soc-pcm.c has soc_pcm_components_close() but not have its open()
side function. This kind of unbalance function is very unreadable.
And, current error handling is not correct.
Because it is using for_each_rtdcom() loop, we need to call
soc_pcm_components_close() anyway even though
CPU DAI .startup() failed.
This patch adds soc_pcm_components_open(), and fixup error
handling issue.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Codec side is setting codec_dai->rate = 0 when error case
at soc_pcm_hw_params(), but there is not such setting for CPU side.
This patch adds it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
cpu_dai related operation is separated by component operation at
soc_pcm_hw_params() somehow.
It is not readable, let's do it at same place
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If playback/capture is paused and system enters S3, after system returns
from suspend, BE dai needs to call prepare() callback when playback/capture
is released from pause if RESUME_INFO flag is not set.
Currently, the dpcm_be_dai_prepare() function will block calling prepare()
if the pcm is in SND_SOC_DPCM_STATE_PAUSED state. This will cause the
following test case fail if the pcm uses BE:
playback -> pause -> S3 suspend -> S3 resume -> pause release
The playback may exit abnormally when pause is released because the BE dai
prepare() is not called.
This patch allows dpcm_be_dai_prepare() to call dai prepare() callback in
SND_SOC_DPCM_STATE_PAUSED state.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Like for hw_params, hw_free should not be called on codec dai for
which the current stream is invalid.
Fixes: cde79035c6 ("ASoC: Handle multiple codecs with split playback / capture")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A stream may specify a rate range using 'rate_min' and 'rate_max', so a
stream may be valid and not specify any rates. However, as stream cannot
be valid and not have any channel. Let's use this condition instead to
determine if a stream is valid or not.
Fixes: cde79035c6 ("ASoC: Handle multiple codecs with split playback / capture")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some drivers mandate setting up hw params after resuming from system sleep.
Since, the hw_params ioctl is not invoked upon resuming, the fixed-up BE
dai hw params should be saved so the driver can use it in its resume
sequence.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Handle error before returning when try_module_get() fails
to prevent inconsistent mutex lock/unlock.
Fixes: 52034add7 (ASoC: pcm: update module refcount if
module_get_upon_open is set)
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Setting the module_get_upon_open field for component driver
prevents the module refcount from being incremented during
component probe(). This could lead to the module being
allowed to be unloaded when a pcm stream is open. So,
if this field is set, the module's refcount should be
incremented during pcm open to prevent module removal
when the component is in use. And, the refcount should
be decremented upon pcm close.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If for any reason, the backend does not have the requested substream
(like capture on a playback only backend), the BE will be skipped in
dpcm_be_dai_startup().
However, dpcm_apply_symmetry() does not skip those BE and will
dereference the be_substream (NULL) pointer anyway.
Like in dpcm_be_dai_startup(), just skip those BE.
Fixes: 906c7d690c ("ASoC: dpcm: Apply symmetry for DPCM")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.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>
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>
Lots and lots of new drivers so far, a highlight being the MediaTek
BTCVSD which is a driver for a Bluetooth radio chip - the first such
driver we've had upstream. Hopefully we will soon also see a baseband
with an upstream driver!
- Support for only powering up channels that are actively being used.
- Quite a few improvements to simplify the generic card drivers,
especially the merge of the SCU cards into the main generic drivers.
- Lots of fixes for probing on Intel systems, trying to rationalize
things to look more standard from a framework point of view.
- New drivers for Asahi Kasei Microdevices AK4497, Cirrus Logic CS4341,
Google ChromeOS embedded controllers, Ingenic JZ4725B, MediaTek
BTCVSD, MT8183 and MT6358, NXP MICFIL, Rockchip RK3328, Spreadtrum
DMA controllers, Qualcomm WCD9335, Xilinx S/PDIF and PCM formatters.
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Merge tag 'asoc-v5.1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v5.1
Lots and lots of new drivers so far, a highlight being the MediaTek
BTCVSD which is a driver for a Bluetooth radio chip - the first such
driver we've had upstream. Hopefully we will soon also see a baseband
with an upstream driver!
- Support for only powering up channels that are actively being used.
- Quite a few improvements to simplify the generic card drivers,
especially the merge of the SCU cards into the main generic drivers.
- Lots of fixes for probing on Intel systems, trying to rationalize
things to look more standard from a framework point of view.
- New drivers for Asahi Kasei Microdevices AK4497, Cirrus Logic CS4341,
Google ChromeOS embedded controllers, Ingenic JZ4725B, MediaTek
BTCVSD, MT8183 and MT6358, NXP MICFIL, Rockchip RK3328, Spreadtrum
DMA controllers, Qualcomm WCD9335, Xilinx S/PDIF and PCM formatters.
Currently all widgets attached to a DAI link will be powered
up when the DAI is active, however this may include routes
that are not actually in use if there are unused channels
available on the DAI.
The macros for creating AIF widgets already include an entry for
slot, it is proposed to change that to channel. The effective
difference here being respresenting the logical channel index
rather than the physical slot index. The CODECs currently
using the slot entry on the DAPM_AIF macros are using it in
a manner consistent with this, the CODECs not using it just
have the field set to zero.
A variable is added to snd_soc_dapm_widget to represent
this channel index and then for each AIF widget attached to
a DAI this is compared against the number of channels on
the stream. Enabling the links for those which will be in
use. This has the nice property that the CODECs which haven't
used the slot/channel entry in the macro will function exactly
as before due to all the AIF widgets having a channel of zero
and a stream by definition having at least one channel.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Until now we rely on each driver calling snd_pcm_suspend*() explicitly
at its own PM handling. However, this can be done far more easily by
setting the PM ops to each actual snd_pcm device object.
This patch adds the device_type object for PCM stream and assigns to
each PCM stream object. The type contains only the PM ops for system
suspend; we don't need to deal with the resume in general.
The suspend hook simply calls snd_pcm_suspend_all() for the given PCM
streams. This implies that the PM order is correctly put, i.e. PCM is
suspended before the main (or codec) driver, which should be true in
general. If a special ordering is needed, you'd need to adjust the
device PM order manually later.
This patch introduces a new flag, snd_pcm.no_device_suspend, too.
With this flag set, the PCM device object won't invoke
snd_pcm_suspend_all() by itself. This is needed for ASoC who wants to
manage the PM call orders in its serialized way, and the flag is set
in soc_new_pcm() as default.
For the non-ASoC world, we can get rid of the manual snd_pcm_suspend
calls. This will be done in the later patches.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To be more readable code, this patch adds
new for_each_dpcm_be() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_dpcm_fe() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_card_rtds() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 0b7990e389 ("ASoC: add for_each_rtd_codec_dai() macro")
added for_each_rtd_codec_dai_reverse(). but _rollback() is better
naming than _reverse(). This patch rename it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 0b7990e389 ("ASoC: add for_each_rtd_codec_dai() macro")
added for_each_rtd_codec_dai(), but it didn't convert few loop
which is not using "rtd". This patch fixup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC snd_soc_pcm_runtime has snd_soc_dai array for codec_dai.
To be more readable code, this patch adds
new for_each_rtd_codec_dai() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the CPU DAI does not initialise rate_max, say if using
using KNOT or CONTINUOUS, then the rate_max field will be
initialised to 0. A value of zero in the rate_max field of
the hardware runtime will cause the sound card to support no
sample rates at all. Obviously this is not desired, just a
different mechanism is being used to apply the constraints. As
such update the setting of rate_max in dpcm_init_runtime_hw
to be consistent with the non-DPCM cases and set rate_max to
UINT_MAX if nothing is defined on the CPU DAI.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Take into account the base delay set in pointer callback.
There are cases where a pointer function populates
runtime->delay, such as:
./sound/pci/hda/hda_controller.c
./sound/soc/intel/atom/sst-mfld-platform-pcm.c
This delay was getting lost and was overwritten by delays
from codec or cpu dai delay function if exposed.
Now,
Total delay = base delay + cpu_dai delay + codec_dai delay
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
As done for format and channels, add the possibility to merge
the backend rates on the frontend rates.
This useful if the backend does not support all rates supported by the
frontend, or if several backends (cpu and codecs) with different
capabilities are connected to the same frontend.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The goal of this patch is to simplify a bit dpcm runtime stream merge
by removing several local variables.
ATM, merge functions return the BE 'filter' values which should then be
filtered against the FE stream values. This create a lot of local
variable and unnecessary init of min and max.
Instead of this, we can pass the FE stream values directly and let the
BE filtering functions perform the merge 'in-place'
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Machine drivers statically define a number of DAI links that currently
cannot be changed or removed by topology. This means PCMs and platform
components cannot be changed by topology at runtime AND machine drivers
are tightly coupled to topology.
This patch allows topology to override the machine driver DAI link config
in order to reuse machine drivers with different topologies and platform
components. The patch supports :-
1) create new FE PCMs with a topology defined PCM ID.
2) destroy existing static FE PCMs
3) change the platform component driver.
4) assign any new HW params fixups.
5) assign a new card name prefix to differentiate this topology to userspace.
The patch requires no changes to the machine drivers, but does add some
platform component flags that the platform component driver can assign
before loading topologies.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When merging codec formats, dpcm_runtime_base_format() should skip
the codecs which are not supporting the current stream direction.
At the moment, if a BE link has more than one codec, and only one
of these codecs has no capture DAI, it becomes impossible to start
a capture stream because the merged format would be 0.
Skipping invalid codec DAI solves the problem.
Fixes: b073ed4e21 ("ASoC: soc-pcm: DPCM cares BE format")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Extend dpcm_merge_chan to also check backend cpu dai channels
capabilities. Apply the same policy as soc_pcm_init_runtime_hw() for
multicodec links and only check cpu dai in this case.
Cc: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As it is, dpcm_runtime_update() performs the old path and new path
update of a frontend before going on to the next frontend DAI.
Depending the order of the FEs within the rtd list, the result of
the update might be different.
For example:
* Frontend A connected to backend C, with a 48kHz playback
* Frontend B connected to backend D, with a 44.1kHz playback
* FE A appears before FE B in the rtd list of the card.
If we reparent BE C to FE B (disconnecting BE D):
* old path update of FE A will run first, and BE C will get hw_free()
and shutdown()
* new path update of FE B will run after and BE C, which is stopped,
so it will be configured at 44.1kHz, as expected
If we reparent BE D to FE A (disconnecting BE C):
* new path update of FE A will run first but since BE D is still running
at 44.1kHz, it won't be reconfigured (no call to startup() or
hw_params())
* old path update of FE B runs after, nothing happens
* In this case, we end up with a BE playing at 44.1kHz a stream which is
supposed to be played at 48Khz (too slow)
To improve this situation, this patch performs all the FE old paths update
before going on to update the new paths. With this, the result should
no longer depend on the order of the FE within the card rtd list.
Please note that there might be a small performance penalty since
dpcm_process_paths() is called twice per stream direction.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current DPCM is caring only FE channel configuration. Sometimes
it will be trouble if user selects channel which isn't supported
by BE.
This patch adds new .dpcm_merged_chan on struct snd_soc_dai_link.
DPCM will use FE / BE merged channel if struct snd_soc_dai_link
has it.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, the core will continue processing open/hw_params
component callbacks after one has failed even though it will abort
immediately afterwards. This is unnecessary and also has the issue
that close/hw_free will be called on the component which failed
open/hw_params which could result in issues if the driver doesn't
expect this behaviour.
Update the core to abort processing open/hw_params when an error
is hit and only call close/hw_free for those components that were
successfully opened.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When the call to hw_params for a component fails, the error code is held
by the variable '__ret' but the error message displays the value held by
the variable 'ret'. Fix the return code shown when hw_params fails for
a component.
Fixes: b8135864d4 ("ASoC: snd_soc_component_driver has snd_pcm_ops")
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should set BE symmetric constraint on FE substream.
in case one BE is used by two FE1/FE2,
the first BE runtime will use FE1's substream->runtime.
hence the FE1's will be constrained by BE symmetry property.
Though, second FE2 call dpcm_apply_symmetry,
the be_substream->runtime == FE1's substream->runtime.
The FE2's substream->runtime will not be constrained
by BE's symmetry property.
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In case, one BE is used by two FE1/FE2
FE1--->BE-->
|
FE2----]
when FE1/FE2 call dpcm_be_dai_hw_free() together
the BE users will be 2 (> 1), hence cannot be hw_free
the be state will leave at, ex. SND_SOC_DPCM_STATE_STOP
later FE1/FE2 call dpcm_be_dai_shutdown(),
will be skip due to wrong state.
leaving the BE not being hw_free and shutdown.
The BE dai will be hw_free later when calling
dpcm_be_dai_shutdown() if still in invalid state.
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Log the correct error code in case the .open() call to a component fails.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now no one is using Codec related code.
Let's remove all
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit ef050bece1 ("ASoC: Remove platform code now everything is
componentised") removed platform code, but it didn't remove
.pcm_new/free which existed only for platform.
This patch remove these
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As all drivers have been moved over to the new generic component
code remove the now unused platform specific code.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Krzysztof Kozlowski reported a NULL dereference in _instantiate_card()
on Odroid XU3 and XU boards which he bisected to 45f8cb57da (ASoC:
core: Allow topology to override machine driver FE DAI link config).
Revert that commit for now, along with f11a5c27f9 (ASoC: core: Add
name prefix for machines with topology rewrites) due to dependency
issues, in order to keep things booting cleanly in -next.
Reported-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Machine drivers statically define a number of DAI links that currently
cannot be changed or removed by topology. This means PCMs and platform
components cannot be changed by topology at runtime AND machine drivers
are tightly coupled to topology.
This patch allows topology to override the machine driver DAI link config
in order to reuse machine drivers with different topologies and platform
components. The patch supports :-
1) create new FE PCMs with a topology defined PCM ID.
2) destroy existing static FE PCMs
3) change the platform component driver.
4) assign any new HW params fixups.
The patch requires no changes to the machine drivers, but does add some
platform component flags that the platform component driver can assign
before loading topologies.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Improve the DPCM BE search debug output to make it easier to debug
issues in topologies.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit fbb16563c6 ("ASoC: snd_soc_component_driver has pmdown_time")
added new .pmdown_time which is for inverted version of current
.ignore_pmdown_time
But it is confusable name. Let's rename it to .use_pmdown_time
Reported-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit f523acebbb ("ASoC: add Component level pcm_new/pcm_free v2")
added component level pcm_new/pcm_free, but flush_delayed_work()
on soc_pcm_private_free() is called in for_each_rtdcom() loop.
It doesn't need to be called many times.
This patch moves it out of loop.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current snd_soc_runtime_ignore_pmdown_time() tallys all Codec and
CPU's "ignore_pmdown_time". Now, CPU (= via compoent)
ignore_pmdown_time is fixed as "true". Codec's one is copied from Codec
driver. This means Codec side default is "false".
Current all Codec driver will be replaced into Component, thus, we can
use for_each_rtdcom() for this totalization. This patch adds new
"pmdown_time" on Component driver. Its inverted value will be used
for this "ignore" totalizaton.
Of course all existing Component driver doesn't have its settings now,
thus, all existing "pmdown_time" is "false". This means all
Components will ignore pmdown time. This is current CPU behavior.
To keep compatibility, snd_soc_runtime_ignore_pmdown_time() totalize
Component's inverted "pmdown_time" (= total will be true) and
Codec's "ignore_pmdown_time" (= depends on Codec driver settings).
Because It is using AND operation, its result is based on Codec driver
settings only.
This means this operation can keep compatibility and doesn't have
nonconformity.
When we replace Codec to Component, the driver which has
".ignore_pmdown_time = true" will be just removed,
and the driver which doesn't have it will have new
".pmdown_time = true".
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Platform will be replaced into Component in the future.
snd_soc_platform_driver has snd_pcm_ops, but snd_soc_component_driver
doesn't have it. To prepare for replacing, this patch adds snd_pcm_ops
on component driver.
platform will be replaced into component, and its code will be removed.
But during replacing, both platform and component process code exists.
To keep compatibility, to avoid platform NULL access and to avoid
platform/component duplicate operation during replacing process, this
patch has such code. Some of this code will be removed when platform was
removed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current ALSA SoC, Platform only has pcm_new/pcm_free feature,
but it should be supported on Component level. This patch adds it.
The v1 was added commit 99b04f4c40 ("ASoC: add Component level
pcm_new/pcm_free") but it called all "card" connected component's
pcm_new/free, it was wrong.
This patch calls "rtd" connected component.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When ASoC driver is unbound dynamically during its operation (i.e. a
kind of hot-unplug), we may hit Oops due to the resource access after
the release by a delayed work, something like:
Unable to handle kernel paging request at virtual address dead000000000220
....
PC is at soc_dapm_dai_stream_event.isra.14+0x20/0xd0
LR is at snd_soc_dapm_stream_event+0x74/0xa8
....
[<ffff000008715610>] soc_dapm_dai_stream_event.isra.14+0x20/0xd0
[<ffff00000871989c>] snd_soc_dapm_stream_event+0x74/0xa8
[<ffff00000871b23c>] close_delayed_work+0x3c/0x50
[<ffff0000080bbd6c>] process_one_work+0x1ac/0x318
[<ffff0000080bbf20>] worker_thread+0x48/0x420
[<ffff0000080c201c>] kthread+0xfc/0x128
[<ffff0000080842f0>] ret_from_fork+0x10/0x18
For fixing the race, this patch adds a sync-point in pcm private_free
callback to finish the delayed work before actually releasing the
resources.
Reported-by: Hiep Cao Minh <cm-hiep@jinso.co.jp>
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Double NULL pointer check for ops and ops->func is difficult to read
and might be forget to check it if new func was add.
This patch adds new null_snd_soc_ops and use it if rtd->dai_link didn't
have it to avoid NULL ops, and reduces ops NULL check.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On soc_add_dai(), it uses null_dai_ops if driver doesn't have
its own ops. This means, dai->driver->ops never been NULL.
dai->driver->ops check is not needed.
This patch removes it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
hw_params may be fixup by be_hw_params_fixup, calling
soc_pcm_params_symmetry() before hw_params will have issue
if there is hw_params changes in be_hw_params_fixup.
For example, with following use case
1. a dai-link which is able to convert sample rate on BE side
2. set BE playback and capture sample rate to 44100Hz
3. play a 48000Hz audio stream with this dai-link
4. record from this dai-link with 44100Hz sample rate
Got following error message when record starts
[ 495.013527] be_link_ak4613: ASoC: unmatched rate symmetry: 48000 - 44100
[ 495.021729] be_link_ak4613: ASoC: hw_params BE failed -22
[ 495.028589] rsnd_link0: ASoC: hw_params BE failed -22
Because in soc_pcm_hw_params(), FE rate is still having value before
it is fixup by be_hw_params_fixup(), when soc_pcm_params_symmetry() checks
symmetry, thus soc_pcm_params_symmetry() complains about the unmatched rate
between the active stream and the new stream tries to start.
This patch moves soc_pcm_params_symmetry() after hw_params to resolve the
above issue.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Basically, current ALSA SoC framework is based on CPU/Codec/Platform,
but its operation doesn't have consistent.
Thus, source code was unreadable, and difficult to understand.
This patch connects each component (= CPU/Codec/Platform) to rtd by
using snd_soc_rtdcom_add(), and convert uneven operations to consistent
operation.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
'debugfs_dpcm_state' member from structure snd_soc_pcm_runtime
is never used at all, so it is safe to remove it.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 99b04f4c40 ("ASoC: add Component level
pcm_new/pcm_free"), which started calling the pcm_new callback for every
component in a *card* when creating a new pcm, something which does not
seem to make any sense.
This specifically led to memory leaks in systems with more than one
platform component and where DMA memory is allocated in the
platform-driver callback. For example, when both mcasp devices are being
used on an am335x board, DMA memory would be allocated twice for every
DAI link during probe.
When CONFIG_SND_VERBOSE_PROCFS was set this fortunately also led to
warnings such as:
WARNING: CPU: 0 PID: 565 at ../fs/proc/generic.c:346 proc_register+0x110/0x154
proc_dir_entry 'sub0/prealloc' already registered
Since there seems to be no users of the new component callbacks, and the
current implementation introduced a regression, let's revert the
offending commit for now.
Fixes: 99b04f4c40 ("ASoC: add Component level pcm_new/pcm_free")
Signed-off-by: Johan Hovold <johan@kernel.org>
Reviewed-by: Linus Walleij <linus.walleij@linaro.org>
Tested-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable <stable@vger.kernel.org> # 4.10
Multiple frontend dailinks may be connected to a backend
dailink at the same time. When one of frontend dailinks is
closed, the associated backend dailink should not be closed
if it is connected to other active frontend dailinks. Change
ensures that backend dailink is closed only after all
connected frontend dailinks are closed.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Now that all users of old copy and silence ops have been converted to
the new PCM ops, the old stuff can be retired and go away.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For supporting the explicit in-kernel copy of PCM buffer data, and
also for further code refactoring, three new PCM ops, copy_user,
copy_kernel and fill_silence, are introduced. The old copy and
silence ops will be deprecated and removed later once when all callers
are converted.
The copy_kernel ops is the new one, and it's supposed to transfer the
PCM data from the given kernel buffer to the hardware ring-buffer (or
vice-versa depending on the stream direction), while the copy_user ops
is equivalent with the former copy ops, to transfer the data from the
user-space buffer.
