soc-topology adds extra dai_link by using snd_soc_add_dai_link(),
and removes it by snd_soc_romove_dai_link().
This snd_soc_add/remove_dai_link() and/or its related
functions are unbalanced before, and now, these are balance-uped.
But, it finds the random operation issue, and it is reported by
Pierre-Louis.
When card was released, topology will call snd_soc_remove_dai_link()
via (A).
static void soc_cleanup_card_resources(struct snd_soc_card *card)
{
struct snd_soc_dai_link *link, *_link;
/* This should be called before snd_card_free() */
(A) soc_remove_link_components(card);
/* free the ALSA card at first; this syncs with pending operations */
if (card->snd_card) {
(B) snd_card_free(card->snd_card);
card->snd_card = NULL;
}
/* remove and free each DAI */
(X) soc_remove_link_dais(card);
for_each_card_links_safe(card, link, _link)
(C) snd_soc_remove_dai_link(card, link);
...
}
At (A), topology calls snd_soc_remove_dai_link().
Then topology rtd, and its related all data are freed.
Next, (B) is called, and then, pcm->private_free = soc_pcm_private_free()
is called.
static void soc_pcm_private_free(struct snd_pcm *pcm)
{
struct snd_soc_pcm_runtime *rtd = pcm->private_data;
/* need to sync the delayed work before releasing resources */
flush_delayed_work(&rtd->delayed_work);
snd_soc_pcm_component_free(rtd);
}
Here, it gets rtd via pcm->private_data.
But, topology related rtd are already freed at (A).
Normal sound card has no damage, becase it frees rtd at (C).
These are finalizing rtd related data.
Thus, these should be called when rtd was freed, not sound card
was freed. It is very natural and understandable.
In other words, pcm->private_free = soc_pcm_private_free()
is no longer needed.
Extra issue is that there is zero chance to call
soc_remove_dai() for topology related dai at (X).
Because (A) removes rtd connection from card too, and,
(X) is based on card connected rtd.
This means, (X) need to be called before (C) (= for normal sound)
and (A) (= for topology).
Now, I want to focus this patch which is the reason why
snd_card_free() = (B) is located there.
commit 4efda5f213
("ASoC: Fix use-after-free at card unregistration")
Original snd_card_free() was called last of this function.
But moved to top to avoid use-after-free issue.
The issue was happen at soc_pcm_free() which was pcm->private_free,
today it is updated/renamed to soc_pcm_private_free().
In other words, (B) need to be called before (C) (= for normal sound)
and (A) (= for topology), because it needs (not yet freed) rtd.
But, (A) need to be called before (B),
because it needs card->snd_card pointer.
If we call flush_delayed_work() and snd_soc_pcm_component_free()
(= same as soc_pcm_private_free()) when rtd was freed (= (C), (A)),
there is no reason to call snd_card_free() at top of this function.
It can be called end of this function, again.
But, in such case, it will likely break unbind again, as Takashi-san
reported. When unbind is performed in a busy state, the code may
release still-in-use resources.
At least we need to call snd_card_disconnect_sync() at the first place.
The final code will be...
static void soc_cleanup_card_resources(struct snd_soc_card *card)
{
struct snd_soc_dai_link *link, *_link;
if (card->snd_card)
(Z) snd_card_disconnect_sync(card->snd_card);
(X) soc_remove_link_dais(card);
(A) soc_remove_link_components(card);
for_each_card_links_safe(card, link, _link)
(C) snd_soc_remove_dai_link(card, link);
...
if (card->snd_card) {
(B) snd_card_free(card->snd_card);
card->snd_card = NULL;
}
}
To avoid release still-in-use resources,
call snd_card_disconnect_sync() at (Z).
(X) is needed for both non-topology and topology.
topology removes rtd via (A), and
non topology removes rtd via (C).
snd_card_free() is no longer related to use-after-free issue.
Thus, locating (B) is no problem.
Fixes: df95a16d2a ("ASoC: soc-core: fix RIP warning on card removal")
Fixes: bc7a9091e5 ("ASoC: soc-core: add soc_unbind_dai_link()")
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87o8xax88g.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch uses rtd instead of pcm at snd_soc_pcm_component_new/free()
parameter.
This is prepare for dai_link remove bug fix on topology.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pnhqx89j.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Expose mixer control API for reading and writing controls from the kernel.
This API can be used by ALSA kernel drivers with ADSP support to
read and write firmware-defined memory regions.
Signed-off-by: Li Xu <li.xu@cirrus.com>
Signed-off-by: David Rhodes <david.rhodes@cirrus.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1573847653-17094-2-git-send-email-david.rhodes@cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The tlv320aic31xx devices allow to adjust the output common-mode voltage
for best analog performance. The datasheet states that the common mode
voltage should be set to be <= AVDD/2.
This changes allows to configure the output common-mode voltage via a DT
property. If the property is absent the voltage is automatically chosen
as the highest voltage below/equal to AVDD/2.
Signed-off-by: Lucas Stach <l.stach@pengutronix.de>
Link: https://lore.kernel.org/r/20191118151207.28576-1-l.stach@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
The manifest information is different between CNL, CML and CFL platforms
hence we need to load different files.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191111222901.19892-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In case the RST line is connected to a GPIO line it needs to be pulled high
when the driver probes to be able to use the codec.
Add support also for cases when more than one codec is is controlled by the
same GPIO line by requesting the gpio with GPIOD_FLAGS_BIT_NONEXCLUSIVE.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20191113124734.27984-3-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver forgets to call pm_runtime_disable in remove and
probe failure.
Add the calls to fix it.
Signed-off-by: Chuhong Yuan <hslester96@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20191118073707.28298-1-hslester96@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This driver misses calls to pm_runtime_disable and regulator_bulk_disable
in remove and a call to free_irq in probe failure.
Add the calls to fix it.
Signed-off-by: Chuhong Yuan <hslester96@gmail.com>
Link: https://lore.kernel.org/r/20191118073633.28237-1-hslester96@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Qualcomm DSPs also support the flac decoder, so add support for FLAC
decoder and convert the snd_dec_flac params to qdsp format.
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20191115102705.649976-4-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Qualcomm DSPs expect flac config to be set for flac decoders, so add the
API to program the flac config to the DSP
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20191115102705.649976-3-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
dma_request_slave_channel_reason() is:
#define dma_request_slave_channel_reason(dev, name) \
dma_request_chan(dev, name)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20191113095445.3211-3-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
dma_request_slave_channel_reason() is:
#define dma_request_slave_channel_reason(dev, name) \
dma_request_chan(dev, name)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20191113095445.3211-2-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_cleanup_card_resources() will call card->remove(), but it should be
called if card->probe() or card->late_probe() are called.
snd_soc_bind_card() might be error before calling
card->probe() / card->late_probe().
In that time, card->remove() will be called.
This patch adds card_probed parameter to judge it.
Fixes: bfce78a559 ("ASoC: soc-core: tidyup soc_init_dai_link()")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/87o8xg4ltr.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.
snd_soc_bind_card() is calling snd_soc_dapm_init() for both
card and component.
Let's call paired snd_soc_dapm_shutdown() at paired
soc_cleanup_card_resources().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r22c4lub.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_bind_card() is calling many initialize functions
for each card / link / dai / aux etc, etc, etc...
When error happen, the message is indicated at snd_soc_bind_card(),
not at each functions.
But, only soc_probe_aux_devices() case is indicating error at functions,
not at snd_soc_bind_card().
It is not an issue, but unbalanced.
This patch moves error message to snd_soc_bind_card().
Also avoids deep-nested code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87lfsthkw9.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
having both soc_bind_card() and snd_soc_instantiate_card() is
very confusable. Let's merge these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87mud9hkwj.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
having both soc_remove_component() and soc_cleanup_component() is
very confusable. Let's merge these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o8xphkwt.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't need to separete snd_soc_remove_dai_link() and
soc_unbind_dai_link() anymore. Let's merge these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pni5hkx1.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't need to separete snd_soc_add_dai_link() and
soc_bind_dai_link() anymore. Let's merge these.
One note is that before this patch, it adds list (A)
eventhough if it had dai_link->ignore (1), or already bounded dai_link (2).
But I guess it is wrong. This patch also solve this issue.
/* BEFORE */
int soc_bind_dai_link(...)
{
...
(1) if (dai_link->ignore)
return 0;
(2) if (soc_is_dai_link_bound(...))
return 0;
...
}
int snd_soc_add_dai_link(...)
{
...
=> ret = soc_bind_dai_link(...);
=> if (ret < 0)
=> return ret;
(A) list_add_tail(&dai_link->list, &card->dai_link_list);
...
}
/* AFTER */
int snd_soc_add_dai_link(...)
{
...
(1) if (dai_link->ignore)
return 0;
(2) if (soc_is_dai_link_bound(...))
return 0;
...
(A) list_add_tail(&dai_link->list, &card->dai_link_list);
return 0;
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r22lhkx8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't need to separete snd_soc_unregister_dai() and
soc_del_dai() anymore. Let's merge these
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sgn1hkxg.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Because complex separeted "card pre-listed component" and
"topology added component" duplicated operation is now
becoming simple, we don't need to check already bound dai_link
which is not exist anymore.
This patch removes soc_is_dai_link_bound().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v9rxhkxw.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The volume and bytes kcontrols are currently not freeing their
memory on initialization failures. When an error occurs, all the
widgets loaded so far are unloaded via sof_widget_unload().
But this only happens for the widgets that got successfully loaded.
Fix that by kfree()-ing the allocated memory on load error.
Fixes: 311ce4fe76 ("ASoC: SOF: Add support for loading topologies")
Reviewed-by: Paul Olaru <paul.olaru@nxp.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Dragos Tarcatu <dragos_tarcatu@mentor.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191111222039.19651-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We should suspend audio to D3 by default, for the sake of power saving,
change the condition of D0I3 suspending here to that when there is
stream with suspend_ignored specified.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191111223343.19986-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to check if the DSP should be put in D0i3. This function
returns true if a stream has ignored the SUSPEND trigger to keep the
pipelines running in the DSP.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191111223343.19986-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add check before seeting d0_substate and return success if Audio DSP is
already in the target substate.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191111223343.19986-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Audio DSP state machine with comments. Note that the
'D0<-->runtime D0I3' part is not implemented yet.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191111223343.19986-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We have platforms such as CFL with no known I2S codec being used, and
the ACPI tables are currently empty, so fall-back to using the
firmware filename used in nocodec mode
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191111222901.19892-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Due to firmware manifest/signature differences, we have to use
different firmware names, so split CNL machine table in three (CNL,
CFL, CML).
The CFL table is currently empty since all known platforms use
HDaudio, but let's plan ahead.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191111222901.19892-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds a new mode WM8904_CLK_AUTO which automatically enables the FLL
if a frequency different than the MCLK is set.
These additions make the codec work with the simple-card driver in
general and especially in systems where the MCLK doesn't match the
required clock.
Signed-off-by: Michael Walle <michael@walle.cc>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20191108203152.19098-1-michael@walle.cc
Signed-off-by: Mark Brown <broonie@kernel.org>
On KBL platform, the microphone is attached to external codec(rt5514)
instead of PCH. However, TDM slot between PCH and codec is 16 bits only.
In order to avoid setting wrong format, we should add a constraint to
force to use 16 bits format forever.
Signed-off-by: Yu-Hsuan Hsu <yuhsuan@chromium.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190923162940.199580-1-yuhsuan@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit adds the Dialog DA7213 audio codec as a selectable option
in the kernel config. Currently the driver can only be selected for
Intel Baytrail/Cherrytrail devices or if SND_SOC_ALL_CODECS is enabled.
Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.com>
Link: https://lore.kernel.org/r/20191108174843.11227-3-sebastian.reichel@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The RT5677 DSP needs the I2S MCLK1 to run its DSP. Add a dapm route to
SSP0 CODEC IN so the clock is turned on automatically when the DSP is
turned on.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20191106011335.223061-10-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Due to limitations of the clocking configuration, we have no way of
scheduling our hibernation before the bdw dsp hibernates. This causes
issues when the system suspends with an open stream. We need userspace
to toggle the kcontrol before we are suspended so that any writes on
suspend are not lost and we don't corrupt the regmap.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20191106011335.223061-9-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The irq is disabled at suspend to avoid running the threaded irq
handler after the codec has been powered off. At resume, codec irq is
re-enabled and the interrupt status register is checked to see if
headphone has been pluggnd/unplugged while the device is suspended.
There is still a chance that the headphone gets enabled or disabled
after the codec is suspended. disable_irq syncs the threaded irq
handler, but soc-jack's threaded irq handler schedules a delayed
work to poll gpios (for debounce). This is still OK. The codec won't
be powered back on again because all audio paths have been suspended,
and there are no force enabled supply widgets (MICBIAS1 is disabled).
The gpio status read after codec power off could be wrong, so the
gpio values are checked again after resume.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20191106011335.223061-8-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
MCLK1 gets disabled at suspend and re-enabled at resume. Before
MCLK1 is re-enabled, if the DSP is already on (either the DSP was
left on during suspend, or the DSP is turned on early at resume),
i2c register read returns garbage and corrupts the regmap cache.
This patch stops the DSP before suspend and restarts it after
resume with a dalay to ensure MCLK is on while loading firmware.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20191106011335.223061-7-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The codec dies when RT5677_PWR_ANLG2(MX-64h) is set to 0xACE1
while it's streaming audio over SPI. The DSP firmware turns
on PLL2 (MX-64 bit 8) when SPI streaming starts. However regmap
does not believe that register can change by itself. When
BST1 (bit 15) is turned on with regmap_update_bits(), it doesn't
read the register first before write, so PLL2 power bit is
cleared by accident.
Marking MX-64h as volatile in regmap solved the issue.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20191106011335.223061-6-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a DAPM audio path from "DMIC L1" to "DSP Buffer" so that
when hotwording is enabled, DAPM does not power off the codec
with SND_SOC_BIAS_OFF.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20191106011335.223061-5-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Before a hotword is detected, GPIO1 pin is configured as IRQ
output so that jack detect works. When a hotword is detected,
the DSP firmware configures the GPIO1 pin as GPIO1 and
drives a 1. rt5677_irq() is called after a rising edge on
the GPIO1 pin, due to either jack detect event or hotword
event, or both. All possible events are checked and handled
in rt5677_irq() where GPIO1 pin is configured back to IRQ
output if a hotword is detected.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20191106011335.223061-4-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The firmware rt5677_elf_vad is an ELF binary obtained from
request_firmware(). Sections of the ELF are loaded to
the DSP via SPI. A model (e.g. en_us.mmap) can optionally be
loaded to the DSP at RT5677_MODEL_ADDR to overwrite the
baked-in model in rt5677_elf_vad.
Then we switch to DSP mode, load firmware, and let DSP run.
When a hotword is detected, an interrupt is fired and
rt5677_irq() is called. When 'DSP VAD Switch' is turned off,
the codec is set back to normal mode.
The kcontrol 'DSP VAD Switch' is automatically enabled/disabled
when the hotwording PCM stream is opened/closed.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20191106011335.223061-2-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Audmix support two substream, When two substream start
to run, the trigger function may be called by two substream
in same time, that the priv->tdms may be updated wrongly.
The expected priv->tdms is 0x3, but sometimes the
result is 0x2, or 0x1.
Fixes: be1df61cf0 ("ASoC: fsl: Add Audio Mixer CPU DAI driver")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/1e706afe53fdd1fbbbc79277c48a98f8416ba873.1573458378.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org>
Set L1SEN to make sure the system can enter S0ix, and restore it on
resume.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191101170916.26517-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add check to avoid possible NULL pointer dereference issue.
This issue was reported by static analysis tools, we didn't face this
issue but we can't rule it out either as a false positive.
Reported-by: Keqiao Zhang <keqiao.zhang@intel.com>
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191101170916.26517-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
