This series converts the Tegra DTs and drivers to use the common/
standard DMA and reset bindings, rather than custom bindings. It also
adds complete documentation for the Tegra clock bindings without
actually changing any binding definitions.
This conversion relies on a few sets of patches in branches from outside
the Tegra tree:
1) A patch to add an DMA channel request API which allows deferred probe
to be implemented.
2) A patch to implement a common part of the of_xlate function for DMA
controllers.
3) Some ASoC patches (which in turn rely on (1) above), which support
deferred probe during DMA channel allocation.
4) The Tegra clock driver changes for 3.14.
Consequently, this branch is based on a merge of all of those external
branches.
In turn, this branch is or will be pulled into a few places that either
rely on features introduced here, or would otherwise conflict with the
patches:
a) Tegra's own for-3.14/powergate and for-4.14/dt branches, to avoid
conflicts.
b) The DRM tree, which introduces new code that relies on the reset
controller framework introduced in this branch, and to avoid
conflicts.
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Merge tag 'tegra-for-3.14-dmas-resets-rework' into drm/for-next
ARM: tegra: implement common DMA and resets DT bindings
This series converts the Tegra DTs and drivers to use the common/
standard DMA and reset bindings, rather than custom bindings. It also
adds complete documentation for the Tegra clock bindings without
actually changing any binding definitions.
This conversion relies on a few sets of patches in branches from outside
the Tegra tree:
1) A patch to add an DMA channel request API which allows deferred probe
to be implemented.
2) A patch to implement a common part of the of_xlate function for DMA
controllers.
3) Some ASoC patches (which in turn rely on (1) above), which support
deferred probe during DMA channel allocation.
4) The Tegra clock driver changes for 3.14.
Consequently, this branch is based on a merge of all of those external
branches.
In turn, this branch is or will be pulled into a few places that either
rely on features introduced here, or would otherwise conflict with the
patches:
a) Tegra's own for-3.14/powergate and for-4.14/dt branches, to avoid
conflicts.
b) The DRM tree, which introduces new code that relies on the reset
controller framework introduced in this branch, and to avoid
conflicts.
Tegra124 adds a number of extra modules into the configlink bus, which
must be taken out of reset before the bus is used. Update the AHUB
driver to know about these extra modules (the AHUB HW module hosts the
configlink bus).
Based-on-work-by: Arun Shamanna Lakshmi <aruns@nvidia.com>
Based-on-work-by: Songhee Baek <sbaek@nvidia.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Mark Brown <broonie@linaro.org>
---
This patch depends on "ASoC: tegra: use reset framework" to compile,
which is ack'd and slated to go through a (large) topic branch in the
Tegra tree. So, we can either:
a) Merge that Tegra topic branch into the ASoC tree, then apply this.
Note that I haven't created the topic branch yet, since I'm still
waiting for DMA dependencies to be applied.
b) Apply this change to the Tegra tree too. This change isn't directly
related to the changes in the Tegra tree; it just makes use of the new
reset controller feature that's introduced there.
By passing no flags when calling snd_dmaengine_pcm_register() from
tegra_pcm.c, we end up using dma_request_slave_channel() rather than
dmaengine_pcm_compat_request_channel(), and hence rely on the standard
DMA DT bindings and stashing the DMA slave ID away during channel
allocation. This means there's no need to use a custom DT property to
store the slave ID. So, remove all the code that parsed it.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Mark Brown <broonie@linaro.org>
The Tegra30 I2S driver currently allocates DMA FIFOs from the AHUB only
when an audio stream starts playback. This is theoretically nice for
resource sharing, but makes no practical difference for any configuration
the drivers currently support. However, this deferral prevents conversion
to the standard DMA DT bindings, since conversion requires knowledge of
the specific DMA channel to be allocated, which in turn depends on which
specific FIFO was allocated.
For this reason, move the FIFO allocation into probe() to allow later
conversion to the standard DMA DT bindings.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Mark Brown <broonie@linaro.org>
Call pm_runtime_get_sync() before all register accesses; the HW requires
clocks to be running when accessing registers.
This hasn't been needed to date, since all register IO was performed
while playback was active, and hence the ASoC core had already called
pm_runtime_get(). However, an imminent future commit will allocate and
set up the FIFOs and routing during probe(), when that "protection"
won't be in place.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Mark Brown <broonie@linaro.org>
Tegra's clock driver now provides an implementation of the common
reset API (include/linux/reset.h). Use this instead of the old Tegra-
specific API; that will soon be removed.
This change also renames "clock"/"clk" to "modules"/"mod" in symbols
related to entries in configlink_clocks[], since:
- We don't care about clock handles any more, but rather reset handles,
so the old name isn't applicable.
- It really is a list of modules on the bus, about which we currently
only care about reset handles.
If we start caring about any other aspect of the modules in the future,
we won't have to rename all these symbols again.
Note: The addition of "depends COMMON_CLOCK" is something that was missing
before, not a new requirement.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Thierry Reding <treding@nvidia.com>
This is the work so far on dmaengine for v3.14, it is being cross merged
into the Tegra tree to support a large DMA overhaul there. The main
additions are a change in the DMA request API which allows better
interaction at system startup using deferred probes and methods for
overriding the default device and channel names used to request DMA.
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Merge tag 'asoc-dma-v3.14' into for-3.14/dmas-resets-rework
ASoC: dma: Generic ASoC dmaengine driver enhancements
This is the work so far on dmaengine for v3.14, it is being cross merged
into the Tegra tree to support a large DMA overhaul there. The main
additions are a change in the DMA request API which allows better
interaction at system startup using deferred probes and methods for
overriding the default device and channel names used to request DMA.
Check the return value of dma_request_slave_channel_reason() to see if
deferred probe happens, not the variable the return value will be
assigned to later.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Fixes: 5eda87b890 ("ASoC: dmaengine: support deferred probe for DMA channels")
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Enhance dmaengine_pcm_request_chan_of() to support deferred probe for
DMA channels, by using the new dma_request_slave_channel_or_err() API.
This prevents snd_dmaengine_pcm_register() from succeeding without
acquiring DMA channels due to the relevant DMA controller not yet being
registered.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
AD1986A codec is a pretty old codec and has really many hidden
restrictions. One of such is that each DAC is dedicated to certain
pin although there are possible connections. Currently, the generic
parser tries to assign individual DACs as much as possible, and this
lead to two bad situations: connections where the sound actually
doesn't work, and connections conflicting other channels.
We may fix this by trying to find the best connections more harder,
but as of now, it's easier to give some hints for paired DAC/pin
connections and honor them if available, since such a hint is needed
only for specific codecs (right now only AD1986A, and there will be
unlikely any others in future).
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell machines with codec whose Subsystem Id is 0x10280624,
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/bugs/1259790
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case a single HDA card has both HDMI and S/PDIF outputs, the S/PDIF
outputs will have their IEC958 controls created starting from index 16
and the HDMI controls will be created starting from index 0.
However, HDMI simple_playback_build_controls() as used by old VIA and
NVIDIA codecs incorrectly requests the IEC958 controls to be created
with an S/PDIF type instead of HDMI.
In case the card has other codecs that have HDMI outputs, the controls
will be created with wrong index=16, causing them to e.g. be unreachable
by the ALSA "hdmi" alias.
Fix that by making simple_playback_build_controls() request controls
with HDMI indexes.
Not many cards have an affected configuration, but e.g. ASUS M3N78-VM
contains an integrated NVIDIA HDA "card" with:
- a VIA codec that has, among others, an S/PDIF pin incorrectly
labelled as an HDMI pin, and
- an NVIDIA MCP7x HDMI codec.
Reported-by: MysterX on #openelec
Tested-by: MysterX on #openelec
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org> # 3.8+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not all channels have been initialized, so far, especially when aamix
NID itself doesn't have amps but its leaves have. This patch fixes
these holes. Otherwise you might get unexpected loopback inputs,
e.g. from surround channels.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell Inspiron 3045 machine (codec Subsystem Id: 0x10280628),
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259437
CC: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell Optiplex 3030 machine (codec Subsystem Id: 0x10280623),
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259435
CC: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add fields to struct snd_dmaengine_pcm_config to allow custom:
- DMA channel names.
This is useful when the default "tx" and "rx" channel names don't
apply, for example if a HW module supports multiple channels, each
having different DMA channel names. This is the case with the FIFOs
in Tegra's AHUB. This new facility can replace
SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME.
- DMA device
This allows requesting DMA channels for a device other than the device
which is registering the "PCM" driver. This is quite unusual, but is
currently useful on Tegra. In much HW, and in Tegra20, each DAI HW
module contains its own FIFOs which DMA writes to. However, in Tegra30,
the DMA FIFOs were split out AHUB HW module, which then routes the data
through a cross-bar, and into the DAI HW modules. However, the current
ASoC driver structure does not expose this detail, and acts as if the
FIFOs are still part of the DAI HW modules. Consequently, the "PCM"
driver is registered with the DAI HW module, yet the DMA channels must
be looked up in the AHUB HW module's device tree node. This new config
field allows that to happen. Eventually, the Tegra drivers will be
reworked to fully expose the AHUB, and this config field can be
removed.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
If snd_dmaengine_pcm_register()'s call to snd_soc_add_platform() fails,
all objects allocated during registration are leaked. Fix this by adding
error-handling code.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Restructure the internals of dmaengine_pcm_request_chan_of() as a loop
over all channels to be allocated. This makes it easier to add logic
that applies to all allocated channels, without having to duplicate that
logic in each of the half-duplex/full-duplex paths.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
I've tested the old Dell Vostro 131 with the latest generic parser
and it works just fine, and as a bonus we get better jack detection
features in userspace. Therefore this quirk can be removed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following warning when optimizing for size with gcc-4.6.4:
sound/usb/mixer_quirks.c:1514:6: warning: 'err' may be used uninitialized in this function [-Wuninitialized]
Signed-off-by: Mikulas Patocka <mpatocka@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add quirk for Acer Aspire E-572:
- fix external mic
- limit mic boost for internal mic with maximal noise level of -24dB
Signed-off-by: Oleksij Rempel <linux@rempel-privat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MacBook Air 2,1 has a fairly different pin assignment from its brother
MBA 1,1, and yet another quirks are needed for pin 0x18 and 0x19,
similarly like what iMac 9,1 requires, in order to make the sound
working on it.
Reported-and-tested-by: Bruno Prémont <bonbons@linux-vserver.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A smattering of fixes here, some core ones for the rate combination
issues for things other than simple bitmasks, for readback of byte
controls and for updating the power of value muxes plus a bunch of
driver fixes of varying severity.
The warning fix in the i.MX FIQ driver is fixing a warning introduced
by a previous fix.
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Merge tag 'asoc-v3.13-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.13
A smattering of fixes here, some core ones for the rate combination
issues for things other than simple bitmasks, for readback of byte
controls and for updating the power of value muxes plus a bunch of
driver fixes of varying severity.
The warning fix in the i.MX FIQ driver is fixing a warning introduced
by a previous fix.
In the case of using jackpoll_ms instead of unsol events, the jack
was correctly detected, but ELD info was not refreshed on plug-in.
And without ELD info, no proper restriction of pcm, which can in turn
break sound output on some devices.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I forgot to remove the hp_automute_hook from alc283_fixup_chromebook.
It doesn't need this for other chrome os machine.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to WM8731 "PD, Rev 4.9 October 2012" datasheet, when it
works in DSP mode A, LRP = 1, while works in DSP mode B, LRP = 0.
So, fix LRP for DSP mode as the datesheet specification.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Create single model for HP.
The headset jack module was difference between other chrome book.
It need to manual control Mic jack detect.
Chrome OS loaded driver by models. Remove old assigned fixup table from
ALC269 fixup list entry.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By trial and error, I found this patch could work around an issue
where the headset mic would stop working if you switch between the
internal mic and the headset mic, and the internal mic was muted.
It still takes a second or two before the headset mic actually starts
working, but still better than nothing.
Information update from Kailang:
The verb was ADC digital mute(bit 6 default 1).
Switch internal mic and headset mic will run alc_headset_mode_default.
The coef index 0x11 will set to 0x0041.
Because headset mode was fixed type. It doesn't need to run
alc_determine_headset_type.
So, the value still keep 0x0041. ADC was muted.
BugLink: https://bugs.launchpad.net/bugs/1256840
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that AD1986A cannot manage the dynamic pin on/off for
auto-muting, but rather gets confused. Since each output has own amp,
let's use it instead.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Cc: <stable@vger.kernel.org> [v3.11+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ad_vmaster_eapd_hook() needs to handle the inverted EAPD case
properly, too. Otherwise the output gets broken on Lenovo N100 with
AD1986A codec.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS Z35HL laptop also needs the very same fix as the previous one
that was applied to ASUS W7J.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66231
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The static checker found a possible array overflow in atmel/abdac.c:
static checker warning: "sound/atmel/abdac.c:373 set_sample_rates()
error: buffer overflow 'dac->rates' 6 <= 6"
This patch papers over the buggy point, by ensuring that dac->rates[]
update not overflowing the actual array size.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the probe of snd-hda-intel driver is deferred due to f/w loading
or the nested module loading, complete_all() should be also delayed
until the initialization really finished. Otherwise, vga-switcheroo
client would start switching before the actual init is done.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that EAPD on NID 0x16 is the only control over all outputs on
HP machines with AD1984A while turning EAPD on NID 0x12 breaks the
output. Thus we need to avoid fiddling EAPD on NID. As a quick
workaround, just set own_eapd_ctrl flag for the wrong EAPD, then
implement finer EAPD controls.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66321
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For dmaengine drivers which do not support transfer residue reporting we update
the PCM pointer with period granularity. Set the SNDRV_PCM_INFO_BATCH flag in
this case to let userspace know about this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
N810 audio driver has stopped working at some point. Probably when
OMAP2 was converted to common clock framework since now call to clk_enable
dumps the stack trace in drivers/clk/clk.c: __clk_enable() due
clk->prepare_count is zero.
Fix this by converting clk_enable/_disable calls to those that take care
of clock prepare/unprepare.
I'm not queueing this to linux-stable since OMAP2 common clock framework
conversion in commit ed1ebc4948 ("ARM: OMAP2: clock: Convert to common clk")
happened before N810 was really usable in mainline and user base for N810 is
anyway small. Potential linux-stable candidates are only those after
commit 3d3a6d18ab ("watchdog: introduce retu_wdt driver").
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
platform_set_drvdata(op, pdata) in pcm030_fabric_probe()
will be overwrited when calling snd_soc_register_card(card),
but cm030_fabric_remove() use drvdata as a type of struct
pcm030_audio_data, so we should move platform_set_drvdata()
below snd_soc_register_card() call.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@linaro.org>
Originally snd_hrtimer_callback() used iprtd->period_time for
some jiffies based estimation to determine the right moment
to call snd_pcm_period_elapsed(). As timer drifts may well be a
problem, this was changed in commit b4e82b5b78 to be based
on buffer transmission progress, using iprtd->offset and
runtime->buffer_size to calculate the amount of data since last
period had elapsed.
Unfortunately, iprtd->offset counts in bytes, while
runtime->buffer_size counts frames, so adding these to find some
delta is like comparing apples and oranges, and eventually results
in negative delta values every now and then. This is no big harm,
because it simply causes snd_pcm_period_elapsed() being called
more often than necessary, as negative delta is taken for a
large unsigned value by implicit conversion rule.
Nonetheless, the calculation is broken, so one would replace
the runtime->buffer_size by its equivalent in bytes.
But then, there are chances snd_pcm_period_elapsed() is called
late, because calculating the moment for the elapsed period
into delta is based against the iprtd->last_offset, which is not
necessarily the first byte of the period in question, but some
random byte which the FIQ handler left us with in r8/r9 by
accident. Again, negative impact is low, as there are plenty of
periods already prefilled with data, and snd_pcm_period_elapsed()
will probably be called latest when the following period is
reached. However, the calculation is conceptually broken, and we
are best off removing the clever stuff altogether.
snd_pcm_period_elapsed() is now simply called once everytime
snd_hrtimer_callback() is run, which may not be most accurate,
but at least this way we are quite sure we dont miss an end of
period. There is not much extra effort wasted by superfluous
calls to snd_pcm_period_elapsed(), as the timer frequency
closely matches the period size anyway.
Signed-off-by: Oskar Schirmer <oskar@scara.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The recent kernels got regressions on ASUS W7J with ALC660 codec where
no sound comes out. After a long debugging session, we found out that
setting the pin control on the unused NID 0x10 is mandatory for the
outputs. And, it was found out that another magic of NID 0x0f that is
required for other ASUS laptops isn't needed on this machine.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66081
Reported-and-tested-by: Andrey Lipaev <lipaev@mail.ru>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_bytes_put treats the data in the binary control as big endian
words, however snd_soc_bytes_get uses the endian of the host machine.
This causes the two functions to be inconsistant with how the mask is
applied on little endian machines.
This patch applies the big_endian format used in snd_soc_bytes_put to
snd_soc_bytes_get.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The array limits are supposed to be in units of u32 instead of in bytes.
The current code has a potential array overflow.
Fixes: c614475b0e ('ALSA: dice: add a proc file to show device information')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This machine also has mono output if run through DAC node 0x03.
Cc: stable@vger.kernel.org (v3.10+)
BugLink: https://bugs.launchpad.net/bugs/1256212
Tested-by: David Chen <david.chen@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the previous commit 1f0bbf03cb added the pin config for the bass
speaker, this patch adds the corresponding LFE-only channel map on
ASUS ET2700.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a fixup entry for the missing bass speaker pin 0x16 on ASUS ET2700
AiO desktop. The channel map will be added in the next patch, so that
this can be backported easily to stable kernels.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For many drivers using the generic dmaengine PCM driver one of the few (or the
only) things left to do in the drivers remove function is to unregister the PCM
device. This patch adds a resource managed version of snd_dmaengine_pcm_register()
which makes it possible to simplify the remove function as well as the error
path in the probe function for those drivers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Thierry Reding