This is just a quick hack that needs to be removed once the new units
defined by the audio class v2.0 standard are supported.
However, it allows using these devices for now, without mixer support.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a number of parsers for audio class v2.0. In particular, the
following internals are different and now handled by the code:
* the number of streaming interfaces is now reported by an interface
association descriptor. The old approach using a proprietary
descriptor is deprecated.
* The number of channels per interface is now stored in the AS_GENERAL
descriptor (used to be part of the FORMAT_TYPE descriptor).
* The list of supported sample rates is no longer stored in a variable
length appendix of the format_type descriptor but is retrieved from
the device using a class specific GET_RANGE command.
* Supported sample formats are now reported as 32bit bitmap rather than
a fixed value. For now, this is worked around by choosing just one of
them.
* A devices needs to have at least one CLOCK_SOURCE descriptor which
denotes a clockID that is needed im the class request command.
* Many descriptors (format_type, ...) have changed their layout. Handle
this by casting the descriptors to the appropriate structs.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds some definitions for audio class v2.
Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have
different numerical representations in both standards, so there is need
for a _V1 add-on now. usbmixer.c is changed accordingly.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation of support for v2.0 audio class, use the structs from
linux/usb/audio.h and add some new ones to describe the fields that are
actually parsed by the descriptor decoders.
Also, factor out code from usb_create_streams(). This makes it easier to
adopt the new iteration logic needed for v2.0.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables snoop, eliminating static during playback.
This patch supersedes the previous Cougar Point audio patch.
Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the card->codec on soc_resume to detect if the soc
device is properly initialized.
If the card->codec is NULL, than do not continue the resume
operation, since the device is not initialized properly.
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 for having snd_soc_dais shared among two or more dai_links,
remove the relatively global runtime field from the struct snd_soc_dai
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>
Currently only the atmel driver make use of snd_soc_dai.runtime field.
If the dais are to be shared among two or more dai_links, the field
must be got rid of.
So, in atmel driver reach the substream via dai_link->pcm so as to
not depend of snd_soc_dai.runtime field.
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>
Passing pointer to relevant dai_link provides easier reach to the
ASoC tree in suspend/resume of snd_soc_platform. It also provides
direct access to the dai at the other end of the dai_link.
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 a headphones-only quirk for the Fujitsu Siemens D1289.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Marc Haber <mh+alsa201002@zugschlus.de>
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
With PulseAudio and an application accessing an input device like `gnome-volume-manager` both have high CPU load as reported in [1].
Loading `snd-hda-intel` with `position_fix=1` fixes this issue. Therefore add a quirk for ASUS M2V-MX SE.
The only downside is, when now exiting for example MPlayer when it is playing an audio file a high pitched sound is outputted by the speaker.
$ lspci -vvnn | grep -A10 Audio
20:01.0 Audio device [0403]: VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10)
Subsystem: ASUSTeK Computer Inc. Device [1043:8290]
Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx-
Latency: 0, Cache Line Size: 64 bytes
Interrupt: pin A routed to IRQ 17
Region 0: Memory at fbffc000 (64-bit, non-prefetchable) [size=16K]
Capabilities: <access denied>
Kernel driver in use: HDA Intel
[1] http://sourceforge.net/mailarchive/forum.php?thread_name=1265550675.4642.24.camel%40mattotaupa&forum_name=alsa-user
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the standard unsol_event callback with each setup callback for
IntelMac models with Realtek ALC885 codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/343989
Add a model quirk for the NVIDIA based Macmini hardware, aka Macmini 3,1. The
pinout is almost identical to the mb5 quirk, except for no microphone and
the line-in mixer controls being on a different index. Everything works in
2ch mode, but as I am not sure what needs to be changed for 6ch mode, or
whether the Mac Mini's chip supports 6ch mode, I have simply duplicated
the code from the mb5 quirk for the mac mini chmode management. The new
model parameter for this quirk is "macmini3".
Signed-off-by: Luke Yelavich <luke.yelavich@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/524948
The OR has verified that the existing model=laptop-eapd quirk does not
function correctly but instead needs model=3stack. Make this change
so that manual corrections to module-init-tools file(s) are not
required.
