Don't call snd_jack_report at release of sigmatel and conexnat codecs
which results in Oops at unloading the module.
The Oops is triggered by the power-up sequence during the free due to
the pincfg restoration. Since the power-up sequence is involved with
the unsol handling, the jack reporting may be issued during that.
The Oops occurs with this jack reporting because the jack instances
have been already released but the codec doesn't do the proper
book-keeping.
This patch adds the book-keeping of jack instances to avoid the access
to bogus pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the second go through of the old DMA_nBIT_MASK macro,and there're not
so many of them left,so I put them into one patch.I hope this is the last round.
After this the definition of the old DMA_nBIT_MASK macro could be removed.
Signed-off-by: Yang Hongyang <yanghy@cn.fujitsu.com>
Cc: Russell King <rmk@arm.linux.org.uk>
Cc: Tony Lindgren <tony@atomide.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: James Bottomley <James.Bottomley@HansenPartnership.com>
Cc: Greg KH <greg@kroah.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
It seems that the zero value from the PICB (position in current buffer)
register is not reliable. Use jiffies to correct returned value
from the ring buffer pointer callback.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- use monotonic posix clock to measure time
- try to avoid reading zero from PICB (position in current buffer) register
- show also measured samples
- when clock is near 41000 or 44100, use exactly these values
(they appears to be reference clocks for hardware manufacturers)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
I found two issues with ICH7-M (it should be related to other HDA chipsets
as well):
- the ring buffer position is not reset when stream restarts (after xrun) -
solved by moving azx_stream_reset() call from open() to prepare() callback
and reset posbuf to zero (it might be filled with hw later than position()
callback is called)
- irq_ignore flag should be set also when ring buffer memory area is not
changed in prepare() callback - this patch replaces irq_ignore with
more universal check based on jiffies clock
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (36 commits)
ALSA: hda - Add VREF powerdown sequence for another board
ALSA: oss - volume control for CSWITCH and CROUTE
ALSA: hda - add missing comma in ad1884_slave_vols
sound: usb-audio: allow period sizes less than 1 ms
sound: usb-audio: save data packet interval in audioformat structure
sound: usb-audio: remove check_hw_params_convention()
sound: usb-audio: show sample format width in proc file
ASoC: fsl_dma: Pass the proper device for dma mapping routines
ASoC: Fix null dereference in ak4535_remove()
ALSA: hda - enable SPDIF output for Intel DX58SO board
ALSA: snd-atmel-abdac: increase periods_min to 6 instead of 4
ALSA: snd-atmel-abdac: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: cleanup registers when removing driver
ALSA: snd-atmel-ac97c: do a proper reset of the external codec
ALSA: snd-atmel-ac97c: enable interrupts to catch events for error reporting
ALSA: snd-atmel-ac97c: set correct size for buffer hardware parameter
ALSA: snd-atmel-ac97c: do not overwrite OCA and ICA when assigning channels
ALSA: snd-atmel-ac97c: remove dead break statements after return in switch case
ALSA: snd-atmel-ac97c: cleanup register definitions
...
Replace all DMA_24BIT_MASK macro with DMA_BIT_MASK(24)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_28BIT_MASK macro with DMA_BIT_MASK(28)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_30BIT_MASK macro with DMA_BIT_MASK(30)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_31BIT_MASK macro with DMA_BIT_MASK(31)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_32BIT_MASK macro with DMA_BIT_MASK(32)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Add powerdown sequence for VREF using a shared jack when the headphone
is present and the microphone isn't on.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC889 has two SPDIF outputs: 0x06, 0x10. Board vendors can use either or both.
DX58SO uses 0x10, but the driver assumes 0x06. The safe solution is to add
0x10 as slave output to the existing 0x06.
Reported-by: Jeroen Van Breedam <jeroen.vanbreedam@sgr5.be>
Tested-by: Jeroen Van Breedam <jeroen.vanbreedam@sgr5.be>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added device id in struct for codec 92HD81B1C (0x111d76d5).
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk model=acer-aspire for Acer Ferrari 5000 with ALC883 codec.
Note that model=auto doesn't work for this laptop because of broken BIOS
(that doesn't set the subsystem id properly).
