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>
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>
The jack type detection needs the main bias power of analog.
The modification makes sure the main bias power on/off while jack plug/unplug.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The IRQ function may not work when system suspend.
We remove snd_soc_dapm_force_enable_pin function call to
make sure the bias off when idle and run into suspend/resume function.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Skip for i2s5 in mck_disable which is also bypassed in mck_enable.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
compiler complains about following declarations
sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsdsr_table_pattern1[] = {
^~~~~
sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsdsr_table_pattern2[] = {
^~~~~
sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsisr_table[] = {
^~~~~
sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan288888[] = {
^~~~~
sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan244888[] = {
^~~~~
sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan222222[] = {
^~~~~
This patch moves the 'static' keyword to the front of the
declaration to fix the compiler warnings
Fixes: linux-next commit 7674bec4fc ("ASoC: rsnd: update BSDSR/BSDISR handling")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
lockdep warns us that priv->lock and k->k_lock can cause a
deadlock when after acquire of k->k_lock, process is interrupted
by src, while in another routine of src .init, k->k_lock is
acquired with priv->lock held.
This patch avoids a potential deadlock by not calling soc_device_match()
in SRC .init callback, instead it adds new soc fields in priv->flags to
differentiate SoCs.
Fixes: linux-next commit 7674bec4fc ("ASoC: rsnd: update BSDSR/BSDISR handling")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
- Declare SR as volatile, as it is changed by hardware.
- Remove TXDR from readable and volatile register list,
as it is intended for write accesses only.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
The dpcm get from fe_clients/be_clients
may be free before use
Add a spin lock at snd_soc_card level,
to protect the dpcm instance.
The lock may be used in atomic context, so use spin lock.
Use irq spin lock version,
since the lock may be used in interrupts.
possible race condition between
void dpcm_be_disconnect(
...
list_del(&dpcm->list_be);
list_del(&dpcm->list_fe);
kfree(dpcm);
...
and
for_each_dpcm_fe()
for_each_dpcm_be*()
race condition example
Thread 1:
snd_soc_dapm_mixer_update_power()
-> soc_dpcm_runtime_update()
-> dpcm_be_disconnect()
-> kfree(dpcm);
Thread 2:
dpcm_fe_dai_trigger()
-> dpcm_be_dai_trigger()
-> snd_soc_dpcm_can_be_free_stop()
-> if (dpcm->fe == fe)
Excpetion Scenario:
two FE link to same BE
FE1 -> BE
FE2 ->
Thread 1: switch of mixer between FE2 -> BE
Thread 2: pcm_stop FE1
Exception:
Unable to handle kernel paging request at virtual address dead0000000000e0
pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c
sound/soc/soc-pcm.c:3226
if (dpcm->fe == fe)
lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c
Backtrace:
[<ffffff89602dba80>] notify_die+0x68/0xb8
[<ffffff896028c7dc>] die+0x118/0x2a8
[<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c
[<ffffff89602a27f4>] do_translation_fault+0x64/0xa0
[<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0
[<ffffff8960282ad0>] el1_da+0x24/0x40
[<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c
[<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c
[<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44
[<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c
[<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c
[<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128
[<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0
[<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14
[<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244
[<ffffff8960283740>] el0_svc_naked+0x34/0x38
[<ffffffffffffffff>] 0xffffffffffffffff
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
The current implementation of the hdac_hda codec results in zero-valued
samples on capture and noise with headset playback when SOF is used on
platforms with an on-board HDaudio codec. This is root-caused to SOF
using be_hw_params_fixup, and the prepare() call using invalid runtime
fields to determine the format.
This patch moves the format handling to the hw_params() callback, as
done already for hdac_hdmi, to make sure the fixed-up information is
taken into account but keeps the codec initialization in prepare() as
the stream_tag is only available at that time. Moving everything in the
prepare() callback is possible but the code is less elegant so this
two-step solution was chosen.
