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>
Some model quirks missed the corresponding capsrc_nids. This resulted in
non-working capture source selection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Conexant CX20583-10Z has digital beep device with volume control.
Making use of them.
Signed-off-by: Einar Rünkaru <einarry@smail.ee>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed initialization of internal mic and added internal mic boost control
Renamed analog mic boost control to ext mic boost contol.
Name pair analog/digital seems too confusing for a normal user.
Signed-off-by: Einar Rünkaru <einarry@smail.ee>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
1. Add more ASUS NB model.
2. Fixed alc663_m51va_setup
M51VA has Digital Mic that NID is 0x12. The record source index is
0x9 for ALC663.
So, to modify the alc663_m51va_setup function to index 0x9
and add analog Mic aupport function alc663_mode1_setup.
3. Add ASUS mode7 and mode8 modules for ALC663
Signed-off-by: Kailang Yang <kailang@realtek.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DAC_VOICE_EN was not defined - looks to have been overly enthusiastically
deleted from a previous revision of the patch, pull the value from v1.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
fix precision of PLL computation for TLV320AIC3x SoC driver,
test results are at http://pmeerw.net/clk
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
Acked-by: Vladimir Barinov <vova.barinov@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Recent drivers/mfd/twl4030* renames to twl broke compile for
various boards as the series was missing a patch to change
the board-*.c files.
This patch renames include twl4030.h to include twl.h
and also renames twl4030_i2c_ routines.
Signed-off-by: Balaji T K <balajitk@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reviewed-by: Felipe Balbi <felipe.balbi@nokia.com>
Cc: Samuel Ortiz <sameo@linux.intel.com>
Signed-off-by: Tony Lindgren <tony@atomide.com>
This original driver was created by Dialog Semiconductor,
and cleanuped by Kuninori Morimoto.
Special thanks to David Chen.
This became very simple ASoC codec driver,
and it is tested by EcoVec24 board.
Signed-off-by: David Chen <Dajun.chen@diasemi.com>
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>
Take the regulator framework in use for managing the power sources
Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The driver can be 'generalized' a bit by not hardcoding '2'(the number of
I2Sv3 controllers that the driver can handle) at many places, instead we
define a macro for it. That makes it easier to increase number of controllers
by changing the parameter at just one place, this will be useful when there is
support for newer SoCs, which have the same controller, only more in number.
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>
Removed redundant header includes which make no difference to compilation.
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>
In preparation for multiple device support.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
No need for the mixers to know about this, and it allows for virtual
controls.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Change code so that switching to playing music through the analog output
disables SPDIF out instead of disabling it when stream ends.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com
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: ac97_codec - increase timeout for analog sections to 5 second
ASoC: Correct code taking the size of a pointer
ALSA: hda - Add PCI IDs for Nvidia G2xx-series
ALSA: sound/isa/gus: Correct code taking the size of a pointer
ALSA: hda: Fix max PCM level to 0 dB for AD1981_HP
ALSA: hda: Use ALC260_WILL quirk for another Acer model (0x1025007f)
Makes use of skip_spaces() defined in lib/string.c for removing leading
spaces from strings all over the tree.
It decreases lib.a code size by 47 bytes and reuses the function tree-wide:
text data bss dec hex filename
64688 584 592 65864 10148 (TOTALS-BEFORE)
64641 584 592 65817 10119 (TOTALS-AFTER)
Also, while at it, if we see (*str && isspace(*str)), we can be sure to
remove the first condition (*str) as the second one (isspace(*str)) also
evaluates to 0 whenever *str == 0, making it redundant. In other words,
"a char equals zero is never a space".
