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>
Use db scale for all volume controls according to Crystal's datasheets.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the quirk for Acer Aspire 5930G from model=acer-aspire-4930g to
model=acer-aspre-6530g. The tuba bass gets muted along with the other
built-in speakers upon headphones insertion, the internal mic works
perfectly etc.
Reported-by: Claudio Viano <claudio.viano@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use edma_pause and edma_resume to make missing dma_events
less likely. This may not be needed, but it looks better.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix underruns by using dma to copy 1st to sram
in a ping/pong buffer style and then copying from
the sram to the ASP. This also has the advantage
of tolerating very long interrupt latency on dma
completion.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rename variable master_lch to asp_channel
Rename variable slave_lch to asp_link[0]
Rename local variables:
lch to link
count to asp_count
src to asp_src
dst to asp_dst
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the left and right 16 bit samples to be shifted out as 1
32 bit sample.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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:
ASoC: tlv320aic23 fix rate selection
ASoC: OMAP3 Pandora: update for TWL4030 codec changes
ASoC: Modifying the license string GPLv2 for OMAP3 EVM
ALSA: hda - Fix quirk for VAIO type G
ALSA: usb - Quirk to disable master volume control in PCM2702
The mute-LED isn't synchronized with the actual mute state on some
HP laptops with IDT 92HD83xxx codecs. A similar hack using
check_power_status callback is added for this codec, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the ordering of sr_valid_mask array.
The lower bit of the index represents USB
not bosr.
Reported-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A while ago TWL4030 had it's playback stream name changed, but
pandora needs it for it's playback path. Update to correct stream
name so that playback works again.
Also mark VIBRA output as not connected.
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>
Correcting the license string from GPLv2 -> GPL v2.
Found the problem while building OMAP3 ASoC driver as
module.
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>
Remove requirement that dma_params is 1st in the structures
davinci_audio_dev and davinci_mcbsp_dev.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Making room for namespace for the PCM Controller driver
the platform driver(s3c24xx-pcm) has been renamed to SoC
agnostic name 's3c-dma'.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The s3c24xx_pcm prefix for the soc_platform is inappropriate when
some Samsung SoCs have PCM controllers which will eventually have
drivers and hence namespace ambiguities.
To resolve naming ambiguities in future the following have been
renamed in order
1) s3c24xx_pcm_dma_params -> s3c_dma_params
2) s3c24xx_pcm_preallocate_dma_buffer -> s3c_preallocate_dma_buffer
3) s3c24xx_pcm_dmamask -> s3c_dma_mask
4) s3c24xx_pcm_XXX -> s3c_dma_XXX
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The dual-headphone mode with STAC/IDT codecs is useful only for machines
that have two (or more) built-in headphones.
But, some HP laptops give multiple headphone pin configs, one for the
built-in and another for the separate (likely a docking station) one.
This results in a missing speaker volume control.
This patch adds more check for the dual-headphone mode to avoid this
problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This function is only called from snd_ctl_ioctl() and the file parameter
can never be null so there is no need to check it here.
We dereference file at the start of the function:
struct snd_card *card = file->card;
and it confuses static checkers to dereference a pointer before
checking it.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We tracked down the first-0.5s-hdmi-audio-samples-lost problem to the
AC_VERB_SET_CHANNEL_STREAMID command. It is suspected that many HDMI
sinks need some time to adapt to the new state.
The workaround is to avoid changing stream id/format whenever possible.
Proposed by David.
Signed-off-by: David Härdeman <david@hardeman.nu>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remember the active infoframe, so as to avoid stop/restart infoframe
transmission when switching between audio clips of the same format.
Proposed by Shang and David.
CC: Shane W <shane-alsa@csy.ca>
CC: David Härdeman <david@hardeman.nu>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
And make it right when called for more than one times.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This avoids lost of presence info on module reloading.
The presence info used to be only updated at the (rare) hotplug events.
Proposed by David, thanks!
CC: David Härdeman <david@hardeman.nu>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the fm_port and mpu_port variables directly in a probe function.
This completely eliminates a need to copy the fm_port value to
the snd_miro structure.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update control names to be more closer to their meaning.
Change the "Mono" name to the "Beep" as this line is usually
used to forward the PC beeper signal to sound card's output.
Update names for both cs423x and wss.
Clean up cs4235 controls according to the cs4235 doc. Rename
some of the cs4235 controls to be consistent with the cs4236's
ones.
Also, delete one misnamed cs4231 register define.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: ice1724 - make some bitfields unsigned
ALSA: hda - Dell Studio 1557 hd-audio quirk
ALSA: ice1724 - Fix section mismatch in prodigy_hd2_resume()
ALSA: hda - Add another Nvidia HDMI codec id (10de:0005)
ALSA: hda: Use model=mb5 for MacBookPro 5,2
* 'hostprogs-wmissing-prototypes' of git://git.kernel.org/pub/scm/linux/kernel/git/josh/linux-misc:
Makefile: Add -Wmising-prototypes to HOSTCFLAGS
oss: Mark loadhex static in hex2hex.c
dtc: Mark various internal functions static
dtc: Set "noinput" in the lexer to avoid an unused function
drm: radeon: Mark several functions static in mkregtable
arch/sparc/boot/*.c: Mark various internal functions static
arch/powerpc/boot/addRamDisk.c: Mark several internal functions static
arch/alpha/boot/tools/objstrip.c: Mark "usage" static
Documentation/vm/page-types.c: Declare checked_open static
genksyms: Mark is_reserved_word static
kconfig: Mark various internal functions static
kconfig: Make zconf.y work with current bison
The ALC262 has a quirk entry matching with all Sony Vaio laptops
to use model=sony-assamd as default. But, model=auto works much better
for new models in the recent driver versions, thus it's safer to disable
that default quirk entry.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vaio type G laptop doesn't work with the current quirk setup.
After some tests, it turned out that it should be model=auto as default.
Reported-by: Mattia Dongili <malattia@linux.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Disable the master volume control in the PCM2702 chipset.
The datasheet documents two independent channel volume controls, one
master mute control and one master volume control. All controls are
fully functional except for the master volume control, which returns
USB stalls on all GET requests.
Signed-off-by: Javier Kohen <jkohen@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the {orig,midi}_dev equals num_midis, that's one too
large already.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The beep_mode option value was wrongly defined: it must be 0 = off and
1 = on.
Also, evaluate the beep_mode value at snd_hda_attach_beep_device()
properly so that no device is created when beep_mode=0 is given.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for dynamically created controls to proc codec file
(Control: lines).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is an initial patch to show universal control<->NID assigment in
proc codec file. The change helps to debug codec related problems.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The beep_mode parameter for snd-hda-intel module allows to choose among
different digital beep device registation to the input layer.
0 = do not register to the input layer
1 = register to the input layer all time
2 = use "Beep Switch" control exported to user space mixer applications
Also, introduce CONFIG_SND_HDA_INPUT_BEEP_MODE for default value.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The unregister work should be also canceled in snd_hda_detach_beep_device()
function.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The massive register/unregister calls for input device layer might be
overkill. Delay unregister call by one HZ as workaround.
Also, as benefit, beep->enabled variable is changed immediately now
(not from workqueue).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Original implementation was keeping registered input device for SND_BEEP
and SND_TONE events all time. This patch changes this behaviour:
If digital PC Beep is turned off using universal control switch,
the input device is unregistered.
Explanation: The kd_mksound() send SND_BEEP and SND_TONE only to last
registered device acceping those events. It means that the HDA Intel
audio driver blocks also the internal PC Speaker device (pcspkr.c
driver) even if the HDA Beep is muted. The user can easy disable
all beeps using 'setterm -blength 0' or 'xset b off' command.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a clean up and doesn't change the behavior.
Bit fields should always be unsigned. Otherwise pm_suspend_enabled will
be -1 when you want it to be 1. The other bad thing is that the sparse
checker will complain 36 times if they aren't unsigned.
The other bitfields in that struct are unsigned already.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add proper suspend/resume code for Juli@ cards. Based on ice1724
suspend/resume work of Igor Chernyshev.
Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413
Tested on linux-2.6.31.6
Signed-off-by: Aleksey Kunitskiy <alexey.kv@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_via.c: In function 'via_hp_bind_automute':
sound/pci/hda/patch_via.c:2074: internal compiler error: in do_SUBST, at combine.c:462
Please submit a full bug report,
with preprocessed source if appropriate.
See <URL:http://gcc.gnu.org/bugs.html> for instructions.
[added a comment by tiwai]
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DAPM widgets are now insntantiated by the core when creating the card
so there is no need for the individual CODEC drivers to do so.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The jack_status_check callback function is the interface to check the
status of the jack. Some target provides the method to distinguish what
is the jack inserted - headphone jack, microphone jack, tvout jack, etc,
so we can implement it using the jack_status_check function.
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>
While probing, AC97 codec drivers and soc-core generically execute the
following sequence:
snd_soc_new_ac97_codec -> snd_soc_new_pcms -> reset ac-link/read AC97 ID
to detect ->... -> set platform_data to ac97 by soc-core
commit 474828a40f adds platform_data to
snd_ac97 instance. But ac97 platform data hasn't given to snd_ac97
before actual ac97 operations. Then while ac97_read access platform_data
of snd_ac97 for detecting, NULL pointer oops will fire. That means old
platform_data patch doesn't work in real-life cases.
This patch moves the operation of setting ac97 platform_data earlier
than ac97 reading/writing operations. Then it makes platform_data of
AC97 become practically useful.
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>
Corrected the order of 'source' and 'pll_id' arguments.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many devices need to calculate the bit clock rate desired to
work out the clock configuration required for the device.
Provide utility functions to do this using both hw_params
structures and raw numbers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
CONFIG_SND_HDA_POWER_SAVE is independent from CONFIG_SND_HDA_HWDEP.
Thus snd_hda_hwdep_add_power_sysfs() needs the check of both kconfigs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit e330323520
"ALSA: hda - proc - show which I/O NID is associated to PCM device"
introduces the access to substream pointer. But, PCMs may have no
substreams in one or both directions, and this results in NULL
dereference. Also, print the first substream number doesn't make
sense.
This patch removes the access to the substream pointer, and reformat
to fit to the standard coding style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/462098
Until we can look closer at the verbs, let's use ALC885_MB5 for
codec SSID 0x106b4600 to enable playback and capture for MacBookPro
5,2s.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the power on/off counter and expose via sysfs files.
The sysfs files, power_on_acct and power_off_acct, are created under
each codec hwdep sysfs directory (e.g. /sys/class/sound/hwC0D0).
The files show the msec length of the codec power-on and power-off,
respectively.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The test of index `i' is after the read - too late - and
unsafe: if snd_hda_get_connections() fails in the last
iteration a read beyond the array is possible.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Record the pid of the task that opened a RawMIDI substream.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
