Apply the same fixup for Thinkpad with dock to Thinkpad X1 Carbon 2nd,
too. This reduces the annoying loud cracking noise problem, as well
as the support of missing docking port.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=958439
Reported-and-tested-by: Benjamin Poirier <bpoirier@suse.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo Thinkpads with Realtek codecs may still have some loud
crackling noises at reboot/shutdown even though a few previous fixes
have been applied. It's because the previous fix (disabling the
default shutup callback) takes effect only at transition of the codec
power state. Meanwhile, at reboot or shutdown, we don't take down the
codec power as default, thus it triggers the same problem unless the
codec is powered down casually by runtime PM.
This patch tries to address the issue. It gives two things:
- implement the separate reboot_notify hook to struct alc_spec, and
call it optionally if defined.
- turn off the codec to D3 for Thinkpad models via this new callback
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=958439
Reported-and-tested-by: Benjamin Poirier <bpoirier@suse.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that a workaround for Thinkpad T440s crackling noise can be
applied generically to all Thinkpad models: namely, disabling the
default alc269 shutup callback. This patch moves it to the existing
alc_fixup_tpt440_dock() while also replacing the rest code with
another existing alc_fixup_disable_aamix(). It resulted in a good
code reduction.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=958439
Reported-and-tested-by: Benjamin Poirier <bpoirier@suse.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These laptops support both headphone, headset and mic modes
for the 3.5mm jack.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1526330
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoid getting sample rate on AudioQuest DragonFly as it is unsupported
and causes noisy "cannot get freq at ep 0x1" messages when playback
starts.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AudioQuest DragonFly DAC reports a volume control range of 0..50
(0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which
is obviously incorrect and would cause software using the dB information
in e.g. volume sliders to have a massive volume difference in 100..102%
range.
Commit 2d1cb7f658 ("ALSA: usb-audio: add dB range mapping for some
devices") added a dB range mapping for it with range 0..50 dB.
However, the actual volume mapping seems to be neither linear volume nor
linear dB scale, but instead quite close to the cubic mapping e.g.
alsamixer uses, with a range of approx. -53...0 dB.
Replace the previous quirk with a custom dB mapping based on some basic
output measurements, using a 10-item range TLV (which will still fit in
alsa-lib MAX_TLV_RANGE_SIZE).
Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the
range is 0..50, so if this gets fixed/changed in later HW revisions it
will no longer be applied.
v2: incorporated Takashi Iwai's suggestion for the quirk application
method
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Alienware 17 (2015) has the same card and pin configuration of the
Alienware 15, so the same quirks must be applied.
Signed-off-by: Gabriele Martino <g.martino@gmx.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo Thinkpad T440s suffers from constant background noises, and it
seems to be a generic hardware issue on this model:
https://forums.lenovo.com/t5/ThinkPad-T400-T500-and-newer-T/T440s-speaker-noise/td-p/1339883
As the noise comes from the analog loopback path, disabling the path
is the easy workaround.
Also, the machine gives significant cracking noises at PM suspend. A
workaround found by trial-and-error is to disable the shutup callback
currently used for ALC269-variant.
This patch addresses these noise issues by introducing a new fixup
chain. Although the same workaround might be applicable to other
Thinkpad models, it's applied only to T440s (17aa:220c) in this patch,
so far, just to be safe (you chicken!). As a compromise, a new model
option string "tp440" is provided now, though, so that owners of other
Thinkpad models can test it more easily.
Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=958504
Reported-and-tested-by: Tim Hardeck <thardeck@suse.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have two latest thinkpad laptop models which are all based on the
Intel skylake platforms, and all of them have the codec alc293 on
them. When the machines boot to the desktop, an greeting dialogue
shows up with the notification sound. But on these two models, there
is noise with the notification sound. We have 3 SKUs for each of
the models, all of them have this problem.
So far, this problem is only specific to these two thinkpad models,
we did not find this problem on the old thinkpad models with the
codec alc293 or alc292.
A workaround for this problem is disabling the aamix.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1523517
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the internal mic of the Packard Bell DOTS, one channel
has an inverted signal. Add a quirk to fix this up.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1523232
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In BXT-P A0, HD-Audio DMA requests is later than expected,
and makes an audio stream sensitive to system latencies when
24/32 bits are playing.
Adjusting threshold of DMA fifo to force the DMA request
sooner to improve latency tolerance at the expense of power.
v2: move Intel specific code to hda_intel.c
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
rme96 driver needs to reset DAC depending on the sample rate, and this
results in resetting to the max volume suddenly. It's because of the
missing call of snd_rme96_apply_dac_volume().
However, calling this function right after the DAC reset still may not
work, and we need some delay before this call. Since the DAC reset
and the procedure after that are performed in the spinlock, we delay
the DAC volume restore at the end after the spinlock.
Reported-and-tested-by: Sylvain LABOISNE <maeda1@free.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The numbers aren't always linear, just like in the real world.
Correct to the right numbers stated in the datasheet (although we
can't trust the datasheet as well).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent addition of ELD notifier for Intel HDMI/DP codec may lead
the bad codec connection found as kernel messages like below:
Suspending console(s) (use no_console_suspend to debug)
hdmi_present_sense: snd_hda_codec_hdmi hdaudioC0D2: HDMI status: Codec=2 Pin=6 Presence_Detect=1 ELD_Valid=1
snd_hda_intel 0000:00:1f.3: spurious response 0x0:0x2, last cmd=0x206f2e08
snd_hda_intel 0000:00:1f.3: spurious response 0x0:0x2, last cmd=0x206f2e08
....
snd_hda_codec_hdmi hdaudioC0D2: HDMI: ELD buf size is 0, force 128
snd_hda_intel 0000:00:1f.3: azx_get_response timeout, switching to polling mode: last cmd=0x206f2f00
snd_hda_intel 0000:00:1f.3: No response from codec, disabling MSI: last cmd=0x206f2f00
snd_hda_intel 0000:00:1f.3: azx_get_response timeout, switching to single_cmd mode: last cmd=0x206f2f00
azx_single_wait_for_response: 42 callbacks suppressed
This seems appearing when the sound driver went to suspend before i915
driver. Then i915 driver disables HDMI/DP audio bit and calls the
registered notifier, and the HDA codec tries to handle it as a
hot(un)plug. But since the driver is already in the suspended state,
it fails miserably.
As this is a sort of spurious wakeup, it can be ignored safely, as
long as it's delivered during the system suspend. OTOH, if a
notification comes during the runtime suspend, the situation is
different: we need to wake up. But during the system suspend, such a
notification can't be the reason for a wakeup.
This patch addresses it by a simple check of the current sound card
status. The skipped notification doesn't matter because the HDA
driver will check the plugged status forcibly at the resume in
return.
Then, why the card status, not a runtime PM status or else? The HDA
controller driver is supposed to set the card status to D3 at the
system suspend but not at the runtime suspend. So we can see it as a
flag that is set only for the system suspend. Admittedly, it's a bit
ugly, but it should work well for now.
Reported-and-tested-by: "Zhang, Xiong Y" <xiong.y.zhang@intel.com>
Fixes: 25adc137c5 ('ALSA: hda - Wake the codec up on pin/ELD notify events')
Cc: <stable@vger.kernel.org> # v4.3+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Quite a large batch of fixes have come in since the merge window, mainly
driver specific ones but there's a couple of core ones:
- A fix for DAPM resume on active streams to ensure everything ends up
cleanly in the right state.
- Reset the DAPM cache when freeing widgets to fix a crash on driver
remove and reload.
The PM functions for nau8825 are new code which fix crashes on resume.
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Merge tag 'asoc-fix-v4.4-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.4
Quite a large batch of fixes have come in since the merge window, mainly
driver specific ones but there's a couple of core ones:
- A fix for DAPM resume on active streams to ensure everything ends up
cleanly in the right state.
- Reset the DAPM cache when freeing widgets to fix a crash on driver
remove and reload.
The PM functions for nau8825 are new code which fix crashes on resume.
For DAPM resume, we should first change the power state of the
card and then recheck the endpoints. This ensures the dapm is
resumed first and then userspace can resume the streams.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix kernel-doc warnings in soc-ops.c:
..//sound/soc/soc-ops.c:415: warning: No description found for parameter 'ucontrol'
..//sound/soc/soc-ops.c:415: warning: Excess function parameter 'uinfo' description in 'snd_soc_put_volsw_sx'
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add platform specific data for Terra project.
Signed-off-by: Luke_Yin@asus.com <Luke_Yin@asus.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Correct valid data word register value for 24 bit data width. The
bit value should be 10 (aka 0x2), not 0x10.
This fixes playback of 24 bit audio.
Signed-off-by: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
Reviewed-by: Caesar Wang <wxt@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A new randconfig build failure shows that the fsl-asoc-card module
must not be built-in when the AC97 driver is a loadable module:
sound/built-in.o: In function `fsl_asoc_card_late_probe':
:(.text+0x571d8): undefined reference to `snd_ac97_update_bits'
I couldn't come up with a nice solution, so this adds another dependency
on "X || !X", which is the Kconfig way of saying that we have an
optional dependency on something that might be a loadable module.
Fixes: 50760cad9d ("ASoC: fsl-asoc-card: add AC'97 support")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Gigabyte Z710X mobo with ALC1150 codec gets significant noises from
the analog loopback routes even if their inputs are all muted.
Simply kill the aamix for fixing it.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=108301
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have a machine Dell XPS 13 with the codec alc256, after resume back
from S3, the headphone has noise when play sound.
Through comparing with the coeff vaule before and after S3, we found
restoring a coeff register will help remove noise.
BugLink: https://bugs.launchpad.net/bugs/1519168
Cc: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have requested the firmware but missed releasing it.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
When McASP is used as TX/RX synchronous (TX side generating clocks for RX
side also) and only capture is used we need to configure the number of TX
slots in order McASP to be able to generate the Frame sync.
Fixes: 9273de1940d9e ("ASoC: davinci-mcasp: Add set_tdm_slots() support")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It turned out that many HP laptops suffer from the same problem as
fixed in commit [c932b98c1e: ALSA: hda - Apply pin fixup for HP
ProBook 6550b]. But, it's tiresome to list up all such PCI SSIDs, as
there are really lots of HP machines.
Instead, we do a bit more clever, try to check the supposedly dock and
built-in headphone pins, and apply the fixup when both seem valid.
This rule can be applied generically to all models using the same
quirk, so we'll fix all in a shot.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=107491
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These are all off by one; the playback and bypass switches are the top
two bits of the registers, which are at shifts 7 and 6 not 8 and 7.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set substream player private data.
substream player private data is used in uni_player_irq_handler to lock,
stop & unlock the stream when interrupt indicates underflow/overflow.
If not set, then segmentation fault occurs.
Signed-off-by: Moise Gergaud <moise.gergaud@st.com>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
"st," prefix has been added for ST proprietary DT properties.
Signed-off-by: Moise Gergaud <moise.gergaud@st.com>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For making the speakers on Acer Aspire One Cloudbook 14 to work, we
need the as same quirk as for another Chromebook. This patch adds the
corresponding fixup entry.
Reported-by: Patrick <epictetus@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For SKL, only the HDMI codec is in the display power well while the
HD-A controller isn't. So the codec flag 'link_power_control' is
set to request/release the display power via bus link_power ops.
For BXT, the power well design is the same as SKL, so the patch
should be applied to BXT too.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add HD Audio Device PCI ID for the Intel Broxton platform.
It is an HDA Intel PCH controller.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The i2c is dependency for the i2c codec drivers, so machine should depend on
i2c. WIthout this we get build failures if I2C is not selected
sound/soc/codecs/rl6347a.c: In function 'rl6347a_hw_write':
>> sound/soc/codecs/rl6347a.c:66:8: error: implicit declaration of function
>> 'i2c_master_send' [-Werror=implicit-function-declaration]
ret = i2c_master_send(client, data, 4);
^
sound/soc/codecs/rl6347a.c: In function 'rl6347a_hw_read':
>> sound/soc/codecs/rl6347a.c:114:8: error: implicit declaration of function
>> 'i2c_transfer' [-Werror=implicit-function-declaration]
ret = i2c_transfer(client->adapter, xfer, 2);
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
topology core used wrong callback for TLV bytes control, it should be
snd_soc_bytes_info_ext and not snd_soc_bytes_info
Signed-off-by: Omair M Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The bit allocation for PLL source is 0x80 [13:11] instead of [12:11]
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds pm function and fixes following issues
1.i2c timeout after resume, after resume we saw interrupt handler
is called prior to i2c controller is resumed.This causes i2c timeout
2.no audio after resume
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The rk_spdif_probe uses the device match data as a token to identify a
particular device, but accidentally casts a pointer to 'int', which is
not portable, as gcc points out in this warning on arm64:
rockchip_spdif.c: In function 'rk_spdif_probe':
rockchip_spdif.c:283:6: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
This changes the logic to compare two pointer values instead, using
the same cast that was used for initializing the value in the first
place.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Unmuting headphone has pop noise in particular hardware design. So we extend
the delay time in headphone unmuting sequence to avoid pop.
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The maximum DMIC clock rate is 3.072 MHz for most DMIC. And it will get better
performance in higher clock rate. If we set maximum to 3 MHz in driver, we will
get a clock rate which is not even close to 3 MHz.
For example, if DMIC clock source is 24.576 MHz, the DMIC clock will be about
1.5 MHz in current code. But it will be 3.072 MHz with this patch.
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
One of the many faults of the QinHeng CH345 USB MIDI interface chip is
that it does not handle received SysEx messages correctly -- every second
event packet has a wrong code index number, which is the one from the last
seen message, instead of 4. For example, the two messages "FE F0 01 02 03
04 05 06 07 08 09 0A 0B 0C 0D 0E F7" result in the following event
packets:
correct: CH345:
0F FE 00 00 0F FE 00 00
04 F0 01 02 04 F0 01 02
04 03 04 05 0F 03 04 05
04 06 07 08 04 06 07 08
04 09 0A 0B 0F 09 0A 0B
04 0C 0D 0E 04 0C 0D 0E
05 F7 00 00 05 F7 00 00
A class-compliant driver must interpret an event packet with CIN 15 as
having a single data byte, so the other two bytes would be ignored. The
message received by the host would then be missing two bytes out of six;
in this example, "F0 01 02 03 06 07 08 09 0C 0D 0E F7".
These corrupted SysEx event packages contain only data bytes, while the
CH345 uses event packets with a correct CIN value only for messages with
a status byte, so it is possible to distinguish between these two cases by
checking for the presence of this status byte.
(Other bugs in the CH345's input handling, such as the corruption resulting
from running status, cannot be worked around.)
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
