commit 9cb40eb184 upstream.
We met another Acer Aspire laptop which has the problem on the
headset-mic, the Pin 0x19 is not set the corret configuration for a
mic and the pin presence can't be detected too after plugging a
headset. Kailang suggested that we should set the coeff to enable the
mic and apply the ALC269_FIXUP_LIFEBOOK_EXTMIC. After doing that,
both headset-mic presence and headset-mic work well.
The existing ALC255_FIXUP_ACER_MIC_NO_PRESENCE set the headset-mic
jack to be a phantom jack. Now since the jack can support presence
unsol event, let us imporve it to set the jack to be a normal jack.
https://bugs.launchpad.net/bugs/1821269
Fixes: 5824ce8de7 ("ALSA: hda/realtek - Add support for Acer Aspire E5-475 headset mic")
Cc: Chris Chiu <chiu@endlessm.com>
CC: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 317d931392 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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 0b074ab7fc 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: 7f84ff68be ("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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 8ca5104715 ]
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: 1cc0c054f3 ("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: Sasha Levin <sashal@kernel.org>
[ Upstream commit b820d52e7e ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit ddb351145a ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit ea751227c8 ]
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: ff40260f79 ("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: Sasha Levin <sashal@kernel.org>
[ Upstream commit 30180e8436 ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit 16ec5dfe03 ]
On kbl_rt5663_max98927, commit 38a5882e42
("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: Sasha Levin <sashal@kernel.org>
commit 56df90b631 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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit dad3197da7 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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 891afcf246 upstream.
A mistake was made in the identification of the four variants of the
System76 Gazelle (gaze14). This patch corrects the PCI ID of the
17-inch, GTX 1660 Ti variant from 0x8560 to 0x8551. This patch also
adds the correct fixups for the 15-inch and 17-inch GTX 1650 variants
with PCI IDs 0x8560 and 0x8561.
Tests were done on all four variants ensuring full audio capability.
Fixes: 80a5052db7 ("ALSA: hdea/realtek - Headset fixup for System76 Gazelle (gaze14)")
Signed-off-by: Jeremy Soller <jeremy@system76.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 01c8327667 upstream.
In resume from S3, HDAC HDMI codec driver dapm event callback may be
operated before HDMI codec driver turns on the display audio power
domain because of the contest between display driver and hdmi codec driver.
This patch adds the device_link between soc card device (consumer) and
hdmi codec device (supplier) to make sure the sequence is always correct.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a46eb52322 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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ecb2795c08 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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 80a5052db7 upstream.
On the System76 Gazelle (gaze14), there is a headset microphone input
attached to 0x1a that does not have a jack detect. In order to get it
working, the pin configuration needs to be set correctly, and the
ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied. This is
identical to the patch already applied for the System76 Darter Pro
(darp5).
Signed-off-by: Jeremy Soller <jeremy@system76.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 607ca3bd22 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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 7f641e26a6 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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 8c2e6728c2 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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit cb5173594d 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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 0efa3334d6 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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 47c4cc08cb ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit c85064435f ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit c63adb28f6 ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit f0f2338a9c ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit c47255b611 ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit 19441e35a4 ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit a2225a6d15 ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1c5b6a27e4 ]
If ASRC turns on, HW will use clk_dac as the reference clock
whether recording or playback.
Both of clk_dac and clk_adc should set proper clock while using ASRC.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2b13bee388 ]
After commit fbeec965b8 ("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: fbeec965b8 ("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: Sasha Levin <sashal@kernel.org>
[ Upstream commit 54d1cf78b0 ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit 844a4a362d ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit c899df3e9b ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit 570f18b6a8 ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit b846819297 ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2e95f984aa ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit 82ad759143 ]
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: Sasha Levin <sashal@kernel.org>
commit 43d147be57 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: 639e5eb3c7 ("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>
Cc: Nobuhiro Iwamatsu <nobuhiro1.iwamatsu@toshiba.co.jp>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 17d3069ccf 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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 639e5eb3c7 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: 61fc060c40 ("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: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit aee48a9ffa upstream.
Commit 37c7401e8c ("ASoC: Intel: bytcr_rt5651: Fix DMIC map
headsetmic mapping"), changed the headsetmic mapping from IN3P to IN2P,
this was based on the observation that all bytcr_rt5651 devices I have
access to (7 devices) where all using IN3P for the headsetmic. This was
an attempt to unifify / simplify the mapping, but it was wrong.
None of those devices was actually using a digital internal mic. Now I've
access to a Point of View TAB-P1006W-232 (v1.0) tabler, which does use a
DMIC and it does have its headsetmic connected to IN2P, showing that the
original mapping was correct, so this commit reverts the change changing
the mapping back to IN2P.
Fixes: 37c7401e8c ("ASoC: Intel: bytcr_rt5651: Fix DMIC map ... mapping")
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Hui Wang