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Without calling an appropriate rule, AES/AES32 cards would announce a
theoretical channel count of 64 (HDSPM_MAX_CHANNELS), leading to the
already known bug:
[37422.640481] ------------[ cut here ]------------
[37422.640487] WARNING: at sound/pci/rme9652/hdspm.c:5449
snd_hdspm_ioctl+0x18f/0x202 [snd_hdspm]()
[37422.640489] Hardware name: PRIMERGY RX100 S6
[37422.640490] BUG? (info->channel >= hdspm->max_channels_in)
[37422.640492] Modules linked in: snd_hdspm snd_seq_midi ipmi_watchdog
ipmi_poweroff ipmi_si ipmi_devintf ipmi_msghandler i2c_i801 e1000e
snd_rawmidi power_meter [last unloaded: snd_hdspm]
[37422.640501] Pid: 22231, comm: jackd Tainted: G D W
2.6.36-gentoo-r5 #5
[37422.640502] Call Trace:
[37422.640508] [<ffffffff8103db3a>] warn_slowpath_common+0x80/0x98
[37422.640511] [<ffffffff8103dbe6>] warn_slowpath_fmt+0x41/0x43
[37422.640514] [<ffffffff81034306>] ? get_parent_ip+0x11/0x42
[37422.640518] [<ffffffffa0055763>] snd_hdspm_ioctl+0x18f/0x202
[snd_hdspm]
[37422.640522] [<ffffffff813fd626>] snd_pcm_channel_info+0x73/0x7c
[37422.640525] [<ffffffff814001e9>] snd_pcm_common_ioctl1+0x326/0xb01
[37422.640527] [<ffffffff81034306>] ? get_parent_ip+0x11/0x42
[37422.640531] [<ffffffff8105be6c>] ? __srcu_read_unlock+0x3b/0x59
[37422.640533] [<ffffffff81400bce>] snd_pcm_capture_ioctl1+0x20a/0x227
[37422.640537] [<ffffffff811e599c>] ? file_has_perm+0x90/0x9e
[37422.640540] [<ffffffff81400c15>] snd_pcm_capture_ioctl+0x2a/0x2e
[37422.640543] [<ffffffff810f2c69>] do_vfs_ioctl+0x404/0x453
[37422.640546] [<ffffffff810f2d09>] sys_ioctl+0x51/0x74
[37422.640549] [<ffffffff81002aab>] system_call_fastpath+0x16/0x1b
[37422.640552] ---[ end trace 0cd919cd68118082 ]---
We already have all the right values in place, we simply have to inform
the upper layers about this restriction.
Note that snd_hdspm_hw_rule_rate_out_channels and
snd_hdspm_hw_rule_rate_in_channels must not be called on AES32, because
the channel count is always 16, no matter of the samplerate in use.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only RayDAT and AIO provide sane buffer pointers that can be used with
HDSPM_BufferPositionMask, on all other cards, this would result in a
wrong HW pointer leading to xruns and these messages:
[260808.916788] BUG: pcmC0D0p:0, pos = 2976, buffer size = 1024, period size = 512
[260808.961124] BUG: pcmC0D0c:0, pos = 4944, buffer size = 1024, period size = 512
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a USB audio device is disconnected, snd_usb_audio_disconnect()
kills all audio URBs. At the same time, the application, after being
notified of the disconnection, might close the device, in which case
ALSA calls the .hw_free callback, which should free the URBs too.
Commit de1b8b93a0ba "[ALSA] Fix hang-up at disconnection of usb-audio"
prevented snd_usb_hw_free() from freeing the URBs to avoid a hang that
resulted from this race, but this introduced another race because the
URB callbacks could now be executed after snd_usb_hw_free() has
returned, and try to access already freed data.
Fix the first race by introducing a mutex to serialize the disconnect
callback and all PCM callbacks that manage URBs (hw_free and hw_params).
Reported-and-tested-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
Cc: <stable@kernel.org>
[CL: also serialize hw_params callback]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The default WM8958 jack detection handler implements a full set of buttons
and also support for video detection. Support for multi-button jacks is
fairly system specific and will usually require some tuning for headsets
so simplify the implementation to only report a simple short to ground
button, leaving multi-button headsets to be handled by system specific
code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8958 has a different microphone bias architecture to WM8994 so needs
different configuration to WM8994. Support this in platform data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow direct routing of the WM8958 microphone detection signal to a GPIO
to be used, saving the need to demux the interrupt.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
As bias level changes can be quite time consuming and the bias changes
for multiple devices aren't strongly tied to each other (if anything it
can be advantageous to bring different devices up together) we can improve
the state transition time for multi-component systems by running the bias
level changes for all the devices in parallel. This is very simple to
achieve using the kernel async functionality so use that to schedule the
work.
This should have no practical effect for the overwhelming majority of
systems which have a single DAPM context - we'll bounce into another
thread to do the bias level change but otherwise everything will happen
in exactly the same order as it did before.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
We can get the card from the DAPM context so don't bother passing it as
an argument.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The microphone detection functionality requires a clock to work. In any
non-detection case where the MICBIAS is enabled CLK_SYS will be needed
anyway so there is no negative impact on power consumption.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: Ensure supplies are maintained for force enabled widgets
ASoC: WM8994: Improve playback robustness
ASoC: WM8994: Improve robustness in some use cases
ASoC: WM8903: Fix mic detection enable logic
ASoC: WM8903: Fix mic detection register definitions
ASoC: CX20442: fix wrong reg_cache_default content
ASoC: Sync initial widget state with hardware
This typo caused some microphone inputs not to be correctly
initialized on VIA codecs.