Reported-by: Lasse Havelund <lasse@havelund.org>
CC: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_cs46xx_codec_reset() bypassing the register cache, so as to not
clobber the cached register value during resume.
Signed-off-by: Florian Zumbiehl <florz@florz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Disable the amplifiers for the headset outputs, and do not select
routings by default to the headset outputs.
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 a division by zero error in the irq handler.
There is a small window between the hw_params() callback and when
runtime->frame_bits is set by ALSA middle layer. When another substream is
already running, if an interrupt is delivered during that window the irq
handler calls pcm_pointer() which does a division by zero. The patch below
makes the irq handler skip substreams that are initialized but not started
yet. Cc to Clemens Ladisch because he proposed an alternate fix.
For more information, please read the original thread in the linux-kernel
mailing list: http://lkml.org/lkml/2010/2/2/187
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
OSCSET calculation was not correct in case of 44.1KHz
sampling rate.
With small adjustment both 48 and 44.1 KHz calculation
now gives the correct value.
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 repeated playback the FIFOFLUSH bit remained set, and
never has been cleared.
Clear it during the setup phase.
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>
Provide a sysfs file allowing userspace to inspect and change the
pmdown_time setting at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Make the pmdown_time a per-card setting rather than a global one,
initialised before the card initialisation runs. This allows cards
to override the default setting if it makes sense to do so (for
example, due to an unavoidable pop).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The usbmixer proc file contains mapping between ALSA control API and
USB mixer control units. The purpose of this file is for debugging
and a problem diagnostics.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Here's a patch that adds MIDI support through USB for one of the Access
Music synths, the VirusTI.
The synth uses standard USBMIDI protocol on its USB interface 3, although
it does signal "vendor specific" class. A magic string has to be sent on
interface 3 to enable the sending of MIDI from the synth (this string was
found by sniffing usb communication of the Windows driver). This is all
my patch does, and it works on my computer.
Please note that the synth can also do standard usb audio I/O on its
interfaces 2&3, which already works with the current snd-usb-audio driver,
except for the audio input from the synth. I'm going to work on it when I
have some time.
Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> (cosmetics, list terminator)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Extend the list of devices whose firmware does not expect more than one
USB MIDI packet in one USB packet.
bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3752
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The MSI blacklist entry for ASUS mobo added in the commit
8ce28d6abf was based on the alsa-info
output wrongly posted. Fix the id to the right one now.
Reported-by: Sid Boyce <sboyce@blueyonder.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds rearranges parts of the initialization code and adds
suspend and resume callbacks.
This patch adds suspend and resume callbacks.
It also rearranges parts of the initialization code so it can be
used in both the first initialization (when the module is loaded we
also have to load default settings) and the resume callback (where
we have to restore the previous settings).
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the controls init code outside the init_hw() function because is must
not be called during resume.
This patch moves the code that initializes the card's controls with
default valued from the init_hw() function into a separated
set_mixer_defaults() function (one for each of the 16 supported
cards). This change is necessary because during resume we must
resurrect the hardware without losing the previous
settings. set_mixer_defaults() must be called only once when the
module is loaded.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements a simple cache for the firmware files when CONFIG_PM is defined.
This patch changes get_firmware(), free_firmware() and adds
free_firmware_cache(). The first two functions implement a very
simple cache and the latter is used to actually release all the stored
firmwares when the module is unloaded.
When CONFIG_PM is not enabled those functions act as before, that is
free_firmware() releases the firmware immediately and
free_firmware_cache() does nothing.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changes the way the firmware is passed through functions.
When CONFIG_PM is enabled the firmware cannot be released because the
driver will need it again to resume the card.
With this patch the firmware is passed as an index of the struct
firmware card_fw[] in place of a pointer. That same index is then used
to locate the firmware in the firmware cache.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add patch for the Conexant 5066 HDA codec to support the Lenovo IdeaPad U150
Signed-off-by: Greg Alexander <greigs@galexander.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the zero-division warning message with WARN_ON_ONCE() per the
advice by Linus. This shouldn't happen, but if it happens, it's
possible that the bug happens often due to buggy IRQs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM2000 is a low power, high quality handset receiver speaker
driver with Wolfson myZone™ Ambient Noise Cancellation (ANC). It
provides enhanced voice communication quality in a noisy environment
if the handset acoustics are designed appropriately.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Audio on Migo-R cannot work if CONFIG_SH_DMA_API=y, but compilation should not
break anyway.
