to the AC97 controller.
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>
Add the AC97 controller driver for Samsung SoCs that have one.
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>
Sid Boyce reported that his machine locks up without enable_msi=0 option.
This looks like another ASUS mobo with Nvidia combo.
Reported-by: Sid Boyce <sboyce@blueyonder.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The code in pcm_lib updating runtime->hw_ptr_interrupt expects
that runtime->boundary is divisible with runtime->period_size.
Thanks are going to Clemens Ladisch for the notice.
Fix the runtime->boundary calculation using buffer_size * period_size
as base and find a least common multiple for 32bit platforms when
the expression might overflow.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Barry Song <Barry.Song@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Barry Song <Barry.Song@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr
update functions" commit:
"It is possible for the status/delay ioctls to be called when the sound
card's pointer register alreay shows a position at the beginning of the
new period, but immediately before the interrupt is actually executed.
(This happens regularly on a SMP machine with mplayer.) When that
happens, the code thinks that the position must be at least one period
ahead of the current position and drops an entire buffer of data."
Return back the hw_ptr_interrupt variable. The last interrupt pointer
is always computed from the latest hw_ptr instead of tracking it
separately (in this case all hw_ptr checks and modifications might
influence also hw_ptr_interrupt and it is difficult to keep it
consistent).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Sometimes after a suspend-resume cycle, the ALSA application
restarts the stream when resume fails and McASP fails to work
as the clock is not enabled. This patch corrects this bug.
Testes on TI DA850/OMAP-L138 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In azx_stream_start() and azx_stream_stop(),
it use azx_readb/azx_writeb to read/write SIE,
it just enable/disable 8 streams.
But according to the HDA spec, it support 30 streams,
and the new HDA controller will support more then 8
streams. So we should use azx_readl/azx_writel to
read/write SIE.
Signed-off-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make sure that capture DMA doesn't stay enabled after system resume
as that potentially prevents the processor from entering deep sleep
states.
Signed-off-by: Florian Zumbiehl <florz@florz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
tests show frequent suspend/resume(frequent poweroff/on ad1836 internal
components) maybe make ad1836 clock mode wrong sometimes after wakeup.
This patch reset/restore ad1836 clock mode while executing PM, then
ad1836 can always resume to right clock status.
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>
Several SuperH platforms, including sh7722, sh7343, sh7354, sh7367 include
a Sound Interface Unit (SIU). This patch adds DAI and platform / DMA
drivers for this interface.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The COEF setup for model=auto seems problematic on some laptops,
resulting in the silent speaker output. Better to disable it for now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The output of COEF index/value in the proc file for Realtek codecs is
rather useless since the value varies together with the index.
Let's get rid of it again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remember to free the temporary register-cache.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Now one can choose speaker configuration in e.g. PulseAudio mixer
Signed-off-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP pin (0x16) control has to be changed dynamically depending on
the master volume switch as well as the speaker pin (0x1a). Otherwise
the headphone still sounds with master off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the wrongly set up for SPDIF output on Conexant 5051 codec.
It must point to the audio out widget instead of a pin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize the mic ports B & C on Conexant 5051 codec dynamically
according to the mic jack detection, instead of static init arrays.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support for Toshiba Satellite M300 with Conexant 5051 codec.
Since the laptop has no port C connection and the pin reports always
the jack sense true, we need to ignore port-C unsol event.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Minor fixes for HP Compaq Presario F700 quirks with Cxt5051 codec:
- changed the capture mixer elements to the standard name.
- fixed the quirk name string without a space
- sorted the quirk list
- updated the documentation
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SPDIF-input pin 0x1c is muted by default in hardware. Unmute appropriate
pin to get captured samples instead zeros. Tested on Lenovo Thinkstation.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This provides a small power saving when audio is inactive.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently ASoC always maintains the bias of the CODEC while the system
is active. With older mobile CODECs this is required since the outputs
are referenced to a non-zero voltage and enabling or disabling this
voltage without audible pops or clicks in the output takes too long to
do when starting or stopping audio.
As a result of features such as ground referenced outputs and class D
speaker drivers current generation devices are able to power on and off
much more quickly without these system level issues so provide a new
flag idle_bias_off in snd_soc_codec which will cause the core to turn
off the CODEC bias. The distinction between STANDBY and OFF is still
maintained. This is partly for consistency but also allows for
potential future extensions such as per-machine overrides or deferring
the bias removal.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The same information is now visible via debugfs and with large modern
devices dumping everything to the console can be very resource
intensive, causing more harm than good.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O"
commit. New sleeping queue is introduced to separate user space and kernel
space wake_ups. runtime->nowake is renamed to twake (transfer wake).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add possibility to configure the burst mode BCLK divider through platform
data structure.
The BCLK divider changes the actual speed of the serial bus in burst mode,
which is faster than the sampling frequency of the running stream.
In this way platforms can experiment with the optimal burst speed without
the need to modify the codec driver itself.
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>
The BCLK divider was not configured in case of mode7.
This leads to unpredictable behavior when switching between FIFO modes.
Configure the BCLK divider depending on the fifo_mode (FIFO is in use,
or FIFO bypass).
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 patch fixes the aut-mute setup on HP T5735 with ALC262 codec.
Instead of wrong amp, use pin control toggling for muting the speaker now.
Tested-by: Lee Trager <lee.trager@hp.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some codecs disable widgets used for output pins and reserve as vendor-
spec widgets. Thus we need to check the widget type and pin cap before
actually sending SET_EAPD verbs in the auto-configuration mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC259 has a widget NID 0x21 for the output pin, but it wasn't handled
properly in alc268_new_analog_output().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the prefill number of samples as the same as the lower
threshold in mode7.
In this way the codec will read the same amount of data on
startup and during the running playback.
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>
As snd_seq_timer_set_tick_resolution() is always called with the same
three fields of struct snd_seq_timer, it suffices to give that as the
only parameter.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
pgprot_noncached() can be set for vmalloc'ed buffers safely, and we'd
need non-cached behavior more or less, even for the intermediate ring-
buffers.
Now snd_pcm_lib_mmap_vmalloc() is added as the common PCM mmap callback
that is coupled with snd_pcm_lib_alloc_vmalloc_buffer() & co.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a card instance with a higher card number is registered, warning
messages are spewed eventually with stack traces due to the invalid minor
number for OSS device registration. For example, thinkpad-acpi registers
the card number 29 as default, and you'll see always these messages.
This is rather confusing (and worries users), thus better to return
simply the error code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently they don't build due to cross tree dependencies, they will be
reenabled once the arch/arm side has merged.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sony VAIO VGN-P11G with ALC262 codec has only one input pin, and the
recording doesn't work with model=auto because ALC262 parser sets the
wrong cap NIDs to choose the route and the default route for the sole
input pin wasn't initialized properly. This patch solves these issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This has been superceeded by Sascha's new driver but was not removed in
the patch series due to cutdowns for review.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While dynamically allocated DAIs are the way forward the core doesn't
yet support anything except matching with a pointer to the actual DAI
so convert to doing that so that machine drivers don't have to jump
through hoops to register themselves.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Javier Martin <javier.martin@vista-silicon.com>
Don't unconditionally include the i.MX2x DMA driver, the arch/arm
functions it uses aren't available for i.MX3x.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Javier Martin <javier.martin@vista-silicon.com>
The old driver has the number of SSI units in the system hardcoded,
does not make use of the device model and works only on i.MX21/27.
This driver replaces it. It works in DMA mode on i.MX21/27 and using
an FIQ handler on other systems. It also supports AC97 mode of
the SSI units.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Javier Martin <javier.martin@vista-silicon.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for sample rates other than 44100Khz on raumfeld audio
devices. At startup time, call snd_soc_dai_set_sysclk() with 0 as 'freq'
argument so it offers all the sample rates. Later, the function is
called again to give proper constraints.
Use the external audio clock generator to provide double data rate
clocks as the PXA's internal baud generator does anything but what's
described in the datasheets.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For setups with variable MCLKs, the current logic of limiting the
available sampling rates at startup time is not sufficient. We need to
be able to change the setting at a later point, and so the codec must
offer all possible rates until the hw_params are given.
This patches allows that by passing 0 as 'freq' argument to
cs4270_set_dai_sysclk().
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On my laptop (HP dv6-1110ax), there are no OEM strings in SMBIOS of type
"HP_Mute_LED*". Hence, the GPIO for the mute button LED doesn't get set
properly. I didn't find the strings in my cousin's laptop (HP dv9500t CTO)
either.
As per the documentation of find_mute_led_gpio(), these strings occur
in HP B-series systems - so, before scanning the SMBIOS strings, we need to
check if we're dealing with a B-series system.
Need to get confirmation from HP if this logic takes care of all the
systems. I'm trying to poke a friend there.
Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is needed for built-in drivers which are built before the sound directory,
like thinkpad_acpi.
Otherwise, registering a card fails.
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture-related mixer elements are missing with ALC861/ALC660 codecs
when quirks are present, due to missing call of set_capture_mixer().
Reference: Novell bnc#567340
http://bugzilla.novell.com/show_bug.cgi?id=567340
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Signed-off-by: Thomas Weber <weber@corscience.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added a new function to look up a quirk entry with the given PCI SSID
instead of a pci device pointer. This can be used when the searched ID
is overridden for debugging or such a purpose.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for automatically muting the speakers when headphones
are inserted, as well as relabelling the headphone widgets from the
non-standard "HP" to the standard "Headphone" for the mb5 model.
Signed-off-by: Alex Murray <murray.alex@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The alc664-mode4 model doesn't seem to fit with Toshiba NB205 correctly.
NB205 uses the pin 0x17 connected with the mixer 0x0f for the speaker
output, which isn't controlled by mode4 model at all.
Rather model=auto works fine as is on the latest driver, so let it back
again.
Tested-by: Nickolas Lloyd <ultrageek.lloyd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The transmitter supports all sample rates up to 192KHz, so the driver
should not give a limit.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The codec structure initialization statements should be
separated by semicolons.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the Intel Cougar Point (PCH) HD Audio Controller DeviceIDs.
Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture source or input source mixer element wasn't created properly
for ALC861-VD codec due to the wrong NID passed to
alc_auto_create_input_ctls().
References: Novell bnc#568305
http://bugzilla.novell.com/show_bug.cgi?id=568305
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
tpa6140a2 uses different names for the regulators.
Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the STMicroelectronics ST7597 codec and an unknown codec
from the same manufacturer found on the Creative SB 128 card (CT4810).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This model needs both 'Headphone Jack Sense' and 'Line Jack Sense' muted
for audible playback, so just add it to the ad1981 jack sense blacklist.
Cc: stable@kernel.org
Tested-by: Pete <x41215201@gmail.com>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
We need to set the LRCLK inversion bit to select DSP mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
The patch "sbawe: fix memory detection" fixed detection
for memoryless SB32 cards but broke detection of memory
above 512KB. This patch fixes the regression.
The patch has been tested on the SB32 card (CT3670) with
0MB, 2MB and 8MB memory installed.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The problem is that in the original code sound_nblocks could go up to 1024
which would be an array overflow.
This was found with a static checker and has been compile tested only.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The XO's audio hardware is wired up to allow DC sensors (e.g. light
sensors, thermistors, etc) to be plugged in through the microphone jack.
Add sound mixer controls to allow this mode to be enabled and tweaked.
Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The XO-1.5 has a microphone LED designed to indicate to the user when
something is being recorded.
This light is controlled by the microphone bias voltage and it is
currently coming on all the time.
This patch defers the microphone port configuration until when recording
is actually taking place, fixing the behaviour of the LED.
Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added patch for Hewlett-Packard Company Device Subsystem id - 103c:30ea.