Julia Lawall tried the semantic patch (http://coccinelle.lip6.fr) below,
and found occurrences of this pattern on 3 more files:
drivers/leds/led-class.c
drivers/leds/ledtrig-timer.c
drivers/video/output.c
@@
expression str;
@@
( // ignore skip_spaces cases
while (*str && isspace(*str)) { \(str++;\|++str;\) }
|
- *str &&
isspace(*str)
)
Signed-off-by: André Goddard Rosa <andre.goddard@gmail.com>
Cc: Julia Lawall <julia@diku.dk>
Cc: Martin Schwidefsky <schwidefsky@de.ibm.com>
Cc: Jeff Dike <jdike@addtoit.com>
Cc: Ingo Molnar <mingo@elte.hu>
Cc: Thomas Gleixner <tglx@linutronix.de>
Cc: "H. Peter Anvin" <hpa@zytor.com>
Cc: Richard Purdie <rpurdie@rpsys.net>
Cc: Neil Brown <neilb@suse.de>
Cc: Kyle McMartin <kyle@mcmartin.ca>
Cc: Henrique de Moraes Holschuh <hmh@hmh.eng.br>
Cc: David Howells <dhowells@redhat.com>
Cc: <linux-ext4@vger.kernel.org>
Cc: Samuel Ortiz <samuel@sortiz.org>
Cc: Patrick McHardy <kaber@trash.net>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Previously, OLPC support for the mic extensions was only enabled in the
ALSA driver if CONFIG_OLPC and CONFIG_MGEODE_LX were both set. This was
because the old geode GPIO code was written in a manner that assumed
CONFIG_MGEODE_LX. With the new cs553x-gpio driver, this is no longer the
case; as such, we can drop the requirement on CONFIG_MGEODE_LX and instead
include a requirement on GPIOLIB.
We use the generic GPIO API rather than the cs553x-specific API.
Signed-off-by: Andres Salomon <dilinger@collabora.co.uk>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Jordan Crouse <jordan@cosmicpenguin.net>
Cc: David Brownell <david-b@pacbell.net>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
If ak4642 driver was compiled without I2C configs,
ak4642_modinit return value will become un-stable.
This patch modify this bug
Reported-by: Magnus Damm <damm@opensource.se>
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 HDA_SUBDEV_NID_FLAG is duplicate for amplifier control elements. Move
get_amp_nid_() call to the snd_hda_ctl_add() function.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The purpose of this changeset is to show information about amplifier
setting in the codec proc file. Something like:
Control: name="Front Playback Volume", index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Control: name="Front Playback Switch", index=0, device=0
ControlAmp: chs=3, dir=In, idx=2, ofs=0
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This set of changes add missing NID values to some static control
elemenents. Also, it handles all "Capture Source" or "Input Source"
controls.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
I have a Soundblaster 16PCI. For many years, alsa has had a bug where
not all of the card's controls are detected (many alsa versions,
many kernel versions). In particular, Master Playback Volume is
usually not detected, and so I get no sound or extremely faint sound.
The problem has always been inconsistent: sometimes all of the controls
are detected correctly, and sometimes a partial set is detected. It works
correctly about 10% of the time.
Finally, I got around to tracking down the problem. When the driver
fails, it prints the kernel message "AC'97 0 analog subsections not
ready". This message is generated from the function snd_ac97_mixer()
in ac97_codec.c. The message indicates that the card failed to come
back after reset within the time limit. The time limit is
120 milliseconds.
I tried increasing the time limit to 1 second, and found that this
made the driver work about 70% of the time. I tried increasing it
to 5 seconds, and it now seems to work 100% of the time.
I expect that this change would be completely harmless for
existing cards that work, and would only introduce additional
delay for cards that do not work.
ALSA bug#4032.
Signed-off-by: Steve Soule <sts11dbxr@gmail.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add experimental support for the Edirol UA-101 audio/MIDI interface.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sizeof(codec->reg_cache) is just the size of the pointer. Elsewhere in the
file, codec->reg_cache is used with sizeof(wm8900_reg_defaults), so the
code is changed to do the same here.
A simplified version of the semantic patch that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression *x;
expression f;
type T;
@@
*f(...,(T)x,...)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert table of pointers to mixer controls into tables
of the mixer controls. It saves about 20% of the snd-sb-common
module size reported by lsmod.
The als4000 uses part of sb16's control table.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sizeof(share_id) is just the size of the pointer. On the other hand,
block->share_id is an array, so its size seems more appropriate.
