Age | Commit message (Collapse) | Author |
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Resolve the line-in record problem.
Signed-off-by: Lionel Xu <r63889@freescale.com>
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When bluetooth handsfree is enabled, sgtl5000 and bt
sound cards are both opened. BT sound card playback
can't use IRAM as sgtl5000 has used IRAM. The bt audio
platform data - ext_ram is not set when kernel porting
to 2.6.31.
Signed-off-by: Wallace Wang <r59996@freescale.com>
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ASRC minmal output sample rate is 32000. When ASRC
is used as ASRC-ESAI per_2_per mode, the unsupported
sample rates should not be included in ASRC mixer.
Signed-off-by: Wallace Wang <r59996@freescale.com>
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No all regiser use _SCT. stmp3xxx_setl may write to wrong address.
please use __raw_write (value, REG_XXX_SET) to write date to SCT
register. Only REG_XXX_SCT is defined when it exist at hardware.
Signed-off-by: Frank.Li <Frank.Li@freescale.com>
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Add ac97 mode support for SSI module
Signed-off-by: Wallace Wang <r59996@freescale.com>
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Fix shift direction.
Don't return left channel value for both channels.
Signed-off-by: Alan Tull <r80115@freescale.com>
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To keep system runs well after plugin headphone, replace disable_irq with
disable_irq_nosync.
Signed-off-by: Guoqing Jiang <b22156@freescale.com>
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Trying to free already-free IRQ 240 causes the warning dumped out
when boot up the kernel.
Return the error code in the probe function, which indicats the
failure of the platform driver probing. Then the platform driver remove
function will not be called, avoiding of freeing an already-free IRQ again.
Signed-off-by: William Lai <b04597@freescale.com>
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The ak4647_codec is not initialized as there is no ak4647 codec existing
on MX35 board. In the 31 kernel, the card and codec drivers are
registered separately, and there is no check whether the codec is probed
successfully or not. On MX35 board, the card is sucessfully registered,
but there is no ak4647 codec. In this case, the system will crash if the
system try to invoke the codec's resume function.
To resolve the problem, add a check in registering the card. If the
codec does not exist, give up registering the card either.
Signed-off-by: William Lai <b04597@freescale.com>
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wm8580 is using regulators that can't change state. The
regulator driver doesn't handle this case well so it crashes
during regulator_bulk_enable. Eventually this will be fixed in the
regulator driver. For now, remove regulators from list in
wm8580 driver.
Fix spi_rw err. hw_write needs to return transferred data length.
Signed-off-by: Alan Tull <r80115@freescale.com>
Signed-off-by: Wallace Wang <r59996@freescale.com>
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In WM8350_FLL_CONTROL_1, keep FLL_RATE at default.
Don't mute outputs by default.
Signed-off-by: Alan Tull <r80115@freescale.com>
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Move regulator control into codec drivers.
Move codec platform data to machine layer.
Remove unused fields from mxc_audio_platform_data.
soc multiple audio cards support:
dai_list is a list for all registered codec/cpu dais.
One codec/cpu dais can't be registered twice, so move the
cpu dai registering into imx-ssi.c/imx-esai.c
Fix ak4647_read_reg error.
Put ssi/esai mode in the cpu_dai private_data.
sgtl5000: Fix the power for MX35
1. MX35 vddio and vddd are directly connected to
external 3V3. It has no regulator control.
2. sgtl5000 needs some time to be ready after vdda
is switched on so msleep(1) is added before regiter
read/write.
3. On MX35 platform, vdda is from VCAM.
Signed-off-by: Alan Tull <r80115@freescale.com>
Signed-off-by: Wallace Wang <r59996@freescale.com>
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Add basic support for integration with the regulator API to WM8580.
Since the core cannot yet disable biases when the CODEC is idle we
simply request and enable the regulators for the entire time the
driver is active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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It's the 8th enum of a zero indexed array. This is why I don't let
new drivers use these arrays of enums...