Signed-off-by: Ken Prox <kprox@users.sourceforge.net>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The WM9705 and WM9703 ops are the same actually so use
the same code for both.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When runtime->periods == 1 or when pointer crosses end of ring buffer,
the delta might be greater than buffer_size.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
As noted by pl bossart <bossart.nospam@gmail.com>, the PCM I/O routines
(snd_pcm_lib_write1, snd_pcm_lib_read1) should block wake_up() calls
until all samples are not processed.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Do general cleanup in snd_pcm_update_hw_ptr*() routines and merge them.
The main change is hw_ptr_interrupt variable removal to simplify code
logic. This variable can be computed directly from hw_ptr.
Ensure that updated hw_ptr is not lower than previous one (it was possible
with old code in some obscure situations when interrupt was delayed or
the lowlevel driver returns wrong ring buffer position value).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
In some debug cases, it might be usefull to see previous ring buffer
positions to determine position problems from the lowlevel drivers.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* 'for-2.6.33' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6:
ASoC: fixup oops in generic AC97 codec glue
ASoC: fix params_rate() macro use in several codecs
ASoC: fsi-ak4642: Remove ak4642_add_i2c_device
Initialize the glue by calling snd_soc_new_ac97_codec() as is done
in other ASoC AC97 codecs. Fixes an oops caused by dereferencing
uninitialized members in snd_soc_new_pcms().
Run-tested on Au1250.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Better integration of the core with the device model means that we now
no longer get the ASoC suspend and resume callbacks without the card
having been set up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The currently available FIFO modes (mode1 and mode7) require master
mode from the codec.
Do not allow the slave configuration when the FIFO is in use.
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>
Mode 7 of tlv320dac33 operates in the following way:
The codec is in master mode.
Host configures upper and lower thresholds in tlv320dac33
During playback the codec will clock in the data until the
upper threshold is reached in FIFO. At this point the codec
stops the colocks on the serial bus.
When the FIFO fill is reaching the lower threshold limit the
codec will enable the clocks on the serial bus, and clocks
in data till the upper threshold is reached.
In this mode, we can also request interrupts for threshold
events (upper, lower and alarm), which could be used for
power management.
At this point the interrupts are not enabled for this mode,
but it can be taken into use in the future, when the surrounding
code makes it possible to use it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.oc.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use switch instead of if statements to configure FIFO bypass
and mode1.
With this change adding new FIFO mode is going to be easier,
and cleaner.
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>
Ensure that the code is going to be readable, when new FIFO modes
are introduced later.
Move the prefill and playback state handling to inlined
functions.
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>
In order to have support for more FIFO modes supported by
tlv320dac33, the switch for enabling/disabling the FIFO
use has to be replaced with an enum.
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>
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>
PM architecture of ad1938 is simple, we don't need a bundle of functions like
ad1938_pll_powerctrl, ad1938_set_bias_level for only PLL. A dapm supply will
handle on/off of PLL.
Since soc-core can poweron/off PLL on-demand, we don't need to poweron/off PLL
in suspend/resume entries too.
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>
Tests show frequent suspend/resume(frequent poweroff/on ad1836 internal
components) maybe make ad1836 clock mode wrong sometimes after wakeup.
This patch reset/restore ad1836 clock mode while executing PM, then
ad1836 can always resume to right clock status.
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>
Detect the HVR-950Q HVR-850 urb data alignment quirk using usbquirk.h
rather than using a case statement in snd_usb_audio_probe.
Signed-off-by: John S. Gruber <JohnSGruber@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Addressing audio quality problem.
In sound/usb/usbaudio.c, for the Hauppage HVR-950Q and HVR-850 only, change
retire_capture_urb to allow transfers on audio sub-slot boundaries rather
than audio slots boundaries.
With these devices the left and right channel samples can be split between
two different urbs. Throwing away extra channel samples causes a sound
quality problem for stereo streams as the left and right channels are
swapped repeatedly, perhaps many times per second.
Urbs unaligned on sub-slot boundaries are still truncated to the next
lowest stride (audio slot) to retain synchronization on samples even
though left/right channel synchronization may be lost in this case.
Detect the quirk using a case statement in snd_usb_audio_probe.
BugLink: https://bugs.launchpad.net/ubuntu/+bug/495745
Signed-off-by: John S. Gruber <JohnSGruber@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since there are devices that do not align the size of their data packets
to frame boundaries, the driver needs to be able to keep track of
partial frames. This patch prepares for support for such devices by
changing the hwptr_done variable from a frame counter to a byte counter.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the release of substreams may be done asynchronously from the
disconnection, close callback needs to check the shutdown flag before
actually accessing the usb interface.
Reference: Novell bnc#505027
http://bugzilla.novell.com/show_bug.cgi?id=565027
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch converts the alc889 Aspire-specific powerdown to a generic
one. Like the previous effort, it currently only handles Front and PCM
but can be easily extended to cover other nids. The existing hook for
alc889 Aspire-specific remains enabled. Upon further testing, I've added
its use for ALC861_AUTO as well. Following patches will enable them for
other quirks.
Tested-by: Dr. David Alan Gilbert <linux@treblig.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch ports powerdown fixes to AD198x. Currently we only turn off
Front and HP for suspend, but this is easily extended for additional
nids.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a common helper function for clearing pin controls before suspend.
Use the pincfg array instead of looking through all widget tree.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change for supporting dynamic beep device allocation caused
a problem resulting in Oops at reloading the driver. Also, it ignores
the error from input device registration.
This patch fixes the wrong check in snd_hda_detach_beep_device(), and
returns an error when the input device registration fails properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use snd_hda_jack_detect() again for jack-sensing.
The triggering problem can be worked around with codec->no_trigger_sense
flag now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Analog Device codecs seem to have problems with the triggering of
pin-sensing although their pincaps give the trigger requirements.
Some reported that constant CPU load on HP laptops with AD codecs.
For avoiding this regression, add a flag to codec struct to notify
explicitly that the codec doesn't suppot the trigger at pin-sensing.
Tested-by: Maciej Rutecki <maciej.rutecki@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we run the following commands in turn (with
CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0),
speaker-test -Dhw:0,3 -c2 -twav # HDMI
speaker-test -Dhw:0,0 -c2 -twav # Analog
The second command will produce sound in the analog lineout _as well as_
HDMI sink. The root cause is, device 0 "reuses" the same stream tag that
was used by device 3, and the "intelhdmi - sticky stream id" patch leaves
the HDMI codec in a functional state. So the HDMI codec happily accepts
the audio samples which reuse its stream tag.
The proposed solution is to remember the last device each azx_dev was
assigned to, and prefer to
1) reuse the azx_dev (and hence the stream tag) the HDMI codec last used
2) or assign a never-used azx_dev for HDMI
With this patch and the above two speaker-test commands,
HDMI codec will use stream tag 8 and Analog codec will use 5.
The stream tag used by HDMI codec won't be reused by others, as long
as we don't run out of the 4 playback azx_dev's. The legacy Analog
codec will continue to use stream tag 5 because its device id is 0
(this is a bit tricky).
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Narrow the dma and irq selection after the DOS driver.
Add ALSA configuration description as well.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sevelar ASoC codec drivers wrongly assume, that the params_rate() macro
returns one of SNDRV_PCM_RATE_* defines instead of the actual numerical
sampling rate. Fix them.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
I2C devices should be registered when platform board setting
in latest ASoC.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes a bug where "virtual" registers were being written to the ac97
bus. This was causing unrelated registers to become corrupted (headphone 0x04,
touchscreen 0x78, etc).
This patch duplicates protection that was included in the wm9713 driver.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Postpone the mixer name setup after the codec patch since the codec
patch may change the codec name string in itself.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the wrong implementation of NID <-> kctl mapping for capture mixers
introduced by the ocmmit 5b0cb1d850.
So far, the driver returns an error at probe.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 2ccafed4 added an extra parameter to the DAI .set_pll() method, but
it missed this call in sound/soc/imx/mx27vis_wm8974.c.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
platform_get_irq returns -ENXIO on failure, so !irq was probably
always true. Better use (int)irq <= 0. Note that a return value of
zero is still handled as error even though this could mean irq0.
This is a followup to 305b3228f9 that
changed the return value of platform_get_irq from 0 to -ENXIO on error.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A machine with AMD CPU with Nvidia board doesn't work with MSI.
Reported-by: Robert J. King <peritus@gurunetwork.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the attached patch I am able to use the sound on a new IMac 27.
What works:
*) Internal speakers
*) Internal microphone
*) Headphone
I don't have an external mic or a SPDIF device to test the rest.
Signed-off-by: Rafael Avila de Espindola <rafael.espindola@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Refine the rate selection by choosing the rate
closer to the requested one in case of selecting
single frequency. Previously, the higher rate was
always selected.
Also, fix problem with the best_diff unsigned int
value wrapping (turning negative).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is one of Sound Blaster Pro compatible chipsets which is supported
by Linux OSS driver and was missing native supoort for ALSA.
The Jazz16 audio codec is Crystal CS4216 which is capable
of playback and recording up to 48 kHz stereo.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Memory amount is increased before a successful write-read
sequence is done. Thus, 512 kB of onboard memory is detected
on memoryless cards like SB32.
Move the increasing of memory counter after successful read
is done.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The direction of rounding is incorrect in the snd_interval_ratnum()
It was detected with following parameters (sb8 driver playing
8kHz stereo file):
- num is always 1000000
- requested frequency rate is from 7999 to 7999 (single frequency)
The first loop calculates div_down(num, freq->min) which is 125.
Thus, a frequency range's minimum value is 1000000 / 125 = 8000 Hz.
The second loop calculates div_up(num, freq->max) which is 126
The frequency range's maximum value is 1000000 / 126 = 7936 Hz.
The range maximum is lower than the range minimum so the function
fails due to empty result range.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current Realtek code makes no specific provision for turning stuff
off. The codec chip is placed into low-power mode generically, but this
doesn't turn off any external hardware connected to it, in particular
external amplifiers.
This patch creates a hook function that is called by the codec
suspend/resume functions. It ought to disable any external hardware in a
device-specific way. I've implemented a generic ALC889 function that
sets the EAPD pin properly, and used it for the Acer Aspire 8930G which
can benefit from this feature.
On my laptop, this results in ~0.5W extra savings.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes some extra mixers that do nothing on the Acer Aspire
8930G.
The CD mixer is useless because the SATA DVD/Blu-Ray drive has no analog
audio output, and the Side mixer is useless because we max out at 6ch
anyway.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch just simplifies the 8930G verb array a bit. Just use the
common ALC889 EAPD verb array to make things more consistent. The file
is already huge enough already.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/479373
The OR has verified with hda-verb that the internal microphone needs
VREF50 set for audible capture.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use kzalloc rather than kcalloc(1,...)
The semantic patch that makes this change is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
@@
- kcalloc(1,
+ kzalloc(
...)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We can use finer-grained locking, which makes things easier when
we gain DMA support.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the recording and playback paths are now the same, eliminate
the needless conditionals.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's no need for a specific rule; ALSA's generic AC'97 support
calculates the necessary rate constraint information itself, and
we can use this directly.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of listing all individual PCI IDs, check the matching with
the PCI class together with the vendor id for Nvidia.
This simplifies the pci id entries.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8955 is a low power, high quality stereo DAC with integrated
headphone and loudspeaker amplifiers, designed to reduce external
component requirements in portable digital audio applications. This is
an initial driver implementing support for the majority of the
functionality in the device, currently OUT3 is not supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The wm8974 datasheet defines BUFIOEN as bit 2.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the
equivalent core functions instead.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the
equivalent core functions instead.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the
equivalent core functions instead.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the
equivalent core functions instead.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the
equivalent core functions instead.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are now five copies of the code to allocate a PCM buffer using
vmalloc(). Add a sixth in the core so that the others can be removed.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When allocating the PCM buffer, use vmalloc_user() instead of vmalloc().
Otherwise, it would be possible for applications to play the previous
contents of the kernel memory to the speakers, or to read it directly if
the buffer is exported to userspace.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
