AD1986A codec is a pretty old codec and has really many hidden
restrictions. One of such is that each DAC is dedicated to certain
pin although there are possible connections. Currently, the generic
parser tries to assign individual DACs as much as possible, and this
lead to two bad situations: connections where the sound actually
doesn't work, and connections conflicting other channels.
We may fix this by trying to find the best connections more harder,
but as of now, it's easier to give some hints for paired DAC/pin
connections and honor them if available, since such a hint is needed
only for specific codecs (right now only AD1986A, and there will be
unlikely any others in future).
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell machines with codec whose Subsystem Id is 0x10280624,
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/bugs/1259790
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case a single HDA card has both HDMI and S/PDIF outputs, the S/PDIF
outputs will have their IEC958 controls created starting from index 16
and the HDMI controls will be created starting from index 0.
However, HDMI simple_playback_build_controls() as used by old VIA and
NVIDIA codecs incorrectly requests the IEC958 controls to be created
with an S/PDIF type instead of HDMI.
In case the card has other codecs that have HDMI outputs, the controls
will be created with wrong index=16, causing them to e.g. be unreachable
by the ALSA "hdmi" alias.
Fix that by making simple_playback_build_controls() request controls
with HDMI indexes.
Not many cards have an affected configuration, but e.g. ASUS M3N78-VM
contains an integrated NVIDIA HDA "card" with:
- a VIA codec that has, among others, an S/PDIF pin incorrectly
labelled as an HDMI pin, and
- an NVIDIA MCP7x HDMI codec.
Reported-by: MysterX on #openelec
Tested-by: MysterX on #openelec
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org> # 3.8+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not all channels have been initialized, so far, especially when aamix
NID itself doesn't have amps but its leaves have. This patch fixes
these holes. Otherwise you might get unexpected loopback inputs,
e.g. from surround channels.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell Inspiron 3045 machine (codec Subsystem Id: 0x10280628),
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259437
CC: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell Optiplex 3030 machine (codec Subsystem Id: 0x10280623),
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259435
CC: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I've tested the old Dell Vostro 131 with the latest generic parser
and it works just fine, and as a bonus we get better jack detection
features in userspace. Therefore this quirk can be removed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following warning when optimizing for size with gcc-4.6.4:
sound/usb/mixer_quirks.c:1514:6: warning: 'err' may be used uninitialized in this function [-Wuninitialized]
Signed-off-by: Mikulas Patocka <mpatocka@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add quirk for Acer Aspire E-572:
- fix external mic
- limit mic boost for internal mic with maximal noise level of -24dB
Signed-off-by: Oleksij Rempel <linux@rempel-privat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A smattering of fixes here, some core ones for the rate combination
issues for things other than simple bitmasks, for readback of byte
controls and for updating the power of value muxes plus a bunch of
driver fixes of varying severity.
The warning fix in the i.MX FIQ driver is fixing a warning introduced
by a previous fix.
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Merge tag 'asoc-v3.13-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.13
A smattering of fixes here, some core ones for the rate combination
issues for things other than simple bitmasks, for readback of byte
controls and for updating the power of value muxes plus a bunch of
driver fixes of varying severity.
The warning fix in the i.MX FIQ driver is fixing a warning introduced
by a previous fix.
In the case of using jackpoll_ms instead of unsol events, the jack
was correctly detected, but ELD info was not refreshed on plug-in.
And without ELD info, no proper restriction of pcm, which can in turn
break sound output on some devices.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I forgot to remove the hp_automute_hook from alc283_fixup_chromebook.
It doesn't need this for other chrome os machine.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to WM8731 "PD, Rev 4.9 October 2012" datasheet, when it
works in DSP mode A, LRP = 1, while works in DSP mode B, LRP = 0.
So, fix LRP for DSP mode as the datesheet specification.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Create single model for HP.
The headset jack module was difference between other chrome book.
It need to manual control Mic jack detect.
Chrome OS loaded driver by models. Remove old assigned fixup table from
ALC269 fixup list entry.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By trial and error, I found this patch could work around an issue
where the headset mic would stop working if you switch between the
internal mic and the headset mic, and the internal mic was muted.
It still takes a second or two before the headset mic actually starts
working, but still better than nothing.
Information update from Kailang:
The verb was ADC digital mute(bit 6 default 1).
Switch internal mic and headset mic will run alc_headset_mode_default.
The coef index 0x11 will set to 0x0041.
Because headset mode was fixed type. It doesn't need to run
alc_determine_headset_type.
So, the value still keep 0x0041. ADC was muted.
BugLink: https://bugs.launchpad.net/bugs/1256840
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that AD1986A cannot manage the dynamic pin on/off for
auto-muting, but rather gets confused. Since each output has own amp,
let's use it instead.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Cc: <stable@vger.kernel.org> [v3.11+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ad_vmaster_eapd_hook() needs to handle the inverted EAPD case
properly, too. Otherwise the output gets broken on Lenovo N100 with
AD1986A codec.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS Z35HL laptop also needs the very same fix as the previous one
that was applied to ASUS W7J.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66231
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The static checker found a possible array overflow in atmel/abdac.c:
static checker warning: "sound/atmel/abdac.c:373 set_sample_rates()
error: buffer overflow 'dac->rates' 6 <= 6"
This patch papers over the buggy point, by ensuring that dac->rates[]
update not overflowing the actual array size.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the probe of snd-hda-intel driver is deferred due to f/w loading
or the nested module loading, complete_all() should be also delayed
until the initialization really finished. Otherwise, vga-switcheroo
client would start switching before the actual init is done.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that EAPD on NID 0x16 is the only control over all outputs on
HP machines with AD1984A while turning EAPD on NID 0x12 breaks the
output. Thus we need to avoid fiddling EAPD on NID. As a quick
workaround, just set own_eapd_ctrl flag for the wrong EAPD, then
implement finer EAPD controls.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66321
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
N810 audio driver has stopped working at some point. Probably when
OMAP2 was converted to common clock framework since now call to clk_enable
dumps the stack trace in drivers/clk/clk.c: __clk_enable() due
clk->prepare_count is zero.
Fix this by converting clk_enable/_disable calls to those that take care
of clock prepare/unprepare.
I'm not queueing this to linux-stable since OMAP2 common clock framework
conversion in commit ed1ebc4948 ("ARM: OMAP2: clock: Convert to common clk")
happened before N810 was really usable in mainline and user base for N810 is
anyway small. Potential linux-stable candidates are only those after
commit 3d3a6d18ab ("watchdog: introduce retu_wdt driver").
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
platform_set_drvdata(op, pdata) in pcm030_fabric_probe()
will be overwrited when calling snd_soc_register_card(card),
but cm030_fabric_remove() use drvdata as a type of struct
pcm030_audio_data, so we should move platform_set_drvdata()
below snd_soc_register_card() call.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@linaro.org>
Originally snd_hrtimer_callback() used iprtd->period_time for
some jiffies based estimation to determine the right moment
to call snd_pcm_period_elapsed(). As timer drifts may well be a
problem, this was changed in commit b4e82b5b78 to be based
on buffer transmission progress, using iprtd->offset and
runtime->buffer_size to calculate the amount of data since last
period had elapsed.
Unfortunately, iprtd->offset counts in bytes, while
runtime->buffer_size counts frames, so adding these to find some
delta is like comparing apples and oranges, and eventually results
in negative delta values every now and then. This is no big harm,
because it simply causes snd_pcm_period_elapsed() being called
more often than necessary, as negative delta is taken for a
large unsigned value by implicit conversion rule.
Nonetheless, the calculation is broken, so one would replace
the runtime->buffer_size by its equivalent in bytes.
But then, there are chances snd_pcm_period_elapsed() is called
late, because calculating the moment for the elapsed period
into delta is based against the iprtd->last_offset, which is not
necessarily the first byte of the period in question, but some
random byte which the FIQ handler left us with in r8/r9 by
accident. Again, negative impact is low, as there are plenty of
periods already prefilled with data, and snd_pcm_period_elapsed()
will probably be called latest when the following period is
reached. However, the calculation is conceptually broken, and we
are best off removing the clever stuff altogether.
snd_pcm_period_elapsed() is now simply called once everytime
snd_hrtimer_callback() is run, which may not be most accurate,
but at least this way we are quite sure we dont miss an end of
period. There is not much extra effort wasted by superfluous
calls to snd_pcm_period_elapsed(), as the timer frequency
closely matches the period size anyway.
Signed-off-by: Oskar Schirmer <oskar@scara.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The recent kernels got regressions on ASUS W7J with ALC660 codec where
no sound comes out. After a long debugging session, we found out that
setting the pin control on the unused NID 0x10 is mandatory for the
outputs. And, it was found out that another magic of NID 0x0f that is
required for other ASUS laptops isn't needed on this machine.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66081
Reported-and-tested-by: Andrey Lipaev <lipaev@mail.ru>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_bytes_put treats the data in the binary control as big endian
words, however snd_soc_bytes_get uses the endian of the host machine.
This causes the two functions to be inconsistant with how the mask is
applied on little endian machines.
This patch applies the big_endian format used in snd_soc_bytes_put to
snd_soc_bytes_get.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The array limits are supposed to be in units of u32 instead of in bytes.
The current code has a potential array overflow.
Fixes: c614475b0e ('ALSA: dice: add a proc file to show device information')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This machine also has mono output if run through DAC node 0x03.
Cc: stable@vger.kernel.org (v3.10+)
BugLink: https://bugs.launchpad.net/bugs/1256212
Tested-by: David Chen <david.chen@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the previous commit 1f0bbf03cb added the pin config for the bass
speaker, this patch adds the corresponding LFE-only channel map on
ASUS ET2700.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a fixup entry for the missing bass speaker pin 0x16 on ASUS ET2700
AiO desktop. The channel map will be added in the next patch, so that
this can be backported easily to stable kernels.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This both devices need limit for internal dmic.
[cosmetic change; renamed fixup name by tiwai]
Signed-off-by: Oleksij Rempel <linux@rempel-privat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current generic parser assumes blindly that the volume and mute
amps are found in the aamix node itself. But on some codecs,
typically Analog Devices ones, the aamix amps are separately
implemented in each leaf node of the aamix node, and the current
driver can't establish the correct amp controls. This is a regression
compared with the previous static quirks.
This patch extends the search for the amps to the leaf nodes for
allowing the aamix controls again on such codecs.
In this implementation, I didn't code to loop through the whole paths,
since usually one depth should suffice, and we can't search too
deeply, as it may result in the conflicting control assignments.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65641
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the setting of the register KIRKWOOD_PLAYCTL which did
always streaming on both I2S and SPDIF, ignoring the DAI ID.
The bug was introduced by the commit 75b9b65ee5
"ASoC: kirkwood: add S/PDIF support"
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch removes the 32 bits format which is not supported by S/PDIF
output.
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_pcm_limit_hw_rates() will initialize the minimum and maximum sample rate for
the PCM stream based on the rates specified in the rates field. Since we call
snd_pcm_limit_hw_rates() after soc_pcm_init_runtime_hw() it will essentially
overwrite the min and max rate set in soc_pcm_init_runtime_hw(). This may cause
the minimum or maximum rate to be set to a value outside the range of one of the
components if one of the components sets either SNDRV_PCM_RATE_CONTINUOUS or
SNDRV_PCM_RATE_KNOT and the other component specified a discrete rate via
SNDRV_PCM_RATE_[0-9]* that is outside of the first component's rate range. To
fix this first calculate the minimum and maximum rates using
snd_pcm_limit_hw_rates() and then on top of that apply the contraints specified
in the snd_soc_pcm_stream structs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Takashi iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In order to make sure that the sample rate is in the supported range of both
components the maximum rate of the card should be the minimum of the maximum
rate of each components. There is one special case to consider though, if
max_rate is set to 0 this means there is no maximum specified, so use
min_not_zero() macro which will give use the desired result.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Takashi iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
These are managed automatically in current revisions.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As the priv is not assigned to card->drvdata, it is NULL, so when
unload module, it will cause NULL pointer oops.
Assign priv to card->drvdata to fix this issue.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the hp mic pin has no VREF bits, the driver forgot to set PIN_IN
bit. Spotted during debugging old MacBook Airs.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a headphone jack is configurable as input, the generic parser
tries to make it retaskable as Headphone Mic. The switching can be
done smoothly if Capture Source control exists (i.e. there is another
input source). Or when user explicitly enables the creation of jack
mode controls, "Headhpone Mic Jack Mode" will be created accordingly.
However, if the headphone mic is the only input source, we have to
create "Headphone Mic Jack Mode" control because there is no capture
source selection. Otherwise, the generic parser assumes that the
input is constantly enabled, thus the headphone is permanently set
as input. This situation happens on the old MacBook Airs where no
input is supported properly, for example.
This patch fixes the problem: now "Headphone Mic Jack Mode" is created
when such an input selection isn't possible.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For Wireless USB audio devices, use multiple isoc packets per URB for
inbound endpoints with a datainterval < 5. This allows the WUSB host
controller to take advantage of bursting to service endpoints whose
logical polling interval is less than the 4ms minimum polling interval
limit in WUSB.
Signed-off-by: Thomas Pugliese <thomas.pugliese@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most Thinkpad Edge series laptops use conexant codec, so far although
the codecs have different minor Vendor Id and minor Subsystem Id,
they all belong to the cxt5066 family, this change can make the
mute/mic-mute LEDs support more generic among cxt_5066 family.
This design refers to the similar solution for the realtek codec
ALC269 family in the patch_realtek.c.
Cc: Alex Hung <alex.hung@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use bus->power_keep_link_on instead. The controller shouldn't go to
D3 when the link isn't reset, so essentially avoiding the link reset
means avoiding the runtime PM.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the missing pcbeep default amp for ALC668.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai