Back-merge to apply the tasklet conversion patches that are based
on the already applied tasklet API changes on 5.9-rc4.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Mic connects to the Nid 0x19, but the configuration of Nid 0x19
is not defined to Mic, and also need to set the coeff to enable the
auto detection on the Nid 0x19. After this change, the Mic plugging
in or plugging out could be detected and could record the sound from
the Mic.
And the coeff value is suggested by Kailang of Realtek.
Cc: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200909020041.8967-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There've been quite a few regression reports about the lowered volume
(reduced to ca 65% from the previous level) on Lenovo Thinkpad X1
after the commit d2cd795c4e ("ALSA: hda - fixup for the bass speaker
on Lenovo Carbon X1 7th gen"). Although the commit itself does the
right thing from HD-audio POV in order to have a volume control for
bass speakers, it seems that the machine has some secret recipe under
the hood.
Through experiments, Benjamin Poirier found out that the following
routing gives the best result:
* DAC1 (NID 0x02) -> Speaker pin (NID 0x14)
* DAC2 (NID 0x03) -> Shared by both Bass Speaker pin (NID 0x17) &
Headphone pin (0x21)
* DAC3 (NID 0x06) -> Unused
DAC1 seems to have some equalizer internally applied, and you'd get
again the output in a bad quality if you connect this to the
headphone pin. Hence the headphone is connected to DAC2, which is now
shared with the bass speaker pin. DAC3 has no volume amp, hence it's
not connected at all.
For achieving the routing above, this patch introduced a couple of
workarounds:
* The connection list of bass speaker pin (NID 0x17) is reduced not to
include DAC3 (NID 0x06)
* Pass preferred_pairs array to specify the fixed connection
Here, both workarounds are needed because the generic parser prefers
the individual DAC assignment over others.
When the routing above is applied, the generic parser creates the two
volume controls "Front" and "Bass Speaker". Since we have only two
DACs for three output pins, those are not fully controlling each
output individually, and it would confuse PulseAudio. For avoiding
the pitfall, in this patch, we rename those volume controls to some
unique ones ("DAC1" and "DAC2"). Then PulseAudio ignore them and
concentrate only on the still good-working "Master" volume control.
If a user still wants to control each DAC volume, they can still
change manually via "DAC1" and "DAC2" volume controls.
Fixes: d2cd795c4e ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen")
Reported-by: Benjamin Poirier <benjamin.poirier@gmail.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Benjamin Poirier <benjamin.poirier@gmail.com>
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207407#c10
BugLink: https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3214171
BugLink: https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3276276
Link: https://lore/kernel.org/r/20200829112746.3118-1-benjamin.poirier@gmail.com
Link: https://lore.kernel.org/r/20200903083300.6333-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix cppcheck warning:
sound/pci/hda/hda_auto_parser.c:353:7: style: Local variable 'i'
shadows outer variable [shadowVariable]
int i = 0;
^
sound/pci/hda/hda_auto_parser.c:182:6: note: Shadowed declaration
int i;
^
sound/pci/hda/hda_auto_parser.c:353:7: note: Shadow variable
int i = 0;
^
It's not clear why a new declaration was added, remove and reuse
variable declared with larger scope.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On SKL+ Intel platforms, the driver selection is handled by the
snd_intel_dspcfg, and when the HDaudio legacy driver is not selected,
be it with the auto-selection or user preferences with a kernel
parameter, the probe aborts with no logs, only a -ENODEV return value.
Having no dmesg trace, even with dynamic debug enabled, makes support
more complicated than it needs to be, and even experienced users can
be fooled. A simple dev_dbg() trace solves this problem.
BugLink: https://github.com/thesofproject/linux/issues/2330
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200902154239.1440537-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Following Christian Lachner's patch for Gigabyte X570-based motherboards,
also patch the MSI X570-A PRO motherboard; the ALC1220 codec requires the
same workaround for Clevo laptops to enforce the DAC/mixer connection
path. Set up a quirk entry for that.
I suspect most if all X570 motherboards will require similar patches.
[ The entries reordered in the SSID order -- tiwai ]
Related buglink: https://bugzilla.kernel.org/show_bug.cgi?id=205275
Signed-off-by: Dan Crawford <dnlcrwfrd@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200829024946.5691-1-dnlcrwfrd@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When system is suspended with active audio playback to HDMI/DP, two
alternative sequences can happen at resume:
a) monitor is detected first and ALSA prepare follows normal
stream setup sequence, or
b) ALSA prepare is called first, but monitor is not yet detected,
so PCM is restarted without a pin,
In case of (b), on i915 systems, haswell_verify_D0() is not called at
resume and the pin power state may be incorrect. Result is lack of audio
after resume with no error reported back to user-space.
Fix the problem by always verifying converter and pin state in the
i915_pin_cvt_fixup().
BugLink: https://github.com/thesofproject/linux/issues/2388
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200826170306.701566-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Galaxy Book Ion NT950XCJ-X716A (15 inches) uses the same ALC298
codec as other Samsung laptops which have the no headphone sound bug. I
confirmed on my own hardware that this fixes the bug.
This also correct the model name for the 13 inches version. It was
incorrectly referenced as NT950XCJ-X716A in commit e17f02d05. But it
should have been NP930XCJ-K01US.
Fixes: e17f02d055 ("ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423
Signed-off-by: Adrien Crivelli <adrien.crivelli@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200826084014.211217-1-adrien.crivelli@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add headphone gain and DAC filter controls, which use the same commands
as the AE-5. Also, change input source enumerated control item count to
exclude front microphone.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-20-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add AE-7 quirk data for setting of microphone. The AE-7 has no front
panel connector, so only rear-mic/line-in have new commands.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-19-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add structures containing the changes that need to happen on output
selection for each quirk. This should streamline the addition of new
quirks.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-9-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the output structures that were in use before and instead set the
DSP commands line by line. Now that the commands use is known, it makes
the functionality more clear this way.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-8-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the surround output selection and merge it with the speaker
output selection. Now that the extra commands that were being run on
surround output setting are known, there's no need to have it be
separate.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-7-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add bass redirection controls for surround outputs. This uses the DSP to
redirect audio below the bass redirection crossover frequency to the LFE
channel from the front/rear L/R speakers. This only goes into effect if
the speakers aren't set as full range, and only if the surround
configuration has an LFE channel.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-6-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add functions for setting full-range speakers and controls to
enable/disable the setting. Setting a speaker to full-range means that
the channels won't have their bass redirected to the LFE channel.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-5-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a surround channel configuration enumeration control. Setting up
different channel configurations allows the DSP to upmix stereo audio
into multi-channel audio, and allows for redirection of bass to a
subwoofer.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-4-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cleanup the ca0132_mmio_init function, separating into two separate
functions, one for Sound Blaster Z/ZxR/Recon3D, and another for the
AE-5.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-2-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 61eee4a7fc ("ALSA: hda: Add support for Loongson
7A1000 controller") to fix the following error on the Loongson LS7A
platform:
rcu: INFO: rcu_preempt self-detected stall on CPU
<SNIP>
NMI backtrace for cpu 0
CPU: 0 PID: 68 Comm: kworker/0:2 Not tainted 5.8.0+ #3
Hardware name: , BIOS
Workqueue: events azx_probe_work [snd_hda_intel]
<SNIP>
Call Trace:
[<ffffffff80211a64>] show_stack+0x9c/0x130
[<ffffffff8065a740>] dump_stack+0xb0/0xf0
[<ffffffff80665774>] nmi_cpu_backtrace+0x134/0x140
[<ffffffff80665910>] nmi_trigger_cpumask_backtrace+0x190/0x200
[<ffffffff802b1abc>] rcu_dump_cpu_stacks+0x12c/0x190
[<ffffffff802b08cc>] rcu_sched_clock_irq+0xa2c/0xfc8
[<ffffffff802b91d4>] update_process_times+0x2c/0xb8
[<ffffffff802cad80>] tick_sched_timer+0x40/0xb8
[<ffffffff802ba5f0>] __hrtimer_run_queues+0x118/0x1d0
[<ffffffff802bab74>] hrtimer_interrupt+0x12c/0x2d8
[<ffffffff8021547c>] c0_compare_interrupt+0x74/0xa0
[<ffffffff80296bd0>] __handle_irq_event_percpu+0xa8/0x198
[<ffffffff80296cf0>] handle_irq_event_percpu+0x30/0x90
[<ffffffff8029d958>] handle_percpu_irq+0x88/0xb8
[<ffffffff80296124>] generic_handle_irq+0x44/0x60
[<ffffffff80b3cfd0>] do_IRQ+0x18/0x28
[<ffffffff8067ace4>] plat_irq_dispatch+0x64/0x100
[<ffffffff80209a20>] handle_int+0x140/0x14c
[<ffffffff802402e8>] irq_exit+0xf8/0x100
Because AZX_DRIVER_GENERIC can not work well for Loongson LS7A HDA
controller, it needs some workarounds which are not merged into the
upstream kernel at this time, so it should revert this patch now.
Fixes: 61eee4a7fc ("ALSA: hda: Add support for Loongson 7A1000 controller")
Cc: <stable@vger.kernel.org> # 5.9-rc1+
Signed-off-by: Tiezhu Yang <yangtiezhu@loongson.cn>
Link: https://lore.kernel.org/r/1598348388-2518-1-git-send-email-yangtiezhu@loongson.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WAKEEN bits are used to indicate which bits in the
STATESTS register may cause wake event during the codec
state change request. Configure the WAKEEN register for
the Tegra to detect the wake events.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Acked-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/20200825052415.20626-3-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Tegra HDA codec HW implementation has an issue related to not
swapping the 2 channel Audio Sample Packet(ASP) channel mapping.
Whatever the FL and FR mapping specified the left channel always
comes out of left speaker and right channel on right speaker. So
add condition to disallow the swapping of FL,FR during the playback.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Acked-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/20200825052415.20626-2-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's another HP desktop has buggy BIOS which flags the Port
Connectivity bit as no connection.
Apply force connectivity quirk to enable DP/HDMI audio.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200811095336.32396-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous fix forgot to remove the unused variable that triggers a
compile warning now:
sound/pci/hda/patch_realtek.c: In function 'alc285_fixup_hp_gpio_led':
sound/pci/hda/patch_realtek.c:4163:19: warning: unused variable 'spec' [-Wunused-variable]
Fix it.
Fixes: 404690649e ("ALSA: hda - reverse the setting value in the micmute_led_set")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Link: https://lore.kernel.org/r/20200812070256.32145-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Before the micmute_led_set() is introduced, the function of
alc_gpio_micmute_update() will set the gpio value with the
!micmute_led.led_value, and the machines have the correct micmute led
status. After the micmute_led_set() is introduced, it sets the gpio
value with !!micmute_led.led_value, so the led status is not correct
anymore, we need to set micmute_led_polarity = 1 to workaround it.
Now we fix the micmute_led_set() and remove micmute_led_polarity = 1.
Fixes: 87dc36482c ("ALSA: hda/realtek - Add LED class support for micmute LED")
Reported-and-suggested-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200811122430.6546-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After installing the Ubuntu Linux, the micmute led status is not
correct. Users expect that the led is on if the capture is disabled,
but with the current kernel, the led is off with the capture disabled.
We tried the old linux kernel like linux-4.15, there is no this issue.
It looks like we introduced this issue when switching to the led_cdev.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200810021659.7429-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tegra HDA has audio data buffer for upto tens of frames, this buffer
can help to avoid underflow. HW will keep issuing new data fetch
request when buffers are not full and current BDL is not done. When SW
disable DMA RUN bit for a stream, HW can't cancel the already issued data
fetch request and hence it can't stop DMA. HW has to wait for all issued
data fetch request get data returned before it stops DMA.
This HW behavior is not in sync with HDA spec which says DMA RUN bit
should be cleared within 1 audio frame. For Tegra, DMA RUN bit was
active for more than one audio frame, due to this the timeout in
snd_hdac_stream_sync function is not helping. When Stream reset set
and clear happens during DMA RUN bit active state it results in Memory
Decode error.
Unfortunately, there is no way to detect when these data accesses have
completed, but testing has shown that a 100us delay between Stream reset
set and clear operation for Tegra avoids the memory decode error.
Therefore, adding a 100us dma stop delay.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Link: https://lore.kernel.org/r/20200805095221.5476-4-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set chip->align_buffer_size to 1 for Tegra platforms to make the buffer
alignment to be multiple of 128 bytes. This fix is applied as gstreamer
alsasink gets stuck with the default buffer-time and latency-time
parameters with 4 byte buffer alignment.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Link: https://lore.kernel.org/r/20200805095221.5476-2-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDMI on some platforms doesn't enable audio support because its Port
Connectivity [31:30] is set to AC_JACK_PORT_NONE:
Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0b000094: OUT Detect HBR HDMI DP
Pin Default 0x58560010: [N/A] Digital Out at Int HDMI
Conn = Digital, Color = Unknown
DefAssociation = 0x1, Sequence = 0x0
Pin-ctls: 0x40: OUT
Unsolicited: tag=00, enabled=0
Power states: D0 D3 EPSS
Power: setting=D0, actual=D0
Devices: 0
Connection: 3
0x02 0x03* 0x04
For now, use a quirk to force connectivity based on SSID. If there are
more platforms affected by the same issue, we can eye for a more generic
solution.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200804155836.16252-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The biggest changes here one again come from Mormioto-san who has
continued his dilligent work cleaning up long standing issues in the
APIs, it's particularly nice to see the transition from digital_mute()
to mute_stream() finally completed. There's also been a lot of work on
the x86 code again, this time a big focus has been on cleaning up some
issues identified by various static tests, and on the Freescale systems.
Otherwise the biggest thing has been a lot of driver additions:
- Convert users of digital_mute() to mute_stream().
- Simplify I/O helper functions.
- Add a helper for getting the RTD from a substream.
- Many, many fixes and cleanups to the x86 code.
- New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim
MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia
Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some
of the first phones I worked on!) and TI J721e EVM.
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Merge tag 'asoc-v5.9' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.9
The biggest changes here one again come from Mormioto-san who has
continued his dilligent work cleaning up long standing issues in the
APIs, it's particularly nice to see the transition from digital_mute()
to mute_stream() finally completed. There's also been a lot of work on
the x86 code again, this time a big focus has been on cleaning up some
issues identified by various static tests, and on the Freescale systems.
Otherwise the biggest thing has been a lot of driver additions:
- Convert users of digital_mute() to mute_stream().
- Simplify I/O helper functions.
- Add a helper for getting the RTD from a substream.
- Many, many fixes and cleanups to the x86 code.
- New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim
MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia
Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some
of the first phones I worked on!) and TI J721e EVM.
This reverts commit 9a6418487b ("ALSA: hda: call runtime_allow()
for all hda controllers").
The reverted patch already introduced some regressions on some
machines:
- on gemini-lake machines, the error of "azx_get_response timeout"
happens in the hda driver.
- on the machines with alc662 codec, the audio jack detection doesn't
work anymore.
Fixes: 9a6418487b ("ALSA: hda: call runtime_allow() for all hda controllers")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208511
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200803064638.6139-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ca0113 command had the wrong group_id, 0x48 when it should've been
0x30. The front microphone selection should now work.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200803002928.8638-3-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