A simplified version of the semantic patch that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression *x;
expression f;
type T;
@@
*f(...,(T)x,...)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/461062
The original reporter states that PCM maxes at +12 dB and results in
very bad distortion. Cap PCM at 0 dB to resolve this symptom.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/418627
The original reporter states that this quirk is necessary to obtain
reasonable gain for playback. Without it, sound is inaudible. Tested
with playback (spkr and hp) and capture.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch renames function names like twl4030_i2c_write_u8,
twl4030_i2c_read_u8 to twl_i2c_write_u8, twl_i2c_read_u8
and also common variable in twl-core.c
Signed-off-by: Rajendra Nayak <rnayak@ti.com>
Signed-off-by: Balaji T K <balajitk@ti.com>
Signed-off-by: Santosh Shilimkar <santosh.shilimkar@ti.com>
Acked-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
The upcoming TWL6030 is companion chip for OMAP4 like the current TWL4030
for OMAP3. The common modules like RTC, Regulator creates opportunity
to re-use the most of the code from twl4030.
This patch renames few common drivers twl4030* files to twl* to enable
the code re-use.
Signed-off-by: Rajendra Nayak <rnayak@ti.com>
Signed-off-by: Balaji T K <balajitk@ti.com>
Signed-off-by: Santosh Shilimkar <santosh.shilimkar@ti.com>
Acked-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
Bring the WM8350 IRQ API more in line with the generic IRQ API by
masking and unmasking interrupts as they are requested and freed.
This is mostly just a case of deleting the mask and unmask calls
from the individual drivers.
The RTC driver is changed to mask the periodic IRQ after requesting
it rather than only unmasking the alarm IRQ. If the periodic IRQ
fires in the period where it is reqested then there will be a
spurious notification but there should be no serious consequences
from this.
The CODEC drive is changed to explicitly disable headphone jack
detection prior to requesting the IRQs. This will avoid the IRQ
firing with no jack set up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
This is done as simple code transformation, the semantics of the
IRQ API provided by the core are are still very different to those
of genirq (mainly with regard to masking).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Overwrite pin config on intel DG45ID board.
intelhdmi - dont power off HDA link
ALSA: hrtimer - Fix lock-up
ALSA: intelhdmi - add channel mapping for typical configurations
ALSA: intelhdmi - channel mapping applies to Pin
ALSA: intelhdmi - accept DisplayPort pin
ALSA: hda - show HBR(High Bit Rate) pin cap in procfs
ALSA: hda - Fix LED GPIO setup for HP laptops with IDT codecs
ASoC: Fix build of OMAP sound drivers
ALSA: opti93x: fix irq releasing if the irq cannot be allocated
Add dB scale for mixer controls. Fix dB scale for
Master Volume control.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pin config provided by BIOS have some problems:
0x0221401f: [Jack] HP Out at Ext Front <-- other association and sequence
0x02a19020: [Jack] Mic at Ext Front <-- other association
0x01113014: [Jack] Speaker at Ext Rear <-- line out (not speaker)
0x01114010: [Jack] Speaker at Ext Rear <-- line out
0x01a19030: [Jack] Mic at Ext Rear <-- other association
0x01111012: [Jack] Speaker at Ext Rear <-- line out
0x01116011: [Jack] Speaker at Ext Rear <-- line out
0x40f000f0: [N/A] Other at Ext N/A
0x40f000f0: [N/A] Other at Ext N/A
0x40f000f0: [N/A] Other at Ext N/A
0x40f000f0: [N/A] Other at Ext N/A
0x40f000f0: [N/A] Other at Ext N/A
0x01451140: [Jack] SPDIF Out at Ext Rear
0x40f000f0: [N/A] Other at Ext N/A
just overwrite it.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move OPTi93x controls definitions to the opti93x driver
from the common wss-lib library module. These controls
are used only by the opti93x driver.
Also, fix capture source names. They are the same as
opl3sa2 names.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The als100 driver is so similar to the dt019x/als007 driver
that one driver's source can be used for both drivers with
only few changes. Merge the dt019x driver into the als100.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For codecs without EPSS support (G45/IbexPeak), the hotplug event will
be lost if the HDA is powered off during the time. After that the pin
presence detection verb returns inaccurate info.
So always power-on HDA link for !EPSS codecs.
KarL offers the fact and Takashi recommends to flag hda_bus. Thanks!
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The timer stop callback can be called from snd_timer_interrupt(), which
is called from the hrtimer callback. Since hrtimer_cancel() waits for
the callback completion, this eventually results in a lock-up.
This patch fixes the problem by just toggling a flag at stop callback
and call hrtimer_cancel() later.