Tested-by: Russ Dill <russ.dill@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace with the standard function calls to use caches for reading
the widget caps and pin caps.
hda_proc.c is still using the direct verbs to get raw values as
much as possible.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In patch_realtek.c, don't create empty or single-item "Input Source"
control elements that are simply superfluous.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check for the amp-output must be done for widget-caps rather than
pin-caps as implemented in the recent change... Simply a thinko.
Also, add the similar checks to all places that put output-amp mutes
in the initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_hda_query_pin_caps() to read and cache pin-cap values
to avoid too frequently issuing the same verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't set amp-out values to pins without PINCAP_OUT capability,
which are usually assigned for digital mics on ALC663/ALC272.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch does two things:
Output Intel HDA Function Id in /proc/asound/cardX/codec#X
Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros
Before:
Vendor Id: 0x11d41884
Subsystem Id: 0x103c281a
Revision Id: 0x100100
After:
Function Id: 0x1
Vendor Id: 0x11d41884
Subsystem Id: 0x103c281a
Revision Id: 0x0100100
As report on the Kernel Bugzilla #12888
Signed-off-by: Pascal de Bruijn <pascal@unilogicnetworks.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the detection of digital-mic inputs on ALC663 / ALC272 codecs
in the auto-detection mode. The automatic mic switch via plugging
isn't implemented yet, though.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the prepare callback is called multiple times, BDL entries
are reset and re-programmed at each time.
This patch adds the check to avoid the reset of BDL entries when the
same parameters are used.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is an omitted unlock in one snd_mixart_hw_params fail path. Fix it.
Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If ratesp or formatsp values are zero, wrong values are passed to ALSA's
the PCM midlevel code. The bug is showed more later than expected.
Also, clean a bit the code.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The position-buffer on ATI controllers are unreliable as well as
on VIA chips, thus the same workaround for DMA position reading as
VIA is useful for ATI.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ATI controllers (at least some SB0600 models) appear buggy to handle
64bit DMA. As a workaround, reset GCAP bit0 and let the driver to
use only 32bit DMA on these controllers.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit breaks the (digital-) beep on ALC662.
ALC662 has the connection index 0x05 while ALC662 and ALC272 have the
index 0x04 for the beep widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC662/663 codecs have Beep Amplifier Index 0x04 not 0x05 in 0x0b NID.
Confirmed by testing on real hardware.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With this patch the drivers do not set the vmixer volume anymore at startup
because it is actually the output volume of the voices and ALSA mandates
that the volume must be 0 by default.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a long standing bug in the drivers for cards with a vmixer because
I overlooked a detail in the c++ generic driver by echoaudio. Those cards
do not have a line-out volume control. It is a virtual control provided by
the generic driver. The bug is harmless because the DSP just ignores the
command to change the volume.
*NB:* It breaks alsa-tools/echomixer. A patch for it will follow.
This patch removes the line-out volume control from vmixer-equipped cards.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the power state of each widget before starting the initialization
work so that all verbs are executed properly.
Also, keep power-up during hwdep reconfiguration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update the places where the 0x1d widget is used for Conexant 5047, fixing
mismatch introduced after changing the connection.
Signed-off-by: Gregorio Guidi <gregorio.guidi@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up Conexant 5047 pareser code:
- Split mixer elements to separate arrays to reduce the duplicated
entires
- Fix mixer element names to the standard ones
- Remove unneeded cxt5047_hp2_unsol_event; the normal unsol_event
handler works fine.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the initial connections of output pins 0x13 and 0x1d for Conexant
5047 codec to point to the mixer amp properly.
Removed unneeded (doubly) verbs from arrays, also removed the unneeded
changing of widget 0x1c, which is now completely unused.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove superfluous verbs from cxt5047_toshiba_init_verbs[].
Also fix comments and minor coding style issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create "Capture Source" control dynamically for Conexant codecs.
If only one capture item is available, don't create such a control
since it's just useless.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of binding volumes, create a virtual master volume for Conexant
codecs. This allows separate HP and speaker volume controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Code rework, comments of mail tiwai@suse.de (2009-03-09) incorporated.