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>
This patch expands the omap3beagle sound soc for the
beagle board clone DevKit8000.
Signed-off-by: Thomas Weber <weber@corscience.de>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
USB devices tends to represent dB ranges in different way than ALSA expects.
Add possibility to override these values and add guessed values for
SoundBlaster MP3+.
Also rename 'Capture Input Source' control to 'Capture Source' for
SoundBlaster MP3+ and Extigy.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The serial interface (TDM/I2S) for the audio block have been
constantly enabled.
Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so
the interface is only enabled, when there is a need for it.
For example when only the analog loopback is enabled, there
is no need to keep the serial interface active.
I have added the persons who contributed to the Voice path
of twl4030 codec driver, so they might have the ability
to test this patch, and send an update for the Voice path,
if it is necessary
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>
Enable the bulk regulators at probe time so we can safely disable them
again when going to suspend without confusing the reference counter.
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 adds sound support for Phytec PhyCORE / PhyCARD
modules in AC97 mode.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On my AMD780V chipset, hda_intel.c can crash the kernel with a divide by
zero
for as-yet unknown reasons. A simple check for zero prevents it, though
the problem that causes it remains. Since the workaround is harmless and
won't affect anyone except victims of this bug, it should be safe;
moreover,
because this crash can be triggered by a user-mode application, there are
denial of service implications on the systems affected by the bug without
the patch.
Signed-off-by: Jody Bruchon <jody@nctritech.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
machine is compatible is an OF-specific call. It should have
the of_ prefix to protect the global namespace.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Michal Simek <monstr@monstr.eu>
In particular, several occurances of funny versions of 'success',
'unknown', 'therefore', 'acknowledge', 'argument', 'achieve', 'address',
'beginning', 'desirable', 'separate' and 'necessary' are fixed.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Joe Perches <joe@perches.com>
Cc: Junio C Hamano <gitster@pobox.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Use DEFINE_PCI_DEVICE_TABLE() to make PCI device ids go to
.devinit.rodata section, so they can be discarded in some cases,
and make them const.
Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit caused a regression on HP laptops with 92HD83x/88x
codecs. The default polarity of mute-LED GPIO is inverted on these
devices.
Reference: Novell bnc#578190
https://bugzilla.novell.com/show_bug.cgi?id=578190
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have now a better mute-LED GPIO detection, and no need to assign the
values statically per model option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge the mute-LED status callback function for both IDT 92HD7x and 8x
codecs to one function. Also it's changed to check all DACs, and called
in the initialization to sync with the current status.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The GPIO pin number for the mute LED control on HP laptops can be
determined more easily by checking the number of available GPIO pins
of the codec chip. On a small package with up to 3 GPIOs, GPIO 0 is
used while GPIO 3 is used for others.
This fixes the missing mute GPIO for some HP laptops with new codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pandora's external DAC is connected to VSIM TWL4030 supply, so let's
start switching it too to save more power.
Also DAC got it's own DAPM handler.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Pandora's external DAC is using 256*Fs output from the TWL4030
codec, and TWL4030 needs to have APLL enabled for it's 256*Fs
output to function.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.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>
The volume register is from 0..0x7f and 0..0x1a range is mute.
Also, fix mute combining in wm_vol_put(). The wrong behaviour was
noticed by Peter Christensen.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This renames the interrupt name in /proc/interrupt.
HDA Intel -> hda_intel
This also eliminates space from the name, probably helping some
parsers.
Don't think anybody depends on this name in userspace
Signed-off-by: Takashi Iwai <tiwai@suse.de>
My sound codec seems sometimes (very rarely) to omit interrupts (ALC268)
However, interrupt mode still works.
Thus if we get timeout, poll the codec once.
If we get 3 such polls in a row, then switch to polling mode.
This patch is maybe an bandaid, but this might be a workaround for hardware bug.
Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I found that the sampling rate locking setting of the ice1712 sound driver
was only half-respected : when the driver was locked to, let's say, 44100Hz,
and a usermode app was requesting 48000Hz playback, the request was succesful
although the soundcard would continue to run at 44100Hz.
Here's a patch that will make those requests to fail.
Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After hours of debugging, I finally found the reason why some source
and runtime combination does not work. The PTP (page table pages)
address must be aligned. I am not sure how much, but alignment to
PAGE_SIZE is sufficient. Also, use ALSA's page allocation routines
to ensure proper virtual -> physical address translation.
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- Add support for ALC665
- Add more ASUS model
- Modify common patch for ALC272 ALC273 ALC661 ALC662 ALC663 ALC665
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add new models ALC269VB_AMIC ALC269VB_DMIC
- Add alc269vb_laptop_dmic_setup
The record source index Dmic is 0x6 for ALC269VB.
- Change eeepc words for ALC269
- Modify init_verb tables of patch_alc269 patch_alc662 patch_alc882
- Modify common patch for ALC270 ALC269VB ALC275
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The module unloading path had several problems:
- it freed up the private structure twice
- it freed up the codec structure, which was allocated as part
of the private structure
- it did not freed up the reg_cache
- it did not unregistered the dais and the codec
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 WM8912 is a DAC only device register compatible with the WM8904
CODEC with ADC portions omitted. Support it within the WM8904 driver
based on the configured I2C device name.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Handle the output PGAs as part of the output powerup since they can
never be powered separately and reorder things so that we remove the
output shorts after both line and headphone outputs have been brought
up, minimising the opportunity for any issues.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
As well as disabling the biases of the CODEC the drop into BIAS_OFF will
also disable all the regulators powering the CODEC, allowing even greater
power savings on appropriately configured systems.
Since the regulator API does not currently provide notification when
regulators are disabled we assume that this always happens when we stop
using the regulators. Once 2.6.34 is merged this code can be optimised
to only sync the cache when power was actually removed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
While the regulators are disabled we cache all register writes.
Currently we assume that the regulator disable actually takes
effect, after the merge with the regulator tree in 2.6.34 the
regulator API will be able to notify us if the power is actually
removed (due to constraints or regulator sharing it may not be).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
At the minute the regulators are simply enabled for the entire
lifetime of the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add a bit to the CODEC structure indicating if a cache sync is required.
By default this will be set if a cache only write is done to a soc-cache
register cache. This allows us to avoid syncing the cache back after
using cache only writes if there were no changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch fixes docking output support for IDT 92HD81/83/88 family codecs.
Typically one of ports 0xE or 0xF is used for docking output, while only
port 0xF is common on all the three codec families. We don't want the
pin to select the analog mixer here.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the soc-cache code will always write to the device, meaning
that we need the device to be powered and active at pretty much all
times the system is active. Allowing cache only writes lays some
groundwork for future enhancements to allow devices to be put into a
full off state when the audio subsystem is idle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
ctrl_outl() has become void at some point, which breaks compilation of fsi.c.
Make writing functions void, as their output is anyway not evaluated, and use
__raw_writel and __raw_readl instead of deprecated ctrl_outl and ctrl_inl
respectively.
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>
This is a cleanup for the dummy driver. The model kernel module parameter
is introduced to select the soundcard emulation.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Unconditionally save the register states when suspending and restore
them again at resume time. Register contents were not preserved over
suspend, and hence the driver takes false assumptions about them.
The clock must be enabled to access the register block.
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>
String constants that are continued on subsequent lines with \
are not good.
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The dependency on MFD_WM8994 rather than I2C went awry.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Instead of padding with blanks and printing "number=0x a", print
"number=0x0a".
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8994 is a highly integrated ultra-low power hi-fi audio subsystem
designed for smartphones and other portable devices rich in multimedia
features. It provides advanced digital mixing facilities enabling low
power high quality interconnection of CPU, baseband and other audio
sources through flexible digital and analogue routing, and integrates
a class W headphone driver and stereo class D speaker drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use a two code correction for optimal performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Perform DC servo offset calibration using a series update sequence
rather than startup update sequence, tuning the configuration of the
WM8993 DC servo to make best use of this.
Also introduce currently unused data allowing us to correct for
any systematic errors in the DC servo calibration results and an
alternative startup path for the headphone output which performs
better with some chip revisions. The alternative setup sequence is
enabled for WM8993.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
String constants that are continued on subsequent lines with \
are not good.
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In case, if OPCLK is not used, and PLL is used for driving the codec, the
choice of PLL output frequency could result in a needlessly imprecise
system clock frequency. Use an iterative process to select a precise
configuration.
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>
Allow TLV blocks that do not have any values; the smallest possible TLV
is an empty container or one where the information is only in the tag.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Creating a control with TLV_COMMAND access was not possible because
snd_ctl_new1() forgot to include it in the mask of allowable access
bits.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
wm8974 and wm8978 codec drivers control DAC and ADC oversampling rates in their
.set_clkdiv() methods, which is wrong, because these are simple boolean
switches and not clock dividers. Move these bits to sound controls. Also remove
manual configuration of the MCLK divider in wm8978, since it is configured
automatically.
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>
Tests showed, that bit 6 of the WM8978_POWER_MANAGEMENT_2 register of wm8978
affects codec clocks. Being useless for suspend / resume, it cannot be used in
bias-level control either. Remove this bit handling.
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>
Add support for audio on sh7722-based Migo-R boards, using SIU and wm8978
codec, recording via external microphone and playback via headphones are
implemented.
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>
It's more robust when references are provided in control names
rather than numid.
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>
Commit 761c9d45 (ASoC: Fix build of OMAP sound drivers) changes
CONFIG_MACH_OMAP3517EVM -> CONFIG_SND_OMAP_SOC_OMAP3517EVM in the
Makefile. Whereas the config option defined in Kconfig is
SND_OMAP_SOC_AM3517EVM. Because of this, ASoC driver for AM3517
was not getting compiled.
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit e9ff5eb2 (Fixing infinite loop in resume path) uses wrong AIC23
register in resume function because of which register writes happen
on some non-existing registers.
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>
In struct device_node, the phandle is named 'linux_phandle' for PowerPC
and MicroBlaze, and 'node' for SPARC. There is no good reason for the
difference, it is just an artifact of the code diverging over a couple
of years. This patch renames both to simply .phandle.
Note: the .node also existed in PowerPC/MicroBlaze, but the only user
seems to be arch/powerpc/platforms/powermac/pfunc_core.c. It doesn't
look like the assignment between .linux_phandle and .node is
significantly different enough to warrant the separate code paths
unless ibm,phandle properties actually appear in Apple device trees.
I think it is safe to eliminate the old .node property and use
phandle everywhere.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: David S. Miller <davem@davemloft.net>
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
This patch adds HP mute LED support for IDT 92HD81/3 family of the codecs.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amp used for the mic input on HP Compaq F700 with Cxt5051 codec
has no multiple inputs, thus its index should be 0 instead of 1.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In hda_codec.c, it has define
"[HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 },",
it support up to device 9 for HDMI.
But in hda_intel.c, it only define AZX_MAX_PCMS as 8.
So if it have 4 hdmi codecs, when run azx_attach_pcm_stream(),
it will show error "Invalid PCM device number 8", and "... number 9",
and return "-EINVAL".
We should change the AZX_MAX_PCMS to 10.
Signed-off-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Also renames a few things to make volumes and switches match up in
alsamixer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The version isn't being updated or used, the kernel revision
tracking is enough.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Change the legacy default register configuration, which left some
internal components on.
Now we have either DAPM, or other ways to control these bits,
so there is no need to enable them by default.
The affected parts:
Disable ADCL and ADCR
Disable ARXL2 and ARXR2 analog PGA (playback)
Disable APLL by default
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>
fsi_master_xxx function should be protected by spin lock,
because it are used from both FSI-A and FSI-B.
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>
Allow the override of vendor-id, subsystem-id, revision-id and chip name
via patch loading. Updated the document, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8978 codec from Wolfson Microelectronics is very similar to
wm8974, but is stereo and also has some differences in pin configuration
and internal signal routing. This driver is based on wm8974 and takes
the differences into account.
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 board supports both GPIO sets for the AC97 bus and the analogue
outputs can be switched between this and the WM8580 so add some
comments saying what the setup the standard kernel expects is.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since, we have generic AC97 controller driver and all the machines
have moved to that, there is no need for old s3c2443-ac97.c to exist.
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>
Switch to use s3c-ac97.c AC97 controller driver.
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>
Switch to use s3c-ac97.c AC97 controller driver.
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>
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>