Reported-and-tested-by: Wojtek Zabolotny <W.Zabolotny@elka.pw.edu.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
IbexPeak is the first Intel HDMI audio codec to support channel mapping.
Currently the outstanding problem is, the HDMI channel order do not
agree with that of ALSA. This patch presents workaround for some
typical use cases. It gives priority to the typical ALSA surround
configurations, and defines channel mapping for them.
We may need better kernel+userspace interactive channel mapping scheme.
For example, in current scheme if user plays with the surround50 device,
the kernel is unaware of this and will still select the surround41
channel allocation and channel mapping..
Thanks to Marcin for offering good tips!
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDA036-A specifies that the Audio Sample Packet (ASP) Channel Mapping
verbs apply to Digital Display Pin Complex instead of Converter.
With this fix, channel mapping is working as expected for IbexPeak.
Thanks to Marcin for pointing this out!
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDA036 spec states:
DP (Display Port) indicates whether the Pin Complex Widget supports
connection to a Display Port sink. Supported if set to 1. Note that
it is possible for the pin widget to support more than one digital
display connection type, e.g. HDMI and DP bit are both set to 1.
Also export the DP pin cap in procfs.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Note that the HBR capability only applies to HDMI pin.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes an error in processing of the HP BIOS configuration to enable
GPIO based mute LED indicator control. That error causes driver to enable
such control on all HP systems with the 92HD75 IDT codecs and results in
unnecessary toggling of the GPIO on mute control manipulation.
It also adds support of the future HP BIOS configuration extension for the
named control. New configuration string has a format HP_Mute_LED_P_G
where P can be 0 or 1 and defines mute LED GPIO control state (low/high)
that corresponds to the NOT muted state of the master volume
and G is the index of the GPIO to use (0..9)
Lastly, it adds more systems to the support of the audio implementation
as found on HP B-series systems
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are build errors when building for some of the omap2/3 boards without
enabling sound:
sound/built-in.o:(.data+0x43bc): undefined reference to `soc_codec_dev_tlv320aic23'
sound/built-in.o:(.data+0x43cc): undefined reference to `tlv320aic23_dai'
Confused me quite a bit since the drivers that had references to the
codec weren't enabled. Turns out the Makefile was using the wrong
config option to enable them. Patch below.
Reported-by: Anand Gadiyar <gadiyar@ti.com>
Signed-off-by: Olof Johansson <olof@lixom.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While Linux provided an O_SYNC flag basically since day 1, it took until
Linux 2.4.0-test12pre2 to actually get it implemented for filesystems,
since that day we had generic_osync_around with only minor changes and the
great "For now, when the user asks for O_SYNC, we'll actually give
O_DSYNC" comment. This patch intends to actually give us real O_SYNC
semantics in addition to the O_DSYNC semantics. After Jan's O_SYNC
patches which are required before this patch it's actually surprisingly
simple, we just need to figure out when to set the datasync flag to
vfs_fsync_range and when not.
This patch renames the existing O_SYNC flag to O_DSYNC while keeping it's
numerical value to keep binary compatibility, and adds a new real O_SYNC
flag. To guarantee backwards compatiblity it is defined as expanding to
both the O_DSYNC and the new additional binary flag (__O_SYNC) to make
sure we are backwards-compatible when compiled against the new headers.
This also means that all places that don't care about the differences can
just check O_DSYNC and get the right behaviour for O_SYNC, too - only
places that actuall care need to check __O_SYNC in addition. Drivers and
network filesystems have been updated in a fail safe way to always do the
full sync magic if O_DSYNC is set. The few places setting O_SYNC for
lower layers are kept that way for now to stay failsafe.
We enforce that O_DSYNC is set when __O_SYNC is set early in the open path
to make sure we always get these sane options.
Note that parisc really screwed up their headers as they already define a
O_DSYNC that has always been a no-op. We try to repair it by using it for
the new O_DSYNC and redefinining O_SYNC to send both the traditional
O_SYNC numerical value _and_ the O_DSYNC one.