Code tested on HP HDX16 (not tested on HDX18 yet).
Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added codec recognition of HP HDX platforms and added support of the
MUTE LED (orange/white). For this feature the CONFIG_SND_HDA_POWER_SAVE
is needed to use event handling for mute control.
Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the HT-Omega Claro (halo) sound cards, the headphone amplifier must
be enabled explicitly by setting a GPIO bit.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Fix headphone-detect regression with multiple HP jacks
ALSA: hda - Fix typos in slave controls in patch_sigmatel.c
Assign DACs to HP and speaker before mic-in/line-in shared outputs.
This improves the usability as it results in more intuitive mixer
names.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In stac92xx_auto_fill_dac_nids[], connect to the primary DAC if no
individual DAC is available for each pin. This ensures that the pin
works somehow at least.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create multiple "Headphone" and "Speaker" controls with non-zero index
numbers instead of "Headphone2", etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Improve the parser to pick up more intuitive control names for the
outputs judging from the pin type, instead of fixed names assigned
to channels.
Also, revive the multi-HP workaround since this change fixes the
problem with the multi-HP detection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent changes over the DAC detection mechanism in patch_sigmatel.c
breaks the HP detection on the machines with multiple HP jacks.
It's basically because of the workaround to support the multi-channel
output. Since the HP detection is more important feature, disable
the HP-swap workaroud temporarily.
Reference: Novell bnc#482052
https://bugzilla.novell.com/show_bug.cgi?id=482052
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"Headphone Playback ..." appears twice in slave_vols[] and slave_sws[].
They should be "Headphone Playback2 ..."
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Noises can be heard on analog outputs of (some model of) Lenovo
Ideapad due to the hardware problem, and the only workaround right now
is to fix the sample rate to 44.1kHz.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ignore MIDI and PCM events in the interrupt handler until the device
gets initialized properly. Otherwise you may get kernel panic by the
access to uninitialized devices via hotplugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mute speaker outputs on headphone insertion for machines that use
3stack-hp model.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent update enabled the model=sony-assamd for all ALC262 with
PCI SSID 104d:90xx. But this includes the VAIO VGN-AR* that has the
primary codec of STAC92xx and the secondary ALC262 as a slave
digital-only codec. For this device, the model=auto must be chosen
to work properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the mic input of HP dv6736 with Conexant 5051 codec chip.
This laptop seems have no mic-switching per jack connection.
A new model hp-dv6736 is introduced to match with the h/w implementation.
Reference: Novell bnc#480753
https://bugzilla.novell.com/show_bug.cgi?id=480753
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's false positive, but annoying.
sound/pci/hda/hda_codec.c: In function ‘get_empty_pcm_device’:
sound/pci/hda/hda_codec.c:2772: warning: ‘dev’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Add probe_mask default for Toshiba laptop with ALC268
ALSA: hda - Add quirk for new HP xw series
ALSA: hda - Fix digital mic on dell-m4-1 and dell-m4-3
Allow more options to be set/reset via hwdep hint entry.
hp_detect, gpio_mask, gpio_dir, gpio_data, eapd_mask and eapd_switch
can be checked.
For example, to disable hp_detect on the fly,
# echo "hp_detect=0" > /sys/class/sound/hwC0D0/hints
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't create "Analog Loopback" controls as default since these controls
are usually more harmful than useful for normal users.
Only created when "loopback = yes" hint is given.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_hda_get_hint() and snd_hda_get_bool_hint() helper functions
to retrieve a hint value.
Internally, the hint is stored in a pair of two strings, key and val.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't return a fatal error to the driver but continue to probe when
any error occurs at creating PCM streams for each codec.
It's often non-fatal and keeping it would help debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert the codec probe instead of returning the error to the driver
when any error occurs at creating the control elements.
The control element conflict can be non-fatal in many cases,
especially if it comes from the digital-only codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert the Toshiba probe_mask quirk for 2.6.29 kernel
(commit 38f1df27e3).
In the current tree, the digital-only codec is handled properly so
no codec conflict should occur.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an ALC268 codec is set up as the digital-only (as found in Toshiba
laptops), it shouldn't contain any beep control that conflict with the
primary codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