The solution was tested with the SST driver with no regressions, and all
the issues with SOF playback and capture are solved.
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>
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>
This patch sets missing stream_name of capture part of the DAI driver
so we can define DAPM routing properly also for the capture stream.
While at it "Playback" suffix is added to the playback stream names
to clearly identify playback/capture.
Together with related dts patch this fixes NULL pointer dereference
when opening ALSA device for recording on Odroid XU3.
Fixes: 64aba9bca5 ("ASoC: samsung: i2s: Add widgets and routes for DPCM support")
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 78a24e10cd ("ASoC: soc-core: clear platform pointers on error")
re-worked the clean-up of any platform pointers that may have been
initialised by the function snd_soc_init_platform(). This commit missed
one error path where if any of the prelinks for a soundcard failed to
initialise, then these platform pointers would not be cleaned-up. This
then prevents the soundcard from being initialised following a probe
deferral when any of the soundcard prelinks cannot be found.
Fix this by ensuring that soc_cleanup_platform() is called when
initialising the soundcard prelinks fails.
Fixes: 78a24e10cd ("ASoC: soc-core: clear platform pointers on error")
Signed-off-by: Jonathan Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Limiting the value of the passed in params->msbits in the hw_params()
callback is redundant on three counts:
1. We already specify in the DAI driver that we can only handle up to
24 bits. This means msbits will be limited to 24 via the ALSA
constraints imposed by the ASoC core, unless we have multiple codecs
that can handle more bits.
2. Nothing in our hw_params() implementation uses this value.
3. The copy of the params that we are passed by the ASoC core never
reads back the msbits value.
Consequently, this code is unnecessary and does nothing useful. Remove
it.
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>
Add error check on set_sync function return.
Add of_node_put() as of_get_parent() takes a reference
which has to be released.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When snd_pcm_stop_xrun() is called in interrupt routine,
substream context may have already been released.
Add protection on substream context.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Allow indexation of sai iec958 controls according
to device id.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation to enabling -Wimplicit-fallthrough, mark switch
cases where we are expecting to fall through.
This patch fixes the following warning:
In file included from sound/soc/codecs/ab8500-codec.c:24:
sound/soc/codecs/ab8500-codec.c: In function ‘ab8500_codec_set_dai_fmt’:
./include/linux/device.h:1485:2: warning: this statement may fall through [-Wimplicit-fallthrough=]
_dev_err(dev, dev_fmt(fmt), ##__VA_ARGS__)
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/ab8500-codec.c:2129:3: note: in expansion of macro ‘dev_err’
dev_err(dai->component->dev,
^~~~~~~
sound/soc/codecs/ab8500-codec.c:2132:2: note: here
default:
^~~~~~~
Warning level 3 was used: -Wimplicit-fallthrough=3
This patch is part of the ongoing efforts to enable
-Wimplicit-fallthrough.
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
Another batch of changes for ASoC, no big core changes - it's mainly
small fixes and improvements for individual drivers.
- A big refresh and cleanup of the Samsung drivers, fixing a number of
issues which allow the driver to be used with a wider range of
userspaces.
- Fixes for the Intel drivers to make them more standard so less likely
to get bitten by core issues.
- New driver for Cirrus Logic CS35L26.
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Merge tag 'asoc-v5.1-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More changes for v5.1
Another batch of changes for ASoC, no big core changes - it's mainly
small fixes and improvements for individual drivers.
- A big refresh and cleanup of the Samsung drivers, fixing a number of
issues which allow the driver to be used with a wider range of
userspaces.
- Fixes for the Intel drivers to make them more standard so less likely
to get bitten by core issues.
- New driver for Cirrus Logic CS35L26.
Dummy write in capture master mode is used to gate
bus clocks. This write is useless in slave mode
as the clocks are not managed by slave.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When snd_pcm_stop_xrun() is called in interrupt routine,
substream context may have already been released.
Add protection on substream context.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Clocks do not need to be released on driver removal,
as this is already managed before.