Cc: Richard Henderson <rth@twiddle.net>
Cc: Ivan Kokshaysky <ink@jurassic.park.msu.ru>
Cc: Grant Grundler <grundler@parisc-linux.org>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Ingo Molnar <mingo@elte.hu>
Cc: "H. Peter Anvin" <hpa@zytor.com>
Cc: Thomas Gleixner <tglx@linutronix.de>
Cc: Al Viro <viro@zeniv.linux.org.uk>
Cc: Andreas Dilger <adilger@sun.com>
Acked-by: Trond Myklebust <Trond.Myklebust@netapp.com>
Acked-by: Kyle McMartin <kyle@mcmartin.ca>
Acked-by: Ulrich Drepper <drepper@redhat.com>
Signed-off-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Jan Kara <jack@suse.cz>
Use the chip->irq to check if the irq should be released so the irq is not released
if it has not been allocated.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The volume levels in original implementation are incorrect and does
not match the dB scale. The real range is linear (in the sense of
the dB scale) from 0dB to -100dB. Remove logaritmic table and make
all volumes from range 0dB..100dB.
The tests are in RedHat's bugzilla #540817.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Quirk for the ALC662 found on the Intel D945GCLF2 (and possibly other)
mainboards.
Signed-off-by: David Santinoli <david@santinoli.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Confirmed from vendor and tests in RedHat bugzilla #536782 .
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The wrong variable was returned in the case of an error
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I added the product IDs of the new revisions of the devices, so owners
can test whether this suffices to make them work. Patched against ALSA
snapshot 20091207.
Signed-off-by: Tobias Hansen <Tobias.Hansen at physik.uni-hamburg.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Realtek codecs, a digital mic pin is connected often only to a single
ADC. But the parser tries to set up all ADCs no matter whether the
digital mic is available, and results in non-selectable input source.
This patch adds a check of input-source availability of each ADC, and
excludes ones that don't support all input sources.
Reference: Novell bnc#561235
http://bugzilla.novell.com/show_bug.cgi?id=561235
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is an updated patch for the Apple iMac 9,1 model to add sound.
Original patch posted here:
http://article.gmane.org/gmane.linux.alsa.devel/61361/match=
I have been using this patch for a while now
and have to say it works vary well, except for a few minor
things:
With the iMac 24-inch 3.06GHz Intel Core 2 Duo
everything seems to be working as it should,
although I have not looked into the microphone
(never really use one, nor have any apps to test,
my guess is it doesn't work, or I never figured out how
to get it to work).
With the iMac 24-inch 2.66GHz Intel Core 2 Duo
everything is the same as with the above machine
except I'm hearing a light scratchy/distortion noise
come out of the speakers when using headphones(above machine
does not do this).
Other than that the sound level is great(especially with good Dj headphones).
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Tested-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PnP data on the OPTI931 and OPTI933 contains io port
range for the MC indirect registers. Use the PnP range
instead of hardwired value 0xE0E.
Also, request region of MC indirect registers so it is
marked as used to other drivers (this was missing previously).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Komuro pointed out that IRQ_FIRST_SHARED is not used at all in the
PCMCIA subsystem, so remove it. Also, remove two bogus assignments.
CC: Karsten Keil <keil@b1-systems.de>
CC: netdev@vger.kernel.org
CC: alsa-devel@alsa-project.org
CC: Komuro <komurojun-mbn@nifty.com>
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8904 is a high performance ultra-low power stereo CODEC
optimised for portable audio applications, with features including
a class W amplifier, FLL with free running mode, Mobile ReTune and
ground referenced headphone and line outputs.
Support for some features, most particularly the digital microphone
interface, is not yet present.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Make it a bit easier to tie DAPM widgets in with the register map
without referring to the source by including the register location
controlled by the widget.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Take the regulator framework in use for managing the power sources.
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>
Add clock enable and disable calls to resume and suspend respectively.
Also add a member to the audio device data structure which tracks the clock
status.
Tested on DA850/OMAP-L138 EVM. For the purpose of testing, the patches[1] which
add suspend-to-RAM support to DA850/OMAP-L138 SoC were applied.
[1] http://linux.davincidsp.com/pipermail/davinci-linux-open-source/
2009-November/016958.html
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>
This renames from a character / to : of controls. A / occurs below error
messages.
ASoC: Failed to create IN2RP/VXRP debugfs file
ASoC: Failed to create IN2LP/VXRN debugfs file
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When primary AC97 is not found, don't fail with tons of AC97 errors.
Assume that the card is SF64-PCR (tuner-only).
This makes the SF64-PCR radio card work "out of the box".