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
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For best performance the DAC sloping stopband filter should be enabled
below 24kHz and not enabled above that so remove the user visible
control for this and do it autonomously in the driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
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Fix the audio pop issue that is caused whenever the system enters and
exits LP-APM mode.
Added some more working points for DVFS-CORE.
Signed-off-by: Ranjani Vaidyanathan-RA5478 <Ranjani.Vaidyanathan@freescale.com>
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This is i.MX BSP 5.0.0 release ported to 2.6.31
Signed-off-by: Rob Herring <r.herring@freescale.com>
Signed-off-by: Alan Tull <r80115@freescale.com>
Signed-off-by: Xinyu Chen <xinyu.chen@freescale.com>
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Add spi interface support to wm8580 codec.
Signed-off-by: Wallace Wang <r59996@freescale.com>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/oxygen' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
sound: oxygen: handle cards with missing EEPROM
sound: oxygen: fix MCLK rate for 192 kHz playback
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The card model detection code introduced in 2.6.30 that tries to work
around partially broken EEPROM contents by reading the EEPROM directly
does not handle cards where the EEPROM has been omitted. In this case,
we have to use the default ID to allow the driver to load.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Ozan Çağlayan <ozan@pardus.org.tr>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Do not forget to program the MCLK ratio for the I2S output.
Otherwise, the master clock frequency can be too high for
the DACs at sample frequencies above 96 kHz.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Fix MacBookPro 3,1/4,1 quirk with ALC889A
ALSA: hda - Add missing mux check for VT1708
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This patch fixes the wrong headphone output routing for MacBookPro 3,1/4,1
quirk with ALC889A codec, which caused the silent headphone output.
Also, this gives the individual Headphone and Speaker volume controls.
Reference: kernel bug#14078
http://bugzilla.kernel.org/show_bug.cgi?id=14078
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
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In patch_vt1708(), the check of MUX nids is missing and this results in
the -EINVAL error in accessing Input Source mixer element. Simpliy
adding the call of get_mux_nids() fixes the problem.
Reference: Novell bnc#534904
https://bugzilla.novell.com/show_bug.cgi?id=534904
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/misc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
sound: pcm_lib: fix unsorted list constraint handling
sound: vx222: fix input level control range check
ALSA: ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready()
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snd_interval_list() expected a sorted list but did not document this, so
there are drivers that give it an unsorted list. To fix this, change
the algorithm to work with any list.
This fixes the "Slave PCM not usable" error with USB devices that have
multiple alternate settings with sample rates in decreasing order, such
as the Philips Askey VC010 WebCam.
http://bugzilla.kernel.org/show_bug.cgi?id=14028
Reported-and-tested-by: Andrzej <adkadk@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix a logic error in the range check of the input level control that
would prevent setting any volume less than the maximum.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Modify loops in such way that the register value is checked also after
the timeout condition, just in case the heavy interrupt load etc. caused
the thread to sleep for the time period exceeding the timeout value.
While at it remove an extra ALI_STIMER read from snd_ali_stimer_ready().
Reported-by: Jack Byer <ojbyer@usa.net>
Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Fix probe of Toshiba laptops with ALC268 codec
ALSA: hda: add model for Intel DG45ID/DG45FC boards
ALSA: hda: enable speaker output for Compaq 6530s/6531s
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There are many variants of Toshiba laptops with ALC268 codec, and
it seems that a few of them don't work with model=toshiba preset
since they have the secondary ALC268 codec just for HDMI output.
This is a regression due to the previous clean-up work to merge all
Toshiba quirk entries into a single check.
This patch adds the identification of such laptops to apply the
standard BIOS-probing method. Unfortunately, Toshiba laptops have
all the same PCI SSID, so we need to check the codec SSID to identify
each device.
Tested-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The BIOS pin configs are in fact correct and shall not be overwritten.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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HP Compaq 6530s and 6531s internal speaker is silence or becomes silence
within 1 minute after fresh boot. It is found that pin 0x1c must be set to
PIN_OUT mode to make the speaker work. This is weird - line-in pin 0x1c and
speaker pin 0x16 seem to be unrelated.
The codec differences before/after patch are:
@@ Node 0x17 [Pin Complex] wcaps 0x40020b:
Pin Default 0x41a6e130: [N/A] Mic at Ext Rear
Conn = Digital, Color = White
DefAssociation = 0x3, Sequence = 0x0
Misc = NO_PRESENCE
- Pin-ctls: 0x24: IN
+ Pin-ctls: 0x40: OUT
@@ Node 0x1c [Pin Complex] wcaps 0x40018d:
Pin Default 0x41813021: [N/A] Line In at Ext Rear
Conn = 1/8, Color = Blue
DefAssociation = 0x2, Sequence = 0x1
- Pin-ctls: 0x24: IN VREF_80
+ Pin-ctls: 0x40: OUT VREF_HIZ
Unsolicited: tag=00, enabled=0
Connection: 1
0x24
Tests show that it won't impact (external) Mic recording.
Reported-by: "Lin, Ming M" <ming.m.lin@intel.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/hda:
ALSA: hda - Don't override ADC definitions for ALC codecs
ALSA: hda - Add missing vmaster initialization for ALC269
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* fix/asoc:
ASoC: Add missing DRV_NAME definitions for fsl/* drivers
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ALC269 and ALC861-VD parsers override the ADC definitions
unconditionally without checking the spec definition. This causes
the problem when any inconsistent ADC is set up in the device quirk
(like ALC272 with digital-mic).
This patch avoids the overriding by adding the proper checks.
Reference: Novell bnc#529467
https://bugzilla.novell.com/show_bug.cgi?id=529467
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Without the initialization of vmaster NID, the dB information got
confused for ALC269 codec.
Reference: Novell bnc#527361
https://bugzilla.novell.com/show_bug.cgi?id=527361
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
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Module builds are broken due to missing DRV_NAME for
efika-audio-fabric and pcm030-audio-fabric.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Read buffer overflow
ALSA: hda: Correct EAPD for Dell Inspiron 1525
ALSA: hda: warn on spurious response
ALSA: hda: remember last command for each codec
ALSA: hda: read CORBWP inside reg_lock
ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io
ALSA: hda: take cmd_mutex in probe_codec()
ALSA: hda: track CIRB/CORB command/response states for each codec
ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527
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Check whether index is within bounds before testing the element.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The commit 24918b61b55c21e09a3e07cd82e1b3a8154782dc statically changes
the model from dell-bios to dell-3stack to solve the sound decreasing
regression (http://lkml.org/lkml/2008/9/12/203), however it leads to another
problem that the 2nd headphone jack doesn't work
(https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3987). So I think
the commit 249**2dc is just a workaround. I would like to give a true solution
here.
The datasheet for STAC9228 says, GPIO2 is the same pin as VOL DOWN, and
the EAPD pin is GPIO0. This is why the sound decreases if we set EAPD as
GPIO2. This patch changes EAPD to GPIO0 to solve the problem.
Signed-off-by: Chengu Wang <wangchengu@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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To help disclose hardware bugs.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This converts the last CORBWP access outside of reg_lock.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Just for safety. azx_init_cmd_io() and azx_free_cmd_io() may be
called when switching to single command mode.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Now that each codec will have its own module, it is possible
for the user to load one codec while another one is running.
So cmd_mutex would be a safe addition to probe_codec().
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Recently we hit a bug in our dev board, whose HDMI codec#3 may emit
redundant/spurious responses, which were then taken as responses to
command for another onboard Realtek codec#2, and mess up both codecs.
Extend the azx_rb.cmds and azx_rb.res to array and track each codec's
commands/responses separately. This helps keep good codec safe from
broken ones.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use model=lenovo instead of model=dallas for Toshiba Satellite A135-S4527
with ALC861-VD codec.
Reference: Novell bnc#526325
https://bugzilla.novell.com/show_bug.cgi?id=526325
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/oss:
sound: mpu401.c: Buffer overflow
sound: aedsp16: Buffer overflow
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