1)
In snd_hda_pick_fixup(), quirks are first matched by PCI SSID and then, if
there is no match, by codec SSID. The Lenovo "ThinkPad X1 Carbon 7th" has
an audio chip with PCI SSID 0x2292 and codec SSID 0x2293[1]. Therefore, fix
the quirk meant for that device to match on .subdevice == 0x2292.
2)
The "Thinkpad X1 Yoga 7th" does not exist. The companion product to the
Carbon 7th is the Yoga 4th. That device has an audio chip with PCI SSID
0x2292 and codec SSID 0x2292[2]. Given the behavior of
snd_hda_pick_fixup(), it is not possible to have a separate quirk for the
Yoga based on SSID. Therefore, merge the quirks meant for the Carbon and
Yoga. This preserves the current behavior for the Yoga.
[1] This is the case on my own machine and can also be checked here
https://github.com/linuxhw/LsPCI/tree/master/Notebook/Lenovo/ThinkPad
https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3225701
[2]
https://github.com/linuxhw/LsPCI/tree/master/Convertible/Lenovo/ThinkPad
https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3176355
Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Fixes: 54a6a7dc107d ("ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen") Cc: Jaroslav Kysela <perex@perex.cz> Cc: Kailang Yang <kailang@realtek.com> Tested-by: Vincent Bernat <vincent@bernat.ch> Tested-by: Even Brenden <evenbrenden@gmail.com> Signed-off-by: Benjamin Poirier <benjamin.poirier@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200703080005.8942-2-benjamin.poirier@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kai Vehmanen [Fri, 3 Jul 2020 15:38:17 +0000 (18:38 +0300)]
ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
When HDMI PCM devices are opened in a specific order, with at least one
HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the
connected monitor, on recent Intel platforms (ICL/JSL and newer). While
this is not a typical sequence, at least Pulseaudio does this every time
when it is started, to discover the available PCMs.
The rootcause is an invalid assumption in hdmi_add_pin(), where the
total number of converters is assumed to be known at the time the
function is called. On older Intel platforms this held true, but after
ICL/JSL, the order how pins and converters are in the subnode list as
returned by snd_hda_get_sub_nodes(), was changed. As a result,
information for some converters was not stored to per_pin->mux_nids.
And this means some pins cannot be connected to all converters, and
application instead gets -EBUSY instead at open.
The assumption that converters are always before pins in the subnode
list, is not really a valid one. Fix the problem in hdmi_parse_codec()
by introducing separate loops for discovering converters and pins.
Alexander Tsoy [Mon, 29 Jun 2020 02:59:33 +0000 (05:59 +0300)]
ALSA: usb-audio: Fix packet size calculation
Commit f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation")
introduced a regression for devices which have playback endpoints with
bInterval > 1. Fix this by taking ep->datainterval into account.
Note that frame and fps are actually mean packet and packets per second
in the code introduces by the mentioned commit. This will be fixed in a
follow-up patch.
Jaroslav Kysela [Thu, 25 Jun 2020 11:58:29 +0000 (13:58 +0200)]
AsoC: amd: add missing snd- module prefix to the acp3x-rn driver kernel module
Signed-off-by: Jaroslav Kysela <perex@perex.cz> Acked-by: Alex Deucher <alexander.deucher@amd.com> Cc: Mark Brown <broonie@kernel.org> Cc: vijendar.mukunda@amd.com Cc: Alexander.Deucher@amd.com Link: https://lore.kernel.org/r/20200625115829.791750-1-perex@perex.cz Signed-off-by: Mark Brown <broonie@kernel.org>
Hui Wang [Thu, 25 Jun 2020 08:38:33 +0000 (16:38 +0800)]
ALSA: hda - let hs_mic be picked ahead of hp_mic
We have a Dell AIO, there is neither internal speaker nor internal
mic, only a multi-function audio jack on it.
Users reported that after freshly installing the OS and plug
a headset to the audio jack, the headset can't output sound. I
reproduced this bug, at that moment, the Input Source is as below:
Simple mixer control 'Input Source',0
Capabilities: cenum
Items: 'Headphone Mic' 'Headset Mic'
Item0: 'Headphone Mic'
That is because the patch_realtek will set this audio jack as mic_in
mode if Input Source's value is hp_mic.
If it is not fresh installing, this issue will not happen since the
systemd will run alsactl restore -f /var/lib/alsa/asound.state, this
will set the 'Input Source' according to history value.
If there is internal speaker or internal mic, this issue will not
happen since there is valid sink/source in the pulseaudio, the PA will
set the 'Input Source' according to active_port.
To fix this issue, change the parser function to let the hs_mic be
stored ahead of hp_mic.
Takashi Iwai [Wed, 24 Jun 2020 12:23:40 +0000 (14:23 +0200)]
ALSA: usb-audio: Fix OOB access of mixer element list
The USB-audio mixer code holds a linked list of usb_mixer_elem_list,
and several operations are performed for each mixer element. A few of
them (snd_usb_mixer_notify_id() and snd_usb_mixer_interrupt_v2())
assume each mixer element being a usb_mixer_elem_info object that is a
subclass of usb_mixer_elem_list, cast via container_of() and access it
members. This may result in an out-of-bound access when a
non-standard list element has been added, as spotted by syzkaller
recently.
This patch adds a new field, is_std_info, in usb_mixer_elem_list to
indicate that the element is the usb_mixer_elem_info type or not, and
skip the access to such an element if needed.
Macpaul Lin [Tue, 23 Jun 2020 11:03:23 +0000 (19:03 +0800)]
ALSA: usb-audio: add quirk for Samsung USBC Headset (AKG)
We've found Samsung USBC Headset (AKG) (VID: 0x04e8, PID: 0xa051)
need a tiny delay after each class compliant request.
Otherwise the device might not be able to be recognized each times.
Shengjiu Wang [Tue, 23 Jun 2020 06:01:11 +0000 (14:01 +0800)]
ASoC: fsl_mqs: Don't check clock is NULL before calling clk API
Because clk_prepare_enable and clk_disable_unprepare should
check input clock parameter is NULL or not internally, then
we don't need to check them before calling the function.
ALSA: usb-audio: Add registration quirk for Kingston HyperX Cloud Flight S
Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud
Alpha S (0951:0x16ea) uses two interfaces, but only the second
interface contains the capture stream. This patch delays the
registration until the second interface appears.
Takashi Iwai [Mon, 22 Jun 2020 11:49:14 +0000 (13:49 +0200)]
Merge tag 'asoc-fix-v5.8-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.8
This is a collection of mostly small fixes, mostly fixing fallout from
some of the DPCM changes that went in last time around which shook out
some issues on i.MX and Qualcomm platforms. The addition of a managed
version of snd_soc_register_dai() is to fix resource leaks.
There's also a few new device IDs for x86 systems.
Qiushi Wu [Sat, 13 Jun 2020 20:51:58 +0000 (15:51 -0500)]
ASoC: rockchip: Fix a reference count leak.
Calling pm_runtime_get_sync increments the counter even in case of
failure, causing incorrect ref count if pm_runtime_put is not called in
error handling paths. Call pm_runtime_put if pm_runtime_get_sync fails.
Fixes: fc05a5b22253 ("ASoC: rockchip: add support for pdm controller") Signed-off-by: Qiushi Wu <wu000273@umn.edu> Reviewed-by: Heiko Stuebner <heiko@sntech.de> Link: https://lore.kernel.org/r/20200613205158.27296-1-wu000273@umn.edu Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Intel: SOF: merge COMETLAKE_LP and COMETLAKE_H
We already have two configurations for CometLake, and a third one
coming. On other platforms, we used a single Kconfig option, so we
should follow the same trend by merging the two cases in a backwards
compatible way.
The backwards compatibility is handled by overloading the COMETLAKE_LP
kconfig as COMETLAKE. In practice we've never seen a case where
COMETLAKE_H is not selected along with COMETLAKE_LP, so keeping one
of the two is enough.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200617164755.18104-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Tue, 16 Jun 2020 12:09:21 +0000 (14:09 +0200)]
ALSA: usb-audio: Fix potential use-after-free of streams
With the recent full-duplex support of implicit feedback streams, an
endpoint can be still running after closing the capture stream as long
as the playback stream with the sync-endpoint is running. In such a
state, the URBs are still be handled and they may call retire_data_urb
callback, which tries to transfer the data from the PCM buffer. Since
the PCM stream gets closed, this may lead to use-after-free.
This patch adds the proper clearance of the callback at stopping the
capture stream for addressing the possible UAF above.
Shengjiu Wang [Tue, 16 Jun 2020 02:53:48 +0000 (10:53 +0800)]
ASoC: fsl_ssi: Fix bclk calculation for mono channel
For mono channel, SSI will switch to Normal mode.
In Normal mode and Network mode, the Word Length Control bits
control the word length divider in clock generator, which is
different with I2S Master mode (the word length is fixed to
32bit), it should be the value of params_width(hw_params).
The condition "slots == 2" is not good for I2S Master mode,
because for Network mode and Normal mode, the slots can also
be 2. Then we need to use (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK)
to check if it is I2S Master mode.
So we refine the formula for mono channel, otherwise there
will be sound issue for S24_LE.
Mark Brown [Mon, 15 Jun 2020 14:18:35 +0000 (15:18 +0100)]
Merge series "ASoC: topology: fix use-after-free when removing components" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
This patchset fixes a memory allocation issue and removes a 100%
reproducible use-after-free report thrown by KASAN in automated module
removal tests across multiple platforms.
All the credit goes to Bard Liao for root-causing the issue. DAIs may
be registered at the same time as a component, or when the topology is
loaded. This two-step registration causes the memory for
topology-based DAIs to allocated last, and conversely to be released
first by devres, before the component is released and the DAIs removed
from the component DAI list with snd_soc_unregister_dais().
When we remove a component, by the time we walk through its dai list
to unregister all dais, the dais allocated by the topology have been
freed already by devres and the list is corrupted with pointers that
are no longer valid.
The suggestion is to add an explicit devm_ based registration for
topology-based dais, so that each dai is cleanly removed from the
component dai list in the release operation before devres releases the
allocated memory.
Brent Lu [Fri, 12 Jun 2020 10:50:48 +0000 (18:50 +0800)]
ASoC: SOF: Intel: hda: Clear RIRB status before reading WP
Port commit 6d011d5057ff ("ALSA: hda: Clear RIRB status before reading
WP") from legacy HDA driver to fix the get response timeout issue.
Current SOF driver does not suffer from this issue because sync write
is enabled in hda_init. The issue will come back if the sync write is
disabled for some reason.
Signed-off-by: Brent Lu <brent.lu@intel.com> Reviewed-by: Takashi Iwai <tiwai@suse.de> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/1591959048-15813-1-git-send-email-brent.lu@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: qcom: common: set correct directions for dailinks
Currently both FE and BE dai-links are configured bi-directional,
However the DSP BE dais are only single directional,
so set the directions as supported by the BE dais.
Fixes: c25e295cd77b (ASoC: qcom: Add support to parse common audio device nodes) Reported-by: John Stultz <john.stultz@linaro.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Tested-by: John Stultz <john.stultz@linaro.org> Reviewed-by: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20200612123711.29130-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to q6afe_is_rx_port() to get direction
of DSP BE dai port, this is useful for setting dailink
directions correctly.
Fixes: c25e295cd77b (ASoC: qcom: Add support to parse common audio device nodes) Reported-by: John Stultz <john.stultz@linaro.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-by: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20200612123711.29130-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: soc-pcm: fix checks for multi-cpu FE dailinks
soc_dpcm_fe_runtime_update() is called for all dailinks, and we want
to first discard all back-ends, then deal with front-ends.
The existing code first reports an error with multi-cpu front-ends,
and that check needs to be moved after we know that we are dealing
with a front-end.
derek.fang [Fri, 12 Jun 2020 05:15:25 +0000 (13:15 +0800)]
ASoC: rt5682: Let dai clks be registered whether mclk exists or not
According to ideal rt5682 CCF, the root clk is mclk.
But in some platforms, mclk is not exported to CCF.
In this condition, rt5682_register_dai_clks will not be called.
This patch lets dai clks could be registered whether mclk exists or not.
The registration of DAIs may be done at two distinct times, once
during a component registration and later when loading a
topology. Since devm_ managed resources are freed in the reverse order
they were allocated, when a component starts unregistering DAIs by
walking through the DAI list, the memory allocated for the
topology-registered DAIs was freed already, which leads to 100%
reproducible KASAN use-after-free reports.
This patch suggests a new devm_ function to force the DAI list to be
updated prior to freeing the memory chunks referenced by the list
pointers.
Yick W. Tse [Sat, 13 Jun 2020 11:40:06 +0000 (11:40 +0000)]
ALSA: usb-audio: add quirk for Denon DCD-1500RE
fix error "clock source 41 is not valid, cannot use"
[] New USB device found, idVendor=154e, idProduct=1002, bcdDevice= 1.00
[] New USB device strings: Mfr=1, Product=2, SerialNumber=0
[] Product: DCD-1500RE
[] Manufacturer: D & M Holdings Inc.
[]
[] clock source 41 is not valid, cannot use
[] usbcore: registered new interface driver snd-usb-audio
Shengjiu Wang [Fri, 12 Jun 2020 07:37:51 +0000 (15:37 +0800)]
ASoC: fsl_asrc_dma: Fix data copying speed issue with EDMA
With EDMA, there is two dma channels can be used for dev_to_dev,
one is from ASRC, one is from another peripheral (ESAI or SAI).
If we select the dma channel of ASRC, there is an issue for ideal
ratio case, the speed of copy data is faster than sample
frequency, because ASRC output data is very fast in ideal ratio
mode.
So it is reasonable to use the dma channel of Back-End peripheral.
then copying speed of DMA is controlled by data consumption
speed in the peripheral FIFO,
Shengjiu Wang [Fri, 12 Jun 2020 07:37:50 +0000 (15:37 +0800)]
ASoC: fsl_asrc_dma: Reuse the dma channel if available in Back-End
The dma channel has been requested by Back-End cpu dai driver already.
If fsl_asrc_dma requests dma chan with same dma:tx symlink, then
there will be below warning with SDMA.
So if we can reuse the dma channel of Back-End, then the issue can be
fixed.
In order to get the dma channel which is already requested in Back-End.
we use the exported two functions (snd_soc_lookup_component_nolocked
and soc_component_to_pcm). If we can get the dma channel, then reuse it,
if can't, then request a new one.
snd_soc_lookup_component_nolocked can be used for the DPCM case
that Front-End needs to get the unused platform component but
added by Back-End cpu dai driver.
If the component is gotten, then we can get the dma chan created
by Back-End component and reused it in Front-End.
Successful send of EOS command does not indicate that EOS is actually
finished, correct event to wait EOS is finished is EOS_RENDERED event.
EOS_RENDERED means that the DSP has finished processing all the buffers
for that particular session and stream.
Mark Brown [Tue, 9 Jun 2020 14:46:20 +0000 (15:46 +0100)]
Merge series "ASoC: Fix dailink checks for DPCM" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
We've had a couple of changes that introduce regressions with the
multi-cpu DAI solutions, and while trying to fix them we found
additional inconsistencies that should also go to stable branches.
Bard Liao (1):
ASoC: core: only convert non DPCM link to DPCM link
Hans de Goede [Mon, 8 Jun 2020 20:46:34 +0000 (22:46 +0200)]
ASoC: rt5645: Add platform-data for Asus T101HA
The Asus T101HA uses the default jack-detect mode 3, but instead of
using an analog microphone it is using a DMIC on dmic-data-pin 1,
like the Asus T100HA. Note unlike the T100HA its jack-detect is not
inverted.
Add a DMI quirk with the correct settings for this model.
Hans de Goede [Mon, 8 Jun 2020 20:46:33 +0000 (22:46 +0200)]
ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT10-A tablet
The Toshiba Encore WT10-A tablet almost fully works with the default
settings for Bay Trail CR devices. The only issue is that it uses a
digital mic. connected the the DMIC1 input instead of an analog mic.
Add a quirk for this model using the default settings with the input-map
replaced with BYT_RT5640_DMIC1_MAP.
ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags
With additional checks on dailinks, we see errors such as
[ 3.000418] sof-nocodec sof-nocodec: CPU DAI DMIC01 Pin for rtd
NoCodec-6 does not support playback
It's not clear why we set the dpcm_playback and dpcm_capture flags
unconditionally, add a check on number of channels for each direction
to avoid invalid configurations.
ASoC: Intel: boards: replace capture_only by dpcm_capture
It's not clear why specific FE dailinks use capture_only flags, likely
blind copy/paste from Chromebook driver to the other. Replace by
dpcm_capture, this will make future alignment and removal of flags
easier.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com> Link: https://lore.kernel.org/r/20200608194415.4663-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Bard Liao [Mon, 8 Jun 2020 19:44:13 +0000 (14:44 -0500)]
ASoC: core: only convert non DPCM link to DPCM link
Additional checks for valid DAIs expose a corner case, where existing
BE dailinks get modified, e.g. HDMI links are tagged with
dpcm_capture=1 even if the DAIs are for playback.
This patch makes those changes conditional and flags configuration
issues when a BE dailink is has no_pcm=0 but dpcm_playback or
dpcm_capture=1 (which makes no sense).
As discussed on the alsa-devel mailing list, there are redundant flags
for dpcm_playback, dpcm_capture, playback_only, capture_only. This
will have to be cleaned-up in a future update. For now only correct
and flag problematic configurations.
Fixes: 218fe9b7ec7f3 ("ASoC: soc-core: Set dpcm_playback / dpcm_capture") Suggested-by: Daniel Baluta <daniel.baluta@nxp.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com> Link: https://lore.kernel.org/r/20200608194415.4663-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Michał Mirosław [Mon, 8 Jun 2020 16:50:39 +0000 (18:50 +0200)]
ALSA: pcm: disallow linking stream to itself
Prevent SNDRV_PCM_IOCTL_LINK linking stream to itself - the code
can't handle it. Fixed commit is not where bug was introduced, but
changes the context significantly.
Takashi Iwai [Fri, 5 Jun 2020 06:41:17 +0000 (08:41 +0200)]
ALSA: usb-audio: Manage auto-pm of all bundled interfaces
Currently USB-audio driver manages the auto-pm of the primary
interface although a card may consist of multiple interfaces.
This may leave the secondary and other interfaces left running
unnecessarily after the auto-suspend.
This patch allows the driver managing the auto-pm of all bundled
interfaces per card. The chip->pm_intf field is extended as
chip->intf[] to contain the array of assigned interfaces, and the
runtime-PM is performed to all those interfaces.
Hui Wang [Mon, 8 Jun 2020 11:55:41 +0000 (19:55 +0800)]
ALSA: hda/realtek - add a pintbl quirk for several Lenovo machines
A couple of Lenovo ThinkCentre machines all have 2 front mics and they
use the same codec alc623 and have the same pin config, so add a
pintbl entry for those machines to apply the fixup
ALC283_FIXUP_HEADSET_MIC.
Michał Mirosław [Mon, 8 Jun 2020 10:06:32 +0000 (12:06 +0200)]
ALSA: pcm: fix snd_pcm_link() lockdep splat
Add and use snd_pcm_stream_lock_nested() in snd_pcm_link/unlink
implementation. The code is fine, but generates a lockdep complaint:
============================================
WARNING: possible recursive locking detected
5.7.1mq+ #381 Tainted: G O
--------------------------------------------
pulseaudio/4180 is trying to acquire lock: ffff888402d6f508 (&group->lock){-...}-{2:2}, at: snd_pcm_common_ioctl+0xda8/0xee0 [snd_pcm]
but task is already holding lock: ffff8883f7a8cf18 (&group->lock){-...}-{2:2}, at: snd_pcm_common_ioctl+0xe4e/0xee0 [snd_pcm]
other info that might help us debug this:
Possible unsafe locking scenario:
CPU0
----
lock(&group->lock);
lock(&group->lock);
*** DEADLOCK ***
May be due to missing lock nesting notation
2 locks held by pulseaudio/4180:
#0: ffffffffa1a05190 (snd_pcm_link_rwsem){++++}-{3:3}, at: snd_pcm_common_ioctl+0xca0/0xee0 [snd_pcm]
#1: ffff8883f7a8cf18 (&group->lock){-...}-{2:2}, at: snd_pcm_common_ioctl+0xe4e/0xee0 [snd_pcm]
[...]
Kai-Heng Feng [Mon, 8 Jun 2020 06:26:28 +0000 (14:26 +0800)]
ALSA: usb-audio: Add vendor, product and profile name for HP Thunderbolt Dock
The HP Thunderbolt Dock has two separate USB devices, one is for speaker
and one is for headset. Add names for them so userspace can apply UCM
settings.
Colin Ian King [Thu, 4 Jun 2020 17:12:16 +0000 (18:12 +0100)]
ASoC: meson: fix memory leak of links if allocation of ldata fails
Currently if the allocation of ldata fails the error return path
does not kfree the allocated links object. Fix this by adding
an error exit return path that performs the necessary kfree'ing.
Takashi Iwai [Wed, 3 Jun 2020 15:37:08 +0000 (17:37 +0200)]
ALSA: usb-audio: Fix inconsistent card PM state after resume
When a USB-audio interface gets runtime-suspended via auto-pm feature,
the driver suspends all functionality and increment
chip->num_suspended_intf. Later on, when the system gets suspended to
S3, the driver increments chip->num_suspended_intf again, skips the
device changes, and sets the card power state to
SNDRV_CTL_POWER_D3hot. In return, when the system gets resumed from
S3, the resume callback decrements chip->num_suspended_intf. Since
this refcount is still not zero (it's been runtime-suspended), the
whole resume is skipped. But there is a small pitfall here.
The problem is that the driver doesn't restore the card power state
after this resume call, leaving it as SNDRV_CTL_POWER_D3hot. So,
even after the system resume finishes, the card instance still appears
as if it were system-suspended, and this confuses many ioctl accesses
that are blocked unexpectedly.
In details, we have two issues behind the scene: one is that the card
power state is changed only when the refcount becomes zero, and
another is that the prior auto-suspend check is kept in a boolean
flag. Although the latter problem is almost negligible since the
auto-pm feature is imposed only on the primary interface, but this can
be a potential problem on the devices with multiple interfaces.
This patch addresses those issues by the following:
- Replace chip->autosuspended boolean flag with chip->system_suspend
counter
- At the first system-suspend, chip->num_suspended_intf is recorded to
chip->system_suspend
- At system-resume, the card power state is restored when the
chip->num_suspended_intf refcount reaches to chip->system_suspend,
i.e. the state returns to the auto-suspended
Also, the patch fixes yet another hidden problem by the code
refactoring along with the fixes above: namely, when some resume
procedure failed, the driver left chip->num_suspended_intf that was
already decreased, and it might lead to the refcount unbalance.
In the new code, the refcount decrement is done after the whole resume
procedure, and the problem is avoided as well.
Fixes: 0662292aec05 ("ALSA: usb-audio: Handle normal and auto-suspend equally") Reported-and-tested-by: Macpaul Lin <macpaul.lin@mediatek.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200603153709.6293-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
Steve Lee [Thu, 4 Jun 2020 05:47:31 +0000 (14:47 +0900)]
ASoC: max98390: Fix potential crash during param fw loading
malformed firmware file can cause out-of-bound access and crash
during dsm_param bin loading.
- add MIN/MAX param size to avoid out-of-bound access.
- read start addr and size of param and check bound.
- add condition that fw->size > param_size + _PAYLOAD_OFFSET
to confirm enough data.
Takashi Iwai [Tue, 2 Jun 2020 16:44:53 +0000 (18:44 +0200)]
ASoC: max98390: Fix incorrect printf qualifier
This patch addresses a compile warning:
sound/soc/codecs/max98390.c:781:3: warning: format ‘%ld’ expects argument of type ‘long int’, but argument 4 has type ‘size_t {aka const unsigned int}’ [-Wformat=]
Fixes: a6e3f4f34cdb ("ASoC: max98390: Added Amplifier Driver") Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20200602164453.29925-1-tiwai@suse.de Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA: usb-audio: Add Pioneer DJ DJM-900NXS2 support
Pioneer DJ DJM-900NXS2 is a widely used DJ mixer with 2 audio USB
interfaces. Both have a MIDI controller, 10 playback and 12 capture
channels. Audio endpoints are vendor-specific and 3 files need to be
patched. All playback and capture channels work fine with all supported
sample rates (44.1k, 48k, 96k). Patches are attached.
Takashi Iwai [Mon, 1 Jun 2020 18:26:07 +0000 (20:26 +0200)]
Merge tag 'asoc-v5.8' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.8
This has been another very active release with a bunch of new drivers,
lots of fixes everywhere and continued core improvements from
Morimoto-san:
- Lots of core cleanups and refactorings from Morimoto-san, factoring
out common operations and making the card abstraction more solid.
- Continued work on cleaning up and improving the Intel drivers, along
with some new platform support for them.
- Fixes to make the Marvell SSPA driver work upstream.
- Support for AMD Renoir ACP, Dialog DA7212, Freescale EASRC and
i.MX8M, Intel Elkhard Lake, Maxim MAX98390, Nuvoton NAU8812 and
NAU8814 and Realtek RT1016.
Card related function should be implemented at soc-card now.
This patch adds it.
card has "card->probe" and "card->late_probe" callbacks,
and "late_probe" callback is called after "probe".
This means, we can set "card->probed" flag afer "late_probe"
for all cases.
Card related function should be implemented at soc-card now.
This patch adds it.
One note here is that card has "card->probe" and "card->late_probe"
callbacks.
Because it needs to care "late_probe", "card->probed" flag is set
under if (card->probe) at snd_soc_card_probe().
ASoC: soc-card: add probed bit field to snd_soc_card
We already have bit field to control snd_soc_card.
Let's add "probed" field on it instead of local variable.
One note here is that soc_cleanup_card_resources()
will be called as (A) formal cleanup or as (B) error handling,
thus, it needs to distinguish these.
In (A) case, card will have "instantiated" flag if all probe
callback functions were called without error.
Thus, snd_soc_unbind_card() is using it to judging card was probed.
But this this patch removes it, because it is no longer needed.
Mark Brown [Fri, 29 May 2020 20:43:38 +0000 (21:43 +0100)]
Merge series "Kconfig updates for DMIC and SOF HDMI support" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
This series provides the following updates to the Intel machine driver
Kconfig:
1. The first patch adds the explicit dependency on GPIOLIB when
SND_SOC_DMIC is selected.
2. SND_SOC_SOF_HDA_AUDIO_CODEC is required for using the legacy
HDA codec driver for HDMI support in SOF. The last 3 three patches
make the required changes to account for this.
changes since v1:
first patch for DMIC was merged already
rebase for HDMI on top of Arnd's RT5682 changes.
Libin Yang (3):
ASoC: intel: add depends on SND_SOC_SOF_HDA_AUDIO_CODEC for common
hdmi
ASoC: sof-sdw: remove CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC condition
ASoC: sof_pcm512x: remove CONFIG_SND_HDA_CODEC_HDMI condition
As CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC is always enabled in sof-soundwire
driver, let's remove the test of CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC
in the code.
Signed-off-by: Libin Yang <libin.yang@linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200529193547.6077-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
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