Also fixes a bug that can cause an oops here:
if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) {
when tea575x_tuner == 16, it passes this check and causes problems
a couple lines below:
chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1];
Tested with SF64-PCR, but I don't have any of those sound or sound+radio cards
to test if I didn't break anything.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix mute state reporting in tea575x-tuner.
This fixes mute function in kradio on SF64-PCR radio card.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current FSI driver use global variable to access device data.
But this style will be broken
if SuperH come with multiple FSI blocks in future.
To solve this problem, this patch use cpu_dai->private_data.
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>
free the allocated pcm platform device in the error path.
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>
platform_device_unregister() frees resources for us, no need to
do it explicitly. Fixes an oops when machine code removes the
soc-audio device.
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>
Remove snd_opti9xx fields which are indirect arguments to
the snd_opti9xx_configure(). Pass these values as function
arguments.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now snd_ac97_pcm_open() is called with the exactly same arguments
for both playback and capture directions. Remove the unneeded check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Muse Pocket use brocken mixer names, so alsamixer and PA can't use it correctly
This patch add quirk to overwirte default mixers.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
FSC Amilo Pi 1505 has a buggy BIOS and doesn't set up the HP and
speaker pins properly. Add the pinfix entry for that.
Reference: Novell bnc#557403
https://bugzilla.novell.com/show_bug.cgi?id=557403
Signed-off-by: Takashi Iwai <tiwai@suse.de>
pcm->r[1].slots is the double rate slot information, not the
capture information. For capture, 'pcm' will already be the
capture ac97 pcm structure.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA's for-2.6.33 branch has a new source argument to
snd_soc_dai_set_pll().
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add support runtime PM.
Driver callbacks for Runtime PM are empty because
the device registers are always re-initialized after
pm_runtime_get_sync(). The Runtime PM functions replaces the
clock framework module stop bit handling in this driver.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM12 and PCM20 can be set into the ISA PnP mode. The PCM12 PnP
was sold as the PnP device.
Add code to handle detection of these cards using ISA PnP framework.
Tested on the PCM20 in PnP mode. The PCM12 PnP has the same MS Windows
INF file except for a card name displayed for user.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Separate common probing code in order to use it
for PnP probing.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of the irq_req_t typedef'd struct can be re-worked quite
easily:
(1) IRQInfo2 was unused in any case, so drop it.
(2) IRQInfo1 was used write-only, so drop it.
(3) Instance (private data to be passed to the IRQ handler):
Most PCMCIA drivers using pcmcia_request_irq() to actually
register an IRQ handler set the "dev_id" to the same pointer
as the "priv" pointer in struct pcmcia_device. Modify the two
exceptions (ipwireless, ibmtr_cs) to also work this waym and
set the IRQ handler's "dev_id" to p_dev->priv unconditionally.
(4) Handler is to be of type irq_handler_t.
(5) Handler != NULL already tells whether an IRQ handler is present.
Therefore, we do not need the IRQ_HANDLER_PRESENT flag in
irq_req_t.Attributes.
CC: netdev@vger.kernel.org
CC: linux-bluetooth@vger.kernel.org
CC: linux-ide@vger.kernel.org
CC: linux-wireless@vger.kernel.org
CC: linux-scsi@vger.kernel.org
CC: alsa-devel@alsa-project.org
CC: Jaroslav Kysela <perex@perex.cz>
CC: Jiri Kosina <jkosina@suse.cz>
CC: Karsten Keil <isdn@linux-pingi.de>
for the Bluetooth parts: Acked-by: Marcel Holtmann <marcel@holtmann.org>
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes two issues:
a) Infinite loop in resume function
b) Writes to non-existing registers in resume function
Cc: stable@kernel.org
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit fe3e78e073
ASoC: Factor out snd_soc_init_card()
removed the error paths that are still valid for wm97* codecs, causing
the compile errors like
sound/soc/codecs/wm9705.c:399: error: label 'reset_err' used but not defined
sound/soc/codecs/wm9712.c:687: error: label 'reset_err' used but not defined
sound/soc/codecs/wm9713.c:1237: error: label 'reset_err' used but not defined
Revert the removed error path codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
STAC/IDT codecs seem to behave weird when SET_PIN_SENSE verb is issued
before reading the jack-detection although the TRIG_REQ pin capability
is given by the hardware.
Since snd_hda_jack_detect() issues the SET_PIN_SENSE verb simply judging
from the pincap, we have to revert the change in the commit
d56757abc1
ALSA: hda - Replace the rest of jack-detections with snd_hda_jack_detect()
to plain GET_PIN_SENSE verb without triggering.
Reported-by: Jiri Slaby <jirislaby@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If a hardware is not detected there is a kernel crash
due to not initialized snd_miro->aci pointer. This pointer
is initialized after detection of the opti (miro) chip.
This bug was introduced by patches to expose
ACI mikser outside the snd-miro driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The non-cohernet PPC arch doesn't give the correct address by a simple
virt_to_page() for pages allocated via dma_alloc_coherent().
This patch adds a hack to fix the conversion similarly like MIPS.
Note that this doesn't fix perfectly: the pages should be marked with
proper pgprot value. This will be done in a future implementation like
the conversion to dma_mmap_coherent().
Acked-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The non-coherent MIPS arch doesn't give the correct address by a simple
virt_to_page() for pages allocated via dma_alloc_coherent().
Original patch by Wu Zhangjin <wuzj@lemote.com>.
[Ralf mentioned: "The origins of this patch go back far further.
The oldest patch I could find which is a superset of this was written
by Atsushi Nemoto and various incarnations of it have been sumitted
to and reject by me a number of times through the years."]
A proper check of the buffer allocation type was added to avoid the
wrong conversion.
Note that this doesn't fix perfectly: the pages should be marked with
proper pgprot value. This will be done in a future implementation like
the conversion to dma_mmap_coherent().
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
RT workqueue is going away in the near future, replace it with
singlethread wq for now, which is still supported.
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 a helper (inline) function as the default page ops. Any hacks wrt
the page address conversion will be applied in this function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use dma_mmap_coherent() for mmapping the buffers allocated via
dma_alloc_coherent() if available. Currently, only ARM has this function,
so we do temporarily have an ifdef pcm_native.c. This should be handled
better globally in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4792
Cristian reported that these models have really bad sound above 6 dB
and proposed the original patch. I've updated the comment to reflect
this change.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Reported-by: Cristian Klein
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/487884
This Gateway model needs External Amplifier muted for audible playback,
so set the inv_eapd quirk for it.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The chip field is no longer needed. Move those of its fields that are
actually used to the device structure itself.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the dependecy from the USB MIDI code on the snd_usb_audio
structure. This allows using the USB MIDI module from another driver
without having to pretend to be the generic USB audio driver.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mixer control to select between the two altsettings on Roland USB
MIDI devices where the input endpoint is either bulk or interrupt.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell Vostro 1015n uses Conexant CX20583-10Z (0x14f1:5067). Patch is
based on "olpc-xo-1_5" branch. Dell uses digital mic.
Signed-off-by: Einar Rünkaru <einarry@smail.ee>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit f2624791a0.
Łukasz Wojniłowicz reported that the change causes both internal and
external mics not working any more. The headphone jacking issue was
fixed by his previous patch, it's better to revert to acer-aspire-4930g
model.
Reported-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we get a stream suspend event force the power down since otherwise
the stream would remain marked as active. In future we'll probably want
to make this stream-specific and add an interface to make the power down
of other widgets optional in order to support leaving bypass paths
active while suspending the processor.
Cc: stable@kernel.org
Reported-by: Joonyoung Shim <jy0922.shim@samsung.com>
Tested-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the buffer size calculation to use the size which ALSA is expecting.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ACI mixer is used to control the radio FM module
installed on the Miro PCM20 sound card. Expose ACI mixer
outside the sound card driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the miro.h header to the include/sound directory. It can
be used in the Miro PCM20 radio driver (v4l).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the snd_card pointer from the snd_miro structure and
do some small code improvements.
Also, move Opti chipset detection before detection of the
ACI mixer, so the mci_base value is set in one place only.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_SND_JACK needs to be selected explicitly only when INPUT=y or
INPUT_SND. The current way, INPUT=SND_HDA_INTEL isn't strict enough.
Reported-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use hweight16 instead of Brian Kernighan's/Peter Wegner's method
Signed-off-by: Akinobu Mita <akinobu.mita@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
