As the recent fix addressed the channel swap problem more properly,
update the comment as well.

Fixes: 1b7ecc24 ("ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109")
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200816084431.102151-1-marcan@marcan.stSigned-off-by: Takashi Iwai <tiwai@suse.de>

Fix spelling mistakes in the comments:
	initailise ==> initialise
	tranfer ==> transfer
	Initialse ==> Initialise
Signed-off-by: Liang Wang <wangliang101@huawei.com>
Link: https://lore.kernel.org/r/20200816071309.121461-1-wangliang101@huawei.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

Capture and playback endpoints on Saffire 6 (USB 1.1) resides on the same
interface. This was not supported by the composite quirk back in the day
when initial support for this device was added, thus only playback was
enabled until now.

Fixes: 11e424e8 ("ALSA: usb-audio: Add support for Focusrite Saffire 6 USB")
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable.vger.kernel.org>
Link: https://lore.kernel.org/r/20200815002103.29247-1-alexander@tsoy.meSigned-off-by: Takashi Iwai <tiwai@suse.de>

The Flex Book uses the same ALC298 codec as other Samsung laptops which
have the no headphone sound bug, like my Samsung Notebook. The Flex Book
owner used Early Patching to confirm that this quirk fixes the bug.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423Signed-off-by: Mike Pozulp <pozulp.kernel@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200814045346.645367-1-pozulp.kernel@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

Freeing chip on error may lead to an Oops at the next time
the system goes to resume. Fix this by removing all
snd_echo_free() calls on error.

Fixes: 47b5d028 ("ALSA: Echoaudio - Add suspend support #2")
Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn>
Link: https://lore.kernel.org/r/20200813074632.17022-1-dinghao.liu@zju.edu.cnSigned-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.comSigned-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: 40469064 ("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.deSigned-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: 87dc3648 ("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.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

The dsp_registers field of struct echoaduio has the volatile modifier,
but it's basically superfluous; the field is accessed only for the
base pointer of readl() and writel(), hence marking with __iomem alone
should suffice.  OTOH, having the volatile prefix causes a compile
warning like:
  sound/pci/echoaudio/echoaudio.c:1878:14: warning: passing argument 1 of 'iounmap' discards 'volatile' qualifier from pointer target type [-Wdiscarded-qualifiers]

So it's better to drop this superfluous modifier.

Link: https://lore.kernel.org/r/20200803143958.24324-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>

The USB device (0x17aa:0x1046) that support Lenovo P620 rear panel
line-in claim to support volume control, but it doens't seem to have an
AMP, so when line-in volume lowers below 80, nothing gets recorded
anymore.

Disable the volume control to workaround the issue.

Fixes: f8c11eb7 ("ALSA: usb-audio: Add support for Lenovo ThinkStation P620")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200810133108.31580-1-kai.heng.feng@canonical.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

This is just another Pioneer device with fixed endpoints. Input is dummy
but used as feedback (it always returns silence).

Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200810082502.225979-1-marcan@marcan.stSigned-off-by: Takashi Iwai <tiwai@suse.de>

Further investigation of the L-R swap problem on the MS2109 reveals that
the problem isn't that the channels are swapped, but rather that they
are swapped and also out of phase by one sample. In other words, the
issue is actually that the very first frame that comes from the hardware
is a half-frame containing only the right channel, and after that
everything becomes offset.

So introduce a new quirk field to drop the very first 2 bytes that come
in after the format is configured and a capture stream starts. This puts
the channels in phase and in the correct order.

Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200810082400.225858-1-marcan@marcan.stSigned-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.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

Matching by device matches all interfaces, which breaks the video/HID
portions of the device depending on module load order.

Fixes: e337bf19 ("ALSA: usb-audio: add quirk for MacroSilicon MS2109")
Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200810045319.128745-1-marcan@marcan.stSigned-off-by: Takashi Iwai <tiwai@suse.de>

The jack on Intel NUC 8 Rugged rear panel doesn't work.

The spec [1] states that the jack supports both headphone and
microphone, so override a Pin Complex which has both Amp-In and Amp-Out
to make the jack work.

Node 0x1b fits the requirement, and user confirmed the jack now works
with new pin config.

[1] https://www.intel.com/content/dam/support/us/en/documents/mini-pcs/NUC8CCH_TechProdSpec.pdf
BugLink: https://bugs.launchpad.net/bugs/1875199Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200807080514.15293-1-kai.heng.feng@canonical.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
Adds support for newer X-Fi Pro card, known as "Model No. SB1095"
with USB ID "041e:3263"
Signed-off-by: Mirko Dietrich <buzz@l4m1.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200806124850.20334-1-buzz@l4m1.deSigned-off-by: Takashi Iwai <tiwai@suse.de>

There is a spelling mistake in a usb_audio_dbg debug message. Also
replace "param" with "parameter".  Fix these.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20200806105134.46447-1-colin.king@canonical.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

Drop duplicated words in sound/pci/.
{and, the, at}
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Link: https://lore.kernel.org/r/20200806021926.32418-1-rdunlap@infradead.orgSigned-off-by: Takashi Iwai <tiwai@suse.de>

Drop duplicated words in sound/isa/.
{be, bit}
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Link: https://lore.kernel.org/r/20200806021916.32369-1-rdunlap@infradead.orgSigned-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.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

A variable dma_stop_delay is added as a new member in hdac_bus
structure to avoid memory decode error incase DMA RUN bit is not
disabled in the given timeout from snd_hdac_stream_sync function and
followed by stream reset which results in memory decode error between
reset set and clear operation.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Link: https://lore.kernel.org/r/20200805095221.5476-3-mkumard@nvidia.comSigned-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.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

Some ioctls via OSS sequencer API may race and lead to UAF when the
port create and delete are performed concurrently, as spotted by a
couple of syzkaller cases.  This patch is an attempt to address it by
serializing the ioctls with the existing register_mutex.

Basically OSS sequencer API is an obsoleted interface and was designed
without much consideration of the concurrency.  There are very few
applications with it, and the concurrent performance isn't asked,
hence this "big hammer" approach should be good enough.

Reported-by: syzbot+1a54a94bd32716796edd@syzkaller.appspotmail.com
Reported-by: syzbot+9d2abfef257f3e2d4713@syzkaller.appspotmail.com
Suggested-by: Hillf Danton <hdanton@sina.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200804185815.2453-1-tiwai@suse.deSigned-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.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

The dock sold from startech (PID: ICUSBAUDIO7D) has no friendly name
and shows up currently as "USB Sound Device" in ALSA.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20200804010616.3399256-1-cujomalainey@chromium.orgSigned-off-by: Takashi Iwai <tiwai@suse.de>

Lenovo ThinkStation P620 is like other TRX40 boards, is equipped with
two USB audio cards.

USB device (17aa:104d) provides functionality for Internal Speaker and
Front Headset. It's UAC v2, so it supports insertion control (jack
detection). However, when trying to get the connector status of the
speaker, an error occurs:
[    5.787405] usb 3-1: cannot get connectors status: req = 0x81, wValue = 0x200, wIndex = 0x1000, type = 0

Since the insertion control works perfectly for the headset, the error
for speaker is probably casued by connecting internally. So let's relax
the error for a bit if it's a speaker, and always reports it's connected.

USB device (17aa:1046) is for rear Line-in, Line-out and Microphone.
The insertion control works for all three jacks. However, there's an
Function Unit that doesn't work:
[    5.905415] usb 3-6: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4
[    5.905418] usb 3-6: 19:0: cannot get min/max values for control 12 (id 19)

So turn off the FU to avoid the error.

Also, add specific card name for both devices, so userspace can easily
indentify both cards.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200803142612.17156-1-kai.heng.feng@canonical.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

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 9a641848 ("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: 9a641848 ("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.comSigned-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.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

Add a new quirk ID for the Recon3D, as tested by me.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200803002928.8638-2-conmanx360@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

When the ZxR headphone gain control was added, the ca0132_switch_get
function was not updated, which meant that the changes to the control
state were not saved when entering/exiting alsamixer.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200803002928.8638-1-conmanx360@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

There are several Loongson-3 based laptops produced by CZC or Lemote,
they use alc269/alc662 codecs and need specific pin-tables, this patch
add their pin-tables.
Signed-off-by: Huacai Chen <chenhc@lemote.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1596360400-32425-1-git-send-email-chenhc@lemote.comSigned-off-by: Takashi Iwai <tiwai@suse.de>

GFP_KRENEL -> GFP_KERNEL
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>

Link: https://lore.kernel.org/r/1596224129-7699-1-git-send-email-Julia.Lawall@inria.frSigned-off-by: Takashi Iwai <tiwai@suse.de>

CONFIG_PCM_XRUN_DEBUG should be CONFIG_SND_PCM_XRUN_DEBUG
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>

Link: https://lore.kernel.org/r/1596223701-7558-1-git-send-email-Julia.Lawall@inria.frSigned-off-by: Takashi Iwai <tiwai@suse.de>

Merge series "ASoC: core: Two step component registration" from Cezary Rojewski <cezary.rojewski@intel.com>:

Provide a mechanism for true two-step component registration. This
mimics device registration flow where initialization is the first step
while addition goes as second in line. Drivers may choose to modify
component's fields before registering component to ASoC subsystem via
snd_soc_add_component.

Patchset achieves status quo - behavior of snd_soc_register_component
remains unchanged.

Cezary Rojewski (3):
  ASoC: core: Relocate and expose snd_soc_component_initialize
  ASoC: core: Simplify snd_soc_component_initialize declaration
  ASoC: core: Two step component registration

 include/sound/soc-component.h         |  3 --
 include/sound/soc.h                   | 11 +++---
 sound/soc/soc-component.c             | 16 ---------
 sound/soc/soc-core.c                  | 52 +++++++++++++++++----------
 sound/soc/soc-generic-dmaengine-pcm.c | 14 +++++---
 sound/soc/stm/stm32_adfsdm.c          |  9 +++--
 6 files changed, 55 insertions(+), 50 deletions(-)

--
2.17.1

Modify snd_soc_add_component so it calls snd_soc_component_initialize
no longer and thus providing true two-step registration. Drivers may
choose to change component's fields before actually adding it to ASoC
subsystem.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200731144146.6678-4-cezary.rojewski@intel.comSigned-off-by: Mark Brown <broonie@kernel.org>

Move 'name' field initialization responsibility back to
snd_soc_component_initialize to prepare snd_soc_add_component function
for being called separatelly as a second registration step.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200731144146.6678-3-cezary.rojewski@intel.comSigned-off-by: Mark Brown <broonie@kernel.org>