On the docking station for the Lenovo T410 and T410s, the line-out
doesn't work. The trouble seems to be that it generates a plug event,
but then doesn't report that the jack is connected. So automute mutes
the jack when you plug something into it. The following patch (next
message) fixes it.
Signed-off-by: John Baboval <john.baboval at virtualcomputer.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/595482
The original reporter states that audible playback from the internal
speaker is inaudible despite the hardware being properly detected. To
work around this symptom, he uses the model=lg quirk to properly enable
both playback, capture, and jack sense. Another user corroborates this
workaround on separate hardware. Add this PCI SSID to the quirk table
to enable it for further LG P1 Expresses.
Reported-and-tested-by: Philip Peitsch <philip.peitsch@gmail.com>
Tested-by: nikhov
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/683695
The original reporter states that headphone jacks do not appear to
work. Upon inspecting his codec dump, and upon further testing, it is
confirmed that the "alienware" model quirk is correct.
Reported-and-tested-by: Cody Thierauf
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In OSS emulation, SNDCTL_DSP_RESET ioctl needs the reset of the internal
buffer state in addition to drop of the running streams. Otherwise the
succeeding access becomes inconsistent.
Tested-by: Amit Nagal <helloin.amit@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/682199
A 2.6.35 (Ubuntu Maverick) user, burningphantom1, reported a regression
in audio: playback was inaudible through both speakers and headphones.
In commit 272a527c04 of sound-2.6.git, a new model was added with this
machine's PCI SSID. Fortunately, it is now sufficient to use the auto
model for BIOS auto-parsing instead of the existing quirk.
Playback, capture, and jack sense were verified working for both
2.6.35 and the alsa-driver snapshot from 2010-11-27 when model=auto is
used.
Reported-and-tested-by: burningphantom1
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When SKU assid gives no valid bits for 0x38, the driver didn't take
any action, so far. This resulted in the missing initialization for
external amps, etc, thus the silent output in the end.
Especially users hit this problem on ALC888 newly since 2.6.35,
where the driver doesn't force to use ALC_INIT_DEFAULT any more.
This patch sets the default initialization scheme to use
ALC_INIT_DEFAULT when no valid bits are set for SKU assid.
Reference:
https://bugzilla.redhat.com/show_bug.cgi?id=657388
Reported-and-tested-by: Kyle McMartin <kyle@redhat.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mixer nids passed to alc_auto_create_input_ctls are wrong: 0x15 is
a pin, and 0x09 is the ADC on both ALC660-VD/ALC861-VD. Thus with
current code, input playback volume/switches and input source mixer
controls are not created, and recording doesn't work. Select correct
mixers, 0x0b (input playback mixer) and 0x22 (capture source mixer).
Reference: https://qa.mandriva.com/show_bug.cgi?id=61159
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch enables ALC887-VD to use the DAC at nid 0x26,
which makes it possible to use this DAC for e g Headphone
volume.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (41 commits)
ALSA: hda - Identify more variants for ALC269
ALSA: hda - Fix wrong ALC269 variant check
ALSA: hda - Enable jack sense for Thinkpad Edge 11
ALSA: Revert "ALSA: hda - Fix switching between dmic and mic using the same mux on IDT/STAC"
ALSA: hda - Fixed ALC887-VD initial error
ALSA: atmel - Fix the return value in error path
ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52J
ALSA: snd-atmel-abdac: test wrong variable
ALSA: azt3328: period bug fix (for PA), add missing ACK on stop timer
ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixup
ALSA: sound/pci/asihpi/hpioctl.c: Remove unnecessary casts of pci_get_drvdata
ALSA: sound/core/pcm_lib.c: Remove unnecessary semicolons
ALSA: sound/ppc: Use printf extension %pR for struct resource
ALSA: ac97: Apply quirk for Dell Latitude D610 binding Master and Headphone controls
ASoC: uda134x - set reg_cache_default to uda134x_reg
ASoC: Add support for MAX98089 CODEC
ASoC: davinci: fixes for multi-component
ASoC: Fix register cache setup WM8994 for multi-component
ASoC: Fix dapm_seq_compare() for multi-component
ASoC: RX1950: Fix hw_params function
...
The refactoring commit d433a67831
ALSA: hda - Optimize the check of ALC269 codec variants
introduced a wrong check for ALC269-vb type. This patch corrects it.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk entry for Thinkpad Edge 11 as well as other TP Edge models.
Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC887-VD is like ALC888-VD. It can not be initialized as ALC882.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the commit c0763e687d
ALSA: snd-atmel-abdac: test wrong variable
the return value via PTR_ERR() had to be fixed as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/677652
The original reporter states that, in 2.6.35, headphones do not appear
to work, nor does inserting them mute the A52J's onboard speakers. Upon
inspecting the codec dump, it appears that the newly committed hp-laptop
quirk will suffice to enable this basic functionality. Testing was done
with an alsa-driver build from 2010-11-21.
Reported-and-tested-by: Joan Creus
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After clk_get() pclk is checked second time instead of sample_clk check.
Signed-off-by: Vasiliy Kulikov <segoon@openwall.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
. Fix PulseAudio "ALSA driver bug" issue
(if we have two alternated areas within a 64k DMA buffer, then max
period size should obviously be 32k only).
Back references:
http://pulseaudio.org/wiki/AlsaIssueshttp://fedoraproject.org/wiki/Features/GlitchFreeAudio
. In stop timer function, need to supply ACK in the timer control byte.
. Minor log output correction
When I did my first PA testing recently, the period size bug resulted
in quite precisely observeable half-period-based playback distortion.
PA-based operation is quite a bit more underrun-prone (despite its
zero-copy optimizations etc.) than raw ALSA with this rather spartan
sound hardware implementation on my puny Athlon.
Note that even with this patch, azt3328 still doesn't work for both
cases yet, PA tsched=0 and tsched
(on tsched=0 it will playback tiny fragments of periods, leading to tiny
stuttering sounds with some pauses in between, whereas with
timer-scheduled operation playback works fine - minus some quite increased
underrun trouble on PA vs. ALSA, that is).
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/677830
The original reporter states that the subwoofer does not mute when
inserting headphones. We need an entry for his machine's SSID in the
subwoofer pin fixup list, so add it there (verified using hda_analyzer).
Reported-and-tested-by: i-NoD
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/669279
The original reporter states: "The Master mixer does not change the
volume from the headphone output (which is affected by the headphone
mixer). Instead it only seems to control the on-board speaker volume.
This confuses PulseAudio greatly as the Master channel is merged into
the volume mix."
Fix this symptom by applying the hp_only quirk for the reporter's SSID.
The fix is applicable to all stable kernels.
Reported-and-tested-by: Ben Gamari <bgamari@gmail.com>
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After checking the code in 2.6.36,
I found this is missing during multi-component conversion.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds initial support for the MAX98089 CODEC.
Signed-off-by: Jesse Marroquin <jesse.marroquin@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Multi-component commit f0fba2ad broke a few things which this patch should
fix. Tested on the DM355 EVM. I've been as careful as I can, but it would be
good if those with access to other Davinci boards could test.
--
The multi-component commit put the initialisation of
snd_soc_dai.[capture|playback]_dma_data into snd_soc_dai_ops.hw_params of the
McBSP, McASP & VCIF drivers (davinci-i2s.c, davinci-mcasp.c & davinci-vcif.c).
The initialisation had to be moved from the probe function in these drivers
because davinci_*_dai changed from snd_soc_dai to snd_soc_dai_driver.
Unfortunately, the DMA params pointer is needed by davinci_pcm_open (in
davinci-pcm.c) before hw_params is called. I have moved the initialisation to
a new snd_soc_dai_ops.startup function in each of these drivers. This fix
indicates that all platforms that use davinci-pcm must have been broken and
need to test with this fix.
--
The multi-component commit also changed the McBSP driver name from
"davinci-asp" to "davinci-i2s" in davinci-i2s.c without updating the board
level references to the driver name. This change is understandable, as there
is a similarly named "davinci-mcasp" driver in davinci-mcasp.c.
There is probably no 'correct' name for this driver. The DM6446 datasheet
calls it the "ASP" and describes it as a "specialised McBSP". The DM355
datasheet calls it the "ASP" and describes it as a "specialised ASP". The
DM365 datasheet calls it the "McBSP". Rather than fix this problem by
reverting to "davinci-asp", I've elected to avoid future confusion with the
"davinci-mcasp" driver by changing it to "davinci-mcbsp", which is also
consistent with the names of the functions in the driver. There are other
fixes required, so it was never going to be as simple as a revert anyway.
--
The DM365 only has one McBSP port (of the McBSP platforms, only the DM355 has
2 ports), so I've changed the the id of the platform_device from 0 to -1.
--
In davinci-evm.c, the DM6446 EVM can no longer share a snd_soc_dai_link
structure with the DM355 EVM as they use different cpu DAI names (the DM355
has 2 ports and the EVM uses the second port, but the DM6446 only has 1 port).
This also means that the 2 boards need different snd_soc_card structures.
--
The codec_name entries in davinci-evm.c didn't match the i2c ids in the board
files. I have only checked and fixed the details of the names used for the
McBSP based platforms. Someone with a McASP based platform (eg DA8xx) should
check the others.
Signed-off-by: Chris Paulson-Ellis <chris@edesix.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
During the multi-component conversion the WM8994 register cache init
got lost.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The big kernel lock has been removed from all these files at some point,
leaving only the #include.
Remove this too as a cleanup.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Ensure that we keep all widget powerups in DAPM sequence by making
the CODEC the last thing we compare on rather than the first thing.
Also fix the fact that we're currently comparing the widget pointers
rather than the CODEC pointers when we do the substraction so we
won't get stable results.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unfortunatelly, I misunderstood datasheet, and on s3c244x-iis
when MPLLin source for master clock is selected, prescaler has
no effect. Remove dividor calculation for 44100 rate; remove 88200
rate at all, rx1950 can't do it.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add Kconfig dependency on AT91_PROGRAMMABLE_CLOCKS for the Atmel SoC
audio SAM9G20-EK and PlayPaq boards. Fixes link errors on missing
clk_set_parent and clk_set_rate when building without
AT91_PROGRAMMABLE_CLOCKS.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
Acked-by: Geoffrey Wossum <gwossum@acm.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Atmel SSC can divide by even numbers, not only powers of two.
Signed-off-by: Peter Rosin <peda@axentia.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In each function, the value apcm is stored in the private_data field of
runtime. At the same time the function ct_atc_pcm_free_substream is stored
in the private_free field of the same structure. ct_atc_pcm_free_substream
dereferences and ultimately frees the value in the private_data field. But
each function can exit in an error case with apcm having been freed, in
which case a subsequent call to the private_free function would perform a
dereference after free. On the other hand, if the private_free field is
not initialized, it is NULL, and not invoked (see snd_pcm_detach_substream
in sound/core/pcm.c). To avoid the introduction of a dangling pointer, the
initializations of the private_data and private_free fields are moved to
the end of the function, past any possible free of apcm. This is safe
because the previous calls to snd_pcm_hw_constraint_integer and
snd_pcm_hw_constraint_minmax, which take runtime as an argument, do not
refer to either of these fields.
In each function, there is one error case where apcm needs to be freed, and
a call to kfree is added.
The sematic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression e,e1,e2,e3;
identifier f,free1,free2;
expression a;
@@
*e->f = a
... when != e->f = e1
when any
if (...) {
... when != free1(...,e,...)
when != e->f = e2
* kfree(a)
... when != free2(...,e,...)
when != e->f = e3
}
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the platform already provides a definition for these accessors
do not redefine them. The warning was caught on MIPS.
Signed-off-by: Florian Fainelli <florian@openwrt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/673075
According to the datasheet of 92HD87B, there is a digital mic
at nid 0x11, so enable it in order to be able to use the mic.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The [vk][cmz]alloc(_node) family of functions return void pointers which
it's completely unnecessary/pointless to cast to other pointer types since
that happens implicitly.
This patch removes such casts from sound/oss/
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ensure that whatever ran before us leaves the WM835x with a sane default
audio interface configuration as we do not override the companding,
loopback or tristate settings and do not reset the chip at startup (as it
is a PMIC).
Reported-by: Keiji Mitsuhisa <Keiji.Mitsuhisa@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8350 driver was using some custom constants to interpret the direction
of the MCLK signal which had the opposite values to those used as standard
by the ASoC core, causing confusion in machine drivers such as the 1133-EV1
board.
Reported-by: Tommy Zhu <Tommy.Zhu@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
When make mini2440_defconfig compilation end with undefined
references to DMA functions. There was missing selection
for S3C2410_DMA when compile ASoC audio for S3C24xx CPU.
Tested on mini2440 board.
Signed-off-by: Marek Belisko <marek.belisko@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The member reg_cache is not used at all and therefore it should be
removed. This member was usually needed for older versions of ASoC
that did not handle caching automatically and had to be done in the
driver itself.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When doing anything with the system, especially DAPM, we need to hold the
CODEC mutex.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
We have different codecs on the pcm038 (ac97 wm9712 and mc13783).
To make alsactl restore work correctly these should have different
names.
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>
We have two different transfer methods on i.MX: FIQ and DMA. Since
the merge of the ASoC multicomponent support the DMA device is lost.
Add it again. Also, imx_ssi_dai_probe has to be called for !AC97
aswell.
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>