Reported-By: Mark Goldstein <goldstein.mark@gmail.com>
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Si4713 FM transmitter on Nokia RX-51/N900 is connected to same Line out
signals of TLV320AIC34 than TPA6130 headphone amplifier.
This patch adds route to transmitter and "FM Transmitter" control to keep
route active when needed.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch changes dependency of ARCH_EXYNOS4 from ARCH_S5PV310
according to the change of ARCH name, EXYNOS4.
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Kukjin Kim <kgene.kim@samsung.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: HDA: Do not announce false surround in Conexant auto
ALSA: HDA: Conexant auto: Handle multiple connections to ADC node
ALSA: HDA: Add position_fix quirk for an Asus device
ALSA: caiaq - Fix possible string-buffer overflow
ALSA: au88x0 - Modify pointer callback to give accurate playback position
Add " Playback Volume" to 10 bands Equalizer Controls of au88x0 so that
alsa-lib won't regard them as "Capture Volume".
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this patch, one line-out and one speaker and
Conexant's auto parser would announce (non-working) surround
capabilities.
BugLink: http://bugs.launchpad.net/bugs/721126
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conexant 20641 has several inputs to its ADC node, with one selector
and individual amps for all inputs. This patch adds support in the
Conexant auto parser to handle that case.
It also means that the pin node's volume is being renamed to "Boost"
to avoid name clash with the new volume controls on the ADC node.
BugLink: http://bugs.launchpad.net/bugs/719524
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make newly created AC97 emulation of azt3328 known to the AC97 layer
side.
- relocate common functions to the top (due to definition after use)
- rename control names
- adjust 3D settings to the card's custom layout of this register
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make use of the very flexible ALSA ac97 layer (hooks for custom I/O!)
on this weird AC97 copycat hardware,
via semi-extended I/O translation/emulation.
Some 5kB binary/loaded size saved (well... additional huge AC97 module
penalty not factored in, of course ;-P).
Given that the driver previously had 20kB that's not bad,
but the much more important thing is to have AC97 layer stress-tested
with a thoroughly weird AC97 copycat (or, simply put, if it were not for
this AC97 test aspect, this effort would merely have been a nut job ;).
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The microphone detection functionality requires a clock to work. In any
non-detection case where the MICBIAS is enabled CLK_SYS will be needed
anyway so there is no negative impact on power consumption.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems wish to use jacks as wake sources. Provide a wake flag in the
GPIO configuration which causes the driver to enable the IRQ as a wake
source.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The only thing that should ever be calling this is soc-core and that is
built as part of the same module so doesn't need the export.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Use AC97 macros (sometimes already existing, or newly added)
instead of error-prone repetition of open-coded values.
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pulseaudio doesnt work with current driver and it was root caused to absense of
hw_params function and malloc_pages in it.
This patch adds this and allows pa to work fine with these drivers
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removes the old method of soc-audio device creation in mfld machine
and makes use of new soc_register_card API to register the card
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The tlv scale is defined as (min, step, mute). The mute is not supported here so
put the value to 0
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch makes the DMIC dynamically connect to TX Mux, earlier code had
erroneously made this as static path
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch makes the codec playback rails (headset and speaker) depend on
actual pins they control. This enables better power management of the codec
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is more usefull to report headset instead of video out cable in response
to jack insertion as this is more usual use-case and because now the headset
feature is supported. Automatic accessory detection is not possible at the
moment so most sensible static accessory type have to be used.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for headset microphone in Nokia RX-51/N900. The mic
signal from audio jack is routed to codec A LINE1L via two switches and the
mic bias is coming from codec B part.
First switch is the tv-out switch that is already supported and the second
switch selects between voltage detection circuit and codecs. As there is
no use for voltage detection at the moment the second switch is connected
statically to codecs in rx51_soc_init.
Headset can be active when control "Jack Function" is set to "Headset".
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Just slight cleanup to be sync with upcoming change.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The bug reporter claims that position_fix=1 is needed for his
microphone to work. The controller PCI vendor-id is [1002:4383] (rev 40).
Reported-by: Kjell L.
BugLink: http://bugs.launchpad.net/bugs/718402
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use strlcpy() to assure not to overflow the string array sizes by
too long USB device name string.
Reported-by: Rafa <rafa@mwrinfosecurity.com>
Cc: stable <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The number of cases has increased so use switch-case rather than
if-statements.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The MK2 generation of Native Instruments' sound cards are in fact
compliant to the USB audio standard of version 2 and other approved USB
standards. However, they come up as vendor-specific device when first
connected but can be told to come up with a new set of descriptors
upon their next enumeration. The interfaces announced by the new
descriptors will be handled by the kernel's class drivers. This is done
by issuing a vendor specific device request and sending the device to
reset.
There are also some vendor-specific USB requests for some mixer elements
that can't be exported in a standard compliant way. The driver now
supports them with quirks handling mechanisms.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the core code where sparse complains. In most cases, this means
just adding annotations to confirm that we indeed want to do the dirty
things we're doing.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