Remove useless remove callback.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DMA configuration is not balanced on start/stop.
Move DMA configuration to trigger callback.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move counter handling to trigger start section
to manage multiple start/stop events.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
I2S supports 16 bits data in 32 channel length.
However the expected driver behavior, is to
set channel length to 16 bits when data format is 16 bits.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because of regmap cache, interrupts may not be cleared
as expected.
Declare IFCR register as write only and make writings
to IFCR register unconditional.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_CROS_EC_CODEC depends on MFD_CROS_EC.
Add that dependency to SND_SOC_SDM845 to fix unmet direct dependencies
warning.
Fixes: 74c6ecf419 (ASoC: qcom: Kconfig: select dmic for sdm845)
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Tested-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Tested-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch enables the reuse of kbl_da7219_max98927 machine driver to
support max98373. The same machine driver is modified for cases where one
amplifier is swapped out with another. Most of the changes are about
renaming the codec and codec_dai names, with minor differences due to
support for 24 bits in one case and 16 in the other.
Signed-off-by: Jenny TC <jenny.tc@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently each SSI unit 's busif mode/adinr/dalign address is
registered by: (in busif4 case)
RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80)
RSND_GEN_M_REG(SSI_BUSIF4_ADINR,0x504, 0x80)
RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80)
But according to user manual 41.1.4 Register Configuration
ssi9 4/5/6/7 busif mode/adinr/dalign register address
( SSI9-[4/5/6/7]_BUSIF_[MODE/ADINR/DALIGN] )
are out of this rule.
This patch registers ssi9 4/5/6/7 mode/adinr/dalign register
as single register, and access these registers in case of
SSI9 BUSIF 4/5/6/7.
Fixes: commit 8c9d750333 ("ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As the compressed stream implementation has acquired support for
multiple DAI links and compressed streams it has become harder to
interpret messages in the kernel log. Add additional macros to include
the compressed DAI name in the log messages, allowing different streams
to be easily disambiguated.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, only a single compressed stream is supported per firmware.
Add support for multiple compressed streams on a single firmware, this
allows additional features like completely independent trigger words or
separate debug capture streams to be implemented.
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make the code slightly clearer and prepare things for the addition of
multiple compressed streams on a single DSP core.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for more refactoring add a helper function to strip the
padding from ADSP data.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The irq_get_irq_data() function doesn't return error pointers, it
returns NULL.
Fixes: 6ba9dd6c89 ("ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This addresses an issue pointed out by compiler warning:
sound/soc/samsung/odroid.c: In function ‘odroid_audio_probe’:
sound/soc/samsung/odroid.c:298:22: warning: ‘cpu_dai’ may be used
uninitialized in this function [-Wmaybe-uninitialized]
priv->clk_i2s_bus = of_clk_get_by_name(cpu_dai, "iis");
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Although qcom_snd_parse_of() tries to manage the of-node refcount,
there are still a few places that lead to the unblanced refcount in
the error code path. Namely,
- for_each_child_of_node() needs to unreference the iterator node if
aborting the loop in the middle,
- cpu, codec and platform node objects have to be unreferenced at each
iteration,
- platform and codec node objects have to be referred before jumping
to the error handling code that unreference them unconditionally.
This patch tries to address these by moving the assignment of platform
and codec node objects to the beginning of the loop and adding the
of_node_put() calls adequately.
Fixes: c25e295cd7 ("ASoC: qcom: Add support to parse common audio device nodes")
Cc: Patrick Lai <plai@codeaurora.org>
Cc: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The apq8016 driver leaves the of-node refcount at aborting from the
loop of for_each_child_of_node() in the error path. Not only the
iterator node of for_each_child_of_node(), the children nodes referred
from it for codec and cpu have to be properly unreferenced.
Fixes: bdb052e81f ("ASoC: qcom: add apq8016 sound card support")
Cc: Patrick Lai <plai@codeaurora.org>
Cc: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
