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commit fa763f1b2858752e6150ffff46886a1b7faffc82 upstream.
We observed the same issue as reported by commit a8d7bde23e7130686b7662
("ALSA: hda - Force polling mode on CFL for fixing codec communication")
We don't have a better solution. So apply the same workaround to CNL.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 2eabc5ec8ab4d4748a82050dfcb994119b983750 upstream.
The snd_seq_ioctl_get_subscription() retrieves the port subscriber
information as a pointer, while the object isn't protected, hence it
may be deleted before the actual reference. This race was spotted by
syzkaller and may lead to a UAF.
The fix is simply copying the data in the lookup function that
performs in the rwsem to protect against the deletion.
Reported-by: syzbot+9437020c82413d00222d@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit b06c58c2a1eed571ea2a6640fdb85b7b00196b1e upstream.
When the output sample rate is [8kHz, 30kHz], the limitation
of the supported ratio range is [1/24, 8]. In the driver
we use (8kHz, 30kHz) instead of [8kHz, 30kHz].
So this patch is to fix this issue and the potential rounding
issue with divider.
Fixes: fff6e03c7b65 ("ASoC: fsl_asrc: add support for 8-30kHz
output sample rate")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit ad6eecbfc01c987e0253371f274c3872042e4350 upstream.
Add regcache_mark_dirty before regcache_sync for power
of codec may be lost at suspend, then all the register
need to be reconfigured.
Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver
support for CS42448/CS42888")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 0e3fb6995bfabb23c172e8b883bf5ac57102678e upstream.
The data for isochronous resources is not destroyed in expected place.
This commit fixes the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 9b2bb4f2f4a2 ("ALSA: firewire-motu: add stream management functionality")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 717f43d81afc1250300479075952a0e36d74ded3 upstream.
ALC255 and ALC256 were some difference for hidden register.
This update was suitable for ALC256.
Fixes: e69e7e03ed22 ("ALSA: hda/realtek - ALC256 speaker noise issue")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit d8fa87c368f5b4096c4746894fdcc195da285df1 upstream.
Stanton SCS.1m can transfer isochronous packet with Multi Bit Linear
Audio data channels, therefore it allows software to capture PCM
substream. However, ALSA oxfw driver doesn't.
This commit changes the driver to add one PCM substream for capture
direction.
Fixes: de5126cc3c0b ("ALSA: oxfw: add stream format quirk for SCS.1 models")
Cc: <stable@vger.kernel.org> # v4.5+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 7c32ae35fbf9cffb7aa3736f44dec10c944ca18e upstream.
The call of unsubscribe_port() which manages the group count and
module refcount from delete_and_unsubscribe_port() looks racy; it's
not covered by the group list lock, and it's likely a cause of the
reported unbalance at port deletion. Let's move the call inside the
group list_mutex to plug the hole.
Reported-by: syzbot+e4c8abb920efa77bace9@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit f495222e28275222ab6fd93813bd3d462e16d340 upstream.
Currently the IRQ handler in HD-audio controller driver is registered
before the chip initialization. That is, we have some window opened
between the azx_acquire_irq() call and the CORB/RIRB setup. If an
interrupt is triggered in this small window, the IRQ handler may
access to the uninitialized RIRB buffer, which leads to a NULL
dereference Oops.
This is usually no big problem since most of Intel chips do register
the IRQ via MSI, and we've already fixed the order of the IRQ
enablement and the CORB/RIRB setup in the former commit b61749a89f82
("sound: enable interrupt after dma buffer initialization"), hence the
IRQ won't be triggered in that room. However, some platforms use a
shared IRQ, and this may allow the IRQ trigger by another source.
Another possibility is the kdump environment: a stale interrupt might
be present in there, the IRQ handler can be falsely triggered as well.
For covering this small race, let's move the azx_acquire_irq() call
after hda_intel_init_chip() call. Although this is a bit radical
change, it can cover more widely than checking the CORB/RIRB setup
locally in the callee side.
Reported-by: Liwei Song <liwei.song@windriver.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 317d9313925cd8388304286c0d3c8dda7f060a2d upstream.
I measured power consumption between power_save_node=1 and power_save_node=0.
It's almost the same.
Codec will enter to runtime suspend and suspend.
That pin also will enter to D3. Don't need to enter to D3 by single pin.
So, Disable power_save_node as default. It will avoid more issues.
Windows Driver also has not this option at runtime PM.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 0b074ab7fc0d575247b9cc9f93bb7e007ca38840 upstream.
The current code performs the cancel of a delayed work at the late
stage of disconnection procedure, which may lead to the access to the
already cleared state.
This patch assures to call cancel_delayed_work_sync() at the beginning
of the disconnection procedure for avoiding that race. The delayed
work object is now assigned in the common line6 object instead of its
derivative, so that we can call cancel_delayed_work_sync().
Along with the change, the startup function is called via the new
callback instead. This will make it easier to port other LINE6
drivers to use the delayed work for startup in later patches.
Reported-by: syzbot+5255458d5e0a2b10bbb9@syzkaller.appspotmail.com
Fixes: 7f84ff68be05 ("ALSA: line6: toneport: Fix broken usage of timer for delayed execution")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 8ca5104715cfd14254ea5aecc390ae583b707607 upstream.
Building with clang shows a variable that is only used by the
suspend/resume functions but defined outside of their #ifdef block:
sound/soc/ti/davinci-mcasp.c:48:12: error: variable 'context_regs' is not needed and will not be emitted
We commonly fix these by marking the PM functions as __maybe_unused,
but here that would grow the davinci_mcasp structure, so instead
add another #ifdef here.
Fixes: 1cc0c054f380 ("ASoC: davinci-mcasp: Convert the context save/restore to use array")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Nathan Chancellor <natechancellor@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit c705247136a523488eac806bd357c3e5d79a7acd upstream.
The call to of_parse_phandle returns a node pointer with refcount
incremented thus it must be explicitly decremented after the last
usage.
Detected by coccinelle with the following warnings:
./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function.
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linuxppc-dev@lists.ozlabs.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit b820d52e7eed7b30b2dfef5f4213a2bc3cbea6f3 upstream.
The call to of_parse_phandle returns a node pointer with refcount
incremented thus it must be explicitly decremented after the last
usage.
Detected by coccinelle with the following warnings:
./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function.
./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function.
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit ddb351145a967ee791a0fb0156852ec2fcb746ba upstream.
is_slave_mode defaults to false because sai structure
that contains it is kzalloc'ed.
Anyhow, if we decide to set the following configuration
SAI slave -> SAI master, is_slave_mode will remain set on true
although SAI being master it should be set to false.
Fix this by updating is_slave_mode for each call of
fsl_sai_set_dai_fmt.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit ea751227c813ab833609afecfeedaf0aa26f327e upstream.
During randconfig builds, I occasionally run into an invalid configuration
of the freescale FIQ sound support:
WARNING: unmet direct dependencies detected for SND_SOC_IMX_PCM_FIQ
Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m]
Selected by [y]:
- SND_SOC_FSL_SPDIF [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m]!=n && (MXC_TZIC [=n] || MXC_AVIC [=y])
sound/soc/fsl/imx-ssi.o: In function `imx_ssi_remove':
imx-ssi.c:(.text+0x28): undefined reference to `imx_pcm_fiq_exit'
sound/soc/fsl/imx-ssi.o: In function `imx_ssi_probe':
imx-ssi.c:(.text+0xa64): undefined reference to `imx_pcm_fiq_init'
The Kconfig warning is a result of the symbol being defined inside of
the "if SND_IMX_SOC" block, and is otherwise harmless. The link error
is more tricky and happens with SND_SOC_IMX_SSI=y, which may or may not
imply FIQ support. However, if SND_SOC_FSL_SSI is set to =m at the same
time, that selects SND_SOC_IMX_PCM_FIQ as a loadable module dependency,
which then causes a link failure from imx-ssi.
The solution here is to make SND_SOC_IMX_PCM_FIQ built-in whenever
one of its potential users is built-in.
Fixes: ff40260f79dc ("ASoC: fsl: refine DMA/FIQ dependencies")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 30180e8436046344b12813dc954b2e01dfdcd22d upstream.
If the hdmi codec startup fails, it should clear the current_substream
pointer to free the device. This is properly done for the audio_startup()
callback but for snd_pcm_hw_constraint_eld().
Make sure the pointer cleared if an error is reported.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 16ec5dfe0327ddcf279957bffe4c8fe527088c63 upstream.
On kbl_rt5663_max98927, commit 38a5882e4292
("ASoC: Intel: kbl_rt5663_max98927: Map BTN_0 to KEY_PLAYPAUSE")
This key pair mapping to play/pause when playing Youtube
The Android 3.5mm Headset jack specification mentions that BTN_0 should
be mapped to KEY_MEDIA, but this is less logical than KEY_PLAYPAUSE,
which has much broader userspace support.
For example, the Chrome OS userspace now supports KEY_PLAYPAUSE to toggle
play/pause of videos and audio, but does not handle KEY_MEDIA.
Furthermore, Android itself now supports KEY_PLAYPAUSE equivalently, as the
new USB headset spec requires KEY_PLAYPAUSE for BTN_0.
https://source.android.com/devices/accessories/headset/usb-headset-spec
The same fix is required on Chrome kbl_da7219_max98357a.
Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Reviewed-by: Benson Leung <bleung@chromium.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 56df90b631fc027fe28b70d41352d820797239bb upstream.
Add patch for realtek codec in Lenovo B50-70 that fixes inverted
internal microphone channel.
Device IdeaPad Y410P has the same PCI SSID as Lenovo B50-70,
but first one is about fix the noise and it didn't seem help in a
later kernel version.
So I replaced IdeaPad Y410P device description with B50-70 and apply
inverted microphone fix.
Bugzilla: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1524215
Signed-off-by: Michał Wadowski <wadosm@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit dad3197da7a3817f27bb24f7fd3c135ffa707202 upstream.
Dell platform with ALC298.
system enter to runtime suspend. Headphone had noise.
Let Headset Mic not shutup will solve this issue.
[ Fixed minor coding style issues by tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 903c220b1ece12f17c868e43f2243b8f81ff2d4c upstream.
case ESAI_HCKT_EXTAL and case ESAI_HCKR_EXTAL should be
independent of each other, so replace fall-through with break.
Fixes: 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 16bbeb2b43c3f5d69e1348477e75a24ae6d55d5a upstream.
In preparation to enabling -Wimplicit-fallthrough, mark switch cases
where we are expecting to fall through.
Addresses-Coverity-ID: 1222121 ("Missing break in switch")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit a46eb523220e242affb9a6bc9bb8efc05f4f7459 upstream.
The current algorithm allows 3 types of transfers, 16bit, 32bit and
burst. According to Realtek, 16bit transfers have a special restriction
in that it is restricted to the memory region of
0x18020000 ~ 0x18021000. This region is the memory location of the I2C
registers. The current algorithm does not uphold this restriction and
therefore fails to complete writes.
Since this has been broken for some time it likely no one is using it.
Better to simply disable the 16 bit writes. This will allow users to
properly load firmware over SPI without data corruption.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Reviewed-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit ecb2795c08bc825ebd604997e5be440b060c5b18 upstream.
The max98090 driver defines 3 DAPM muxes; one for the right line output
(LINMOD Mux), one for the left headphone mixer source (MIXHPLSEL Mux)
and one for the right headphone mixer source (MIXHPRSEL Mux). The same
bit is used for the mux as well as the DAPM enable, and although the mux
can be correctly configured, after playback has completed, the mux will
be reset during the disable phase. This is preventing the state of these
muxes from being saved and restored correctly on system reboot. Fix this
by marking these muxes as SND_SOC_NOPM.
Note this has been verified this on the Tegra124 Nyan Big which features
the MAX98090 codec.
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 607ca3bd220f4022e6f5356026b19dafc363863a upstream.
Let EAPD turn on after set pin output.
[ NOTE: This change is supposed to reduce the possible click noises at
(runtime) PM resume. The functionality should be same (i.e. the
verbs are executed correctly) no matter which order is, so this
should be safe to apply for all codecs -- tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 7f641e26a6df9269cb25dd7a4b0a91d6586ed441 upstream.
On the machines with AMD GPU or Nvidia GPU, we often meet this issue:
after s3, there are 4 HDMI/DP audio devices in the gnome-sound-setting
even there is no any monitors plugged.
When this problem happens, we check the /proc/asound/cardX/eld#N.M, we
will find the monitor_present=1, eld_valid=0.
The root cause is BIOS or GPU driver makes the PRESENCE valid even no
monitor plugged, and of course the driver will not get the valid
eld_data subsequently.
In this situation, we should not report the jack_plugged event, to do
so, let us change the function hdmi_present_sense_via_verbs(). In this
function, it reads the pin_sense via snd_hda_pin_sense(), after
calling this function, the jack_dirty is 0, and before exiting
via_verbs(), we change the shadow pin_sense according to both
monitor_present and eld_valid, then in the snd_hda_jack_report_sync(),
since the jack_dirty is still 0, it will report jack event according
to this modified shadow pin_sense.
After this change, the driver will not report Jack_is_plugged event
through hdmi_present_sense_via_verbs() if monitor_present is 1 and
eld_valid is 0.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 8c2e6728c2bf95765b724e07d0278ae97cd1ee0d upstream.
The driver will check the monitor presence when resuming from suspend,
starting poll or interrupt triggers. In these 3 situations, the
jack_dirty will be set to 1 first, then the hda_jack.c reads the
pin_sense from register, after reading the register, the jack_dirty
will be set to 0. But hdmi_repoll_work() is enabled in these 3
situations, It will read the pin_sense a couple of times subsequently,
since the jack_dirty is 0 now, It does not read the register anymore,
instead it uses the shadow pin_sense which is read at the first time.
It is meaningless to check the shadow pin_sense a couple of times,
we need to read the register to check the real plugging state, so
we set the jack_dirty to 1 in the hdmi_repoll_work().
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit cb5173594d50c72b7bfa14113dfc5084b4d2f726 upstream.
In parse_audio_selector_unit(), the string array 'namelist' is allocated
through kmalloc_array(), and each string pointer in this array, i.e.,
'namelist[]', is allocated through kmalloc() in the following for loop.
Then, a control instance 'kctl' is created by invoking snd_ctl_new1(). If
an error occurs during the creation process, the string array 'namelist',
including all string pointers in the array 'namelist[]', should be freed,
before the error code ENOMEM is returned. However, the current code does
not free 'namelist[]', resulting in memory leaks.
To fix the above issue, free all string pointers 'namelist[]' in a loop.
Signed-off-by: Wenwen Wang <wang6495@umn.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 7f84ff68be05ec7a5d2acf8fdc734fe5897af48f upstream.
The line6 toneport driver has code for some delayed initialization,
and this hits the kernel Oops because mutex and other sleepable
functions are used in the timer callback. Fix the abuse by a delayed
work instead so that everything works gracefully.
Reported-by: syzbot+a07d0142e74fdd595cfb@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 0efa3334d65b7f421ba12382dfa58f6ff5bf83c4 upstream.
Currently in sst_dsp_new() if we get an error return from sst_dma_new()
we just print an error message and then still complete the function
successfully. This means that we are trying to run without sst->dma
properly set up, which will result in NULL pointer dereference when
sst->dma is later used. This was happening for me in
sst_dsp_dma_get_channel():
struct sst_dma *dma = dsp->dma;
...
dma->ch = dma_request_channel(mask, dma_chan_filter, dsp);
This resulted in:
BUG: unable to handle kernel NULL pointer dereference at 0000000000000018
IP: sst_dsp_dma_get_channel+0x4f/0x125 [snd_soc_sst_firmware]
Fix this by adding proper error handling for the case where we fail to
set up DMA.
This change only affects Haswell and Broadwell systems. Baytrail
systems explicilty opt-out of DMA via sst->pdata->resindex_dma_base
being set to -1.
Signed-off-by: Ross Zwisler <zwisler@google.com>
Cc: stable@vger.kernel.org
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit d6ba3f815bc5f3c4249d15c8bc5fbb012651b4a4 upstream.
Fix wrong setting on number of channels. The context wants to set
constraint to 2 channels instead of 4.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 47c4cc08cb5b34e93ab337b924c5ede77ca3c936 upstream.
The chips main power supplies VA and VP are enabled during probe but
then never disabled, this will cause warnings from the regulator
framework on driver removal. Fix this by adding a remove callback and
disabling the supplies, whilst doing so follow best practice and put the
chip back into reset as well.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit c85064435fe7a216ec0f0238ef2b8f7cd850a450 upstream.
This is because set_fmt ops maybe called when PD is off,
and in such case, regmap_ops will lead system hang.
enale PD before doing regmap_ops.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit c63adb28f6d913310430f14c69f0a2ea55eed0cc upstream.
The common pins were mistakenly not added to the DAPM graph.
Adding these pins will allow valid graphs to be created.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit cacea3a90e211f0c111975535508d446a4a928d2 upstream.
w_text_param can be NULL and it is being dereferenced without checking.
Add the missing sanity check to prevent NULL pointer dereference.
Signed-off-by: Pankaj Bharadiya <pankaj.laxminarayan.bharadiya@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit f0f2338a9cfaf71db895fa989ea7234e8a9b471d upstream.
The CS4270 does not by default increment the register address on
consecutive writes. During normal operation it doesn't matter as all
register accesses are done individually. At resume time after suspend,
however, the regcache code gathers the biggest possible block of
registers to sync and sends them one on one go.
To fix this, set the INCR bit in all cases.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit c47255b61129857b74b0d86eaf59335348be05e0 upstream.
Register platform component with a prefix, to avoid warnings
on debugfs entries creation, due to component name
redundancy.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 19441e35a43b616ea6afad91ed0d9e77268d8f6a upstream.
The DFSDM must be stopped when a new setting is applied.
restart systematically DFSDM on multiple prepare calls,
to apply changes.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit a2225a6d155fcb247fe4c6d87f7c91807462966d upstream.
Best to lock across handling the bus error to ensure the DSP doesn't
change power state as we are reading the status registers.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 2b13bee3884926cba22061efa75bd315e871de24 upstream.
After commit fbeec965b8d1c ("ASoC: samsung: odroid: Fix 32000 sample rate
handling") the audio root clock frequency is configured improperly for
44100 sample rate. Due to clock rate rounding it's 20070401 Hz instead
of 22579000 Hz. This results in a too low value of the PSR clock divider
in the CPU DAI driver and too fast actual sample rate for fs=44100. E.g.
1 kHz tone has actual 1780 Hz frequency (1 kHz * 20070401/22579000 * 2).
Fix this by increasing the correction passed to clk_set_rate() to take
into account inaccuracy of the EPLL frequency properly.
Fixes: fbeec965b8d1c ("ASoC: samsung: odroid: Fix 32000 sample rate handling")
Reported-by: JaeChul Lee <jcsing.lee@samsung.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 54d1cf78b0f4ba348a7c7fb8b7d0708d71b6cc8a upstream.
The driver changes the stream name of DAC and ADC to avoid the issue of
widget with prefixed name. When the machine adds prefixed name for codec,
the stream name of DAI may not find the widgets.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 844a4a362dbec166b44d6b9b3dd45b08cb273703 upstream.
The driver has two issues when machine add prefix name for codec.
(1)The stream name of DAI can't find the AIF widgets.
(2)The drivr can enable/disalbe the MICBIAS and SAR widgets.
The patch will fix these issues caused by prefixed name added.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit c899df3e9b0bf7b76e642aed1a214582ea7012d5 upstream.
If playback and capture are enabled concurrently, when the capture stops
the output becomes inaudile. The playback application will become stuck
and underrun after a timeout.
This is caused by mistaken use of the stream_id, which should only be
set for playback and not for capture
Tested on Apollolake and Kabylake with SST driver.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 570f18b6a8d1f0e60e8caf30e66161b6438dcc91 upstream.
On HDaudio platforms, if playback is started when capture is working,
there is no audible output.
This can be root-caused to the use of the rx|tx_mask to store an HDaudio
stream tag.
If capture is stared before playback, rx_mask would be non-zero on HDaudio
platform, then the channel number of playback, which is in the same codec
dai with the capture, would be changed by soc_pcm_codec_params_fixup based
on the tx_mask at first, then overwritten by this function based on rx_mask
at last.
According to the author of tx|rx_mask, tx_mask is for playback and rx_mask
is for capture. And stream direction is checked at all other references of
tx|rx_mask in ASoC, so here should be an error. This patch checks stream
direction for tx|rx_mask for fixup function.
This issue would affect not only HDaudio+ASoC, but also I2S codecs if the
channel number based on rx_mask is not equal to the one for tx_mask. It could
be rarely reproduecd because most drivers in kernel set the same channel number
to tx|rx_mask or rx_mask is zero.
Tested on all platforms using stream_tag & HDaudio and intel I2S platforms.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit b8468192971807c43a80d6e2c41f83141cb7b211 upstream.
Change capabilities exposed in SAI S/PDIF mode, to match
actually supported formats.
In S/PDIF mode only 32 bits stereo is supported.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 2e95f984aae4cf0608d0ba2189c756f2bd50b44a upstream.
When using the S/PDIF DAI, there is no requirement to call
snd_soc_dai_set_fmt() as there is no DAI format definition that defines
S/PDIF. In any case, S/PDIF does not have separate clocks, this is
embedded into the data stream.
Consequently, when attempting to use TDA998x in S/PDIF mode, the attempt
to configure TDA998x via the hw_params callback fails as the
hdmi_codec_daifmt is left initialised to zero.
Since the S/PDIF DAI will only be used by S/PDIF, prepare the
hdmi_codec_daifmt structure for this format.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 82ad759143ed77673db0d93d53c1cde7b99917ee upstream.
This patch fixes a bug that prevents freeing the reset gpio on unloading
the module.
aic3x_i2c_probe is called when loading the module and it calls list_add
with a probably uninitialized list entry aic3x->list (next = prev = NULL)).
So even if list_del is called it does nothing and in the end the gpio_reset
is not freed. Then a repeated module probing fails silently because
gpio_request fails.
When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move
list_del to aic3x_i2c_remove because aic3x_remove may be called
multiple times without aic3x_i2c_remove being called which leads to
a NULL pointer dereference.
Signed-off-by: Philipp Puschmann <philipp.puschmann@emlix.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 43d147be5738a9ed6cfb25c285ac50d6dd5793be upstream.
Trigger stop can be called in situations where trigger start failed
and as such it can't be assumed the buffer is already attached to
the compressed stream or a NULL pointer may be dereferenced.
Fixes: 639e5eb3c7d6 ("ASoC: wm_adsp: Correct handling of compressed streams that restart")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 17d3069ccf06970e2db3f7cbf4335f207524279e upstream.
This patch fixes the sai driver structure overwriting which results in
a cpu dai name equal NULL.
Fixes: 3e086ed ("ASoC: stm32: add SAI driver")
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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commit 639e5eb3c7d67e407f2a71fccd95323751398f6f upstream.
Previously support was added to allow streams to be stopped and
started again without the DSP being power cycled and this was done
by clearing the buffer state in trigger start. Another supported
use-case is using the DSP for a trigger event then opening the
compressed stream later to receive the audio, unfortunately clearing
the buffer state in trigger start destroys the data received
from such a trigger. Correct this issue by moving the call to
wm_adsp_buffer_clear to be in trigger stop instead.
Fixes: 61fc060c40e6 ("ASoC: wm_adsp: Support streams which can start/stop with DSP active")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
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Bard Liao
Takashi Iwai
S.j. Wang
Takashi Sakamoto
Kailang Yang
Arnd Bergmann
Wen Yang
Daniel Baluta
Jerome Brunet
Mac Chiang
Michał Wadowski
Gustavo A. R. Silva
Curtis Malainey
Jon Hunter
Hui Wang
Wenwen Wang
Ross Zwisler
Tzung-Bi Shih
Charles Keepax
Sugar Zhang
Annaliese McDermond
Pankaj Bharadiya
Daniel Mack
Olivier Moysan
Sylwester Nawrocki
John Hsu
Rander Wang
Russell King
Philipp Puschmann
Arnaud Pouliquen