The major difference of the new copy_* and fill_silence ops from the
previous ops is that the new ops take bytes instead of frames for size
and position arguments. It has two merits: first, it allows the
callback implementation often simpler (just call directly memcpy() &
co), and second, it may unify the implementations of both interleaved
and non-interleaved cases, as we'll see in the later patch.
As of this stage, copy_kernel ops isn't referred yet, but only
copy_user is used.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No one is using snd_soc_platform_trigger().
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No existing platform is using .bespoke_trigger.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No existing platform is using .delay.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When multiple front-ends are using the same back-end, putting state of a
front-end to STOP state upon receiving pause command will result in backend
stream getting released by DPCM framework unintentionally. In order to
avoid backend to be released when another active front-end stream is
present, put the stream state to PAUSED state instead of STOP state.
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current ALSA SoC, Platform only has pcm_new/pcm_free feature,
but it should be supported on Component level. This patch adds it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
dpcm_state_string() returns a pointer to a string literal. Modifying a
string literal causes undefined behaviour. So make the return type of the
function const char * to make it explicit that the returned value should
not be modified.
This patch is purely cosmetic, none of the users of dpcm_state_string()
attempt to modify the returned content.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
When operating the BE, we should print out the dai_link name of BE other
than FE. This is useful when analyzing the kernel log.
Signed-off-by: Donglin Peng <pengdonglin@smartisan.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If be_hw_param_fixup is defined for BE then it will
force the BE to a specific configuration supported
by HW. In this case don't apply symmetry.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently in situations where a normal CODEC to CODEC link follows a
DPCM DAI, an error in the following form will be logged:
ASoC: can't get [playback|capture] BE for <widget name>
ASoC: no BE found for <widget name>
This happens because all widgets in a path containing a DPCM DAI will
be passed to dpcm_add_paths, which will try to interpret the CODEC<->CODEC
as if it were a DPCM DAI, in turn causing the error.
This patch aims to resolve the described issue by stopping the DPCM graph
walk, initiated from dpcm_path_get, at the first widget associated with
a DPCM BE.
Signed-off-by: Piotr Stankiewicz <piotrs@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Certain situations may warrant examining DAPM paths only to a certain
arbitrary point, as opposed to always following them to the end. For
instance, when establishing a connection between a front-end DAI link
and a back-end DAI link in a DPCM path, it does not make sense to walk
the DAPM graph beyond the first widget associated with a back-end link.
This patch introduces a mechanism which lets a user of
dai_get_connected_widgets supply a function which will be called for
every node during the graph walk. When invoked, this function can
execute arbitrary logic to decide whether the walk, given a DAPM widget
and walk direction, should be terminated at that point or continued
as normal.
Signed-off-by: Piotr Stankiewicz <piotrs@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
While performing hw_free, DPCM checks the BE state but leaves out
the suspend state. The suspend state needs to be checked as well,
as we might be suspended and then usermode closes rather than
resuming the audio stream.
This was found by a stress testing of system with playback in
loop and killed after few seconds running in background and second
script running suspend-resume test in loop
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Current DPCM doesn't copy dpcm->hw_params and doesn't call be_hw_params
if some FE are connected. But 2nd or later FE might want to know BE hw_params.
This patch solves this issue.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a device would like to use delayed suspending then PM
recommendation is to set ‘power.use_autosuspend’ flag. To allow
users to do so we need to change runtime calls in core to use
autosuspend counterparts.
For user who do not wish to use delayed suspend not setting the
device's ‘power.use_autosuspend’ flag will result in non-delayed
suspend even with these APIs which incidentally is also the default
behaviour, so only users will be impacted who opt in for this.
Signed-off-by: Sanyog Kale <sanyog.r.kale@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DPCM does not fully support symmetry attributes. soc_pcm_apply_symmetry()
is skipped in soc_pcm_open() for DPCM, without being applied elsewhere.
So HW parameters cannot be correctly limited, and user space can do
playback/capture at different rates while HW actually does not support it.
soc_pcm_params_symmetry() will return error and the second stream stops.
This patch adds soc_pcm_apply_symmetry() for FE, BE, and codec DAIs
in DPCM path that was skipped in soc_pcm_open().
Signed-off-by: PC Liao <pc.liao@mediatek.com>
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the number of DAI links is statically defined by the machine
driver at build time using an array. This makes it difficult to shrink/
grow the number of DAI links at runtime in order to reflect any changes
in topology.
We can change the DAI link array in the core to a list so that PCMs and
FE DAI links can be added and deleted at runtime to reflect changes in
use case and DSP topology. The machine driver can still register DAI links
as an array.
As the 1st step, this patch change the PCM runtime array to a list. A new
PCM runtime is added to the list when a DAI link is bound successfully.
Later patches will further implement the DAI link list.
More:
- define snd_soc_new/free_pcm_runtime() to create/free a runtime.
- define soc_add_pcm_runtime() to add a runtime to the rtd list.
- define soc_remove_pcm_runtimes() to clean up the runtime list.
- traverse the rtd list to probe the link components and dais.
- Add a field "num" to PCM runtime struct, used to specify the device
number when creating the pcm device, and for a soc card to access
its dai_props array.
- The following 3rd party machine/platform drivers iterate the rtd list
to check the runtimes:
sound/soc/intel/atom/sst-mfld-platform-pcm.c
sound/soc/intel/boards/cht_bsw_rt5645.c
sound/soc/intel/boards/cht_bsw_rt5672.c
sound/soc/intel/boards/cht_bsw_max98090_ti.c
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During suspend/resume, there is a flow that if a driver does not support
SNDRV_PCM_INFO_RESUME, it will fail at snd_pcm_resume(), and user space
can then issue SNDRV_PCM_IOCTL_PREPARE to let audio continue to play.
However, in dpcm_be_dai_prepare() it only allows BEs to be prepared
in state SND_SOC_DPCM_STATE_HW_PARAMS or SND_SOC_DPCM_STATE_STOP.
The BE state will then stay in SND_SOC_DPCM_STATE_SUSPEND, consequently
dpcm_be_dai_shutdown() is skipped in the end of playback and
be_substream->runtime is not cleared while this runtime is actually freed
by snd_pcm_detach_substream(). If another suspend comes, a NULL pointer
dereference will happen in snd_pcm_suspend() when accessing
BE substream's runtime.
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer and is slightly shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the capability to use multiple codecs on the same DAI linke where
one codec is used for playback and another one is used for capture.
Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS
microphone for capture.
No verification is made that the playback and capture codec setups are
compatible in terms of number of TDM slots (where applicable).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When running dapm_dai_get_connected_widgets() currently in
is_connected_{input,output}_ep() for each widget that gets added the array
is resized and the code also loops over all existing entries to avoid
adding a widget multiple times.
The former can be avoided by collecting the widgets in a linked list and
only once we have all widgets allocate the array.
The later can be avoided by changing when the widget is added. Currently it
is added when walking the neighbor lists of a widget. Since a widget can be
neighbors with multiple other widgets it could get added twice and hence
the check is necessary. But the main body of is_connected_{input,output}_ep
is guaranteed to be only executed at most once per widget. So adding the
widget to the list at the beginning of the function automatically makes
sure that each widget gets only added once. The only difference is that
using this method the starting point itself will also end up on the list,
but it can easily be skipped when creating the array.
Overall this reduces the code size and speeds things slightly up.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
In dpcm_get_be(), it looks for a BE rtd that has the DAI widget
according to current stream type. Only playback_widgets are searched
in the case of playback stream and vice versa. However, the DAI widget
itself can be playback or capture.
If the DAI widget is capture, but current stream type is playback,
dpcm_get_be() will always fail to find a rtd, print error messages,
and continue to the next DAI widget in list. We can just skip this
DAI widget to further suppress error messages. This happens in a
special case when 2 codecs are inter-connected, and the 1st codec's
"capture" widget is used to send data to the 2nd codec during "playback":
mtk-rt5650-rt5676 sound: ASoC: can't get playback BE for Sub AIF2 Capture
rt5650_rt5676 Playback: ASoC: no BE found for Sub AIF2 Capture
Add checks to continue to next DAI widget if current DAI widget's
direction does not match the stream type.
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current DPCM is caring only FE format. but it will be no sound
if FE/BE was below style, and user selects S24_LE format.
FE: S16_LE/S24_LE
BE: S16_LE
DPCM can rewrite the format, so basically we don't want to
constrain with the BE constraints. But sometimes it will be trouble.
This patch adds new .dpcm_merged_format on struct snd_soc_dai_link.
DPCM will use FE / BE merged format if .struct snd_soc_dai_link
has it. We can have other .dpcm_merged_xxx in the future
.dpcm_merged_foramt
.dpcm_merged_rate
.dpcm_merged_chan
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the registration of a debugfs directory fails this is treated as a
non-fatal error in ASoC and operation continues as normal. This means we
need to be careful and check if the parent debugfs directory exists if we
try to register a debugfs file or sub-directory. Otherwise we might end up
passing NULL for the parent and the file or directory will be registered in
the top-level debugfs directory.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Failing to register the debugfs entries is not fatal and will not affect
normal operation of the sound card. Don't abort the card registration if
soc_dpcm_debugfs_add() fails.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA core with commit 257f8cce5d - "ALSA: pcm: Allow nonatomic trigger
operations" allows trigger ops to implemented as nonatomic. For ASoC, we can
specify this in dailinks and is updated while snd_pcm is created
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the new wildcard msbits constraints instead of installing a constraint
for each available sample format width.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both codec and cpu DAI prepare print the same error message making it a bit
more difficult to grep quickly from sources. Fix this by telling it
explicitly.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Do not free BE hw if it's still used by other FE during dpcm runtime
shutdown. Otherwise the BE runtime state will be STATE_HW_FREE and
won't be updated to STATE_CLOSE when shutdown ends, because BE dai
shutdown function won't close pcm when detecting BE is still under
use. With STATE_HW_FREE, BE can't be triggered start again.
This corner case can easily appear when one BE is used by two FE,
without this patch "ASoC: dpcm: Fix race between FE/BE updates and
trigger"(ea9d0d771f). One FE tries to
shutdown but it's raced against xrun on another FE. It improves the
be dai hw_free logic.
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Skip dpcm path checking for playback or capture, if corresponding FE
doesn't support playback or capture, or currently is not ready. It
can reduce the unnecessary cost to search connected widgets.
[Tweaked comments for clarity -- broonie]
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DPCM can update the FE/BE connection states totally asynchronously
from the FE's PCM state. Most of FE/BE state changes are protected by
mutex, so that they won't race, but there are still some actions that
are uncovered. For example, suppose to switch a BE while a FE's
stream is running. This would call soc_dpcm_runtime_update(), which
sets FE's runtime_update flag, then sets up and starts BEs, and clears
FE's runtime_update flag again.
When a device emits XRUN during this operation, the PCM core triggers
snd_pcm_stop(XRUN). Since the trigger action is an atomic ops, this
isn't blocked by the mutex, thus it kicks off DPCM's trigger action.
It eventually updates and clears FE's runtime_update flag while
soc_dpcm_runtime_update() is running concurrently, and it results in
confusion.
Usually, for avoiding such a race, we take a lock. There is a PCM
stream lock for that purpose. However, as already mentioned, the
trigger action is atomic, and we can't take the lock for the whole
soc_dpcm_runtime_update() or other operations that include the lengthy
jobs like hw_params or prepare.
This patch provides an alternative solution. This adds a way to defer
the conflicting trigger callback to be executed at the end of FE/BE
state changes. For doing it, two things are introduced:
- Each runtime_update state change of FEs is protected via PCM stream
lock.
- The FE's trigger callback checks the runtime_update flag. If it's
not set, the trigger action is executed there. If set, mark the
pending trigger action and returns immediately.
- At the exit of runtime_update state change, it checks whether the
pending trigger is present. If yes, it executes the trigger action
at this point.
Reported-and-tested-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
We call mute for codec dai only, we should call this for cpu dai as well to
allow cpu dais (FEs) in DSPs to be muted/unmuted on shutdown/startup
Signed-off-by: Ramesh Babu <ramesh.babu@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In the SNDRV_PCM_STREAM_CAPTURE branch in soc_pcm_apply_msb(), look at
sig_bits of the capture stream, not the playback one.
Spotted by coverity.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
dpcm_path_get may return -ENOMEM when allocating memory for list
fails. We should not keep processing path or start up dpcm dai in
this case.
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a function helper to factorize the hw_params code.
Suggested by Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Refactor the function to facilitate the migration to
multiple codecs.
Fix a trailing space in the header as well.
No functional change.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The component struct already has a name and id field which are initialized to
the same values as the same fields in the CODEC and platform structs. So remove
them from the CODEC and platform structs and used the ones from the component
struct instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
we need to release dapm widget list after dpcm_path_get in
soc_dpcm_runtime_update. otherwise, there will be potential memory
leak. add dpcm_path_put to fix it.
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
This was initially removed in commit 6423c1875 ("ASoC: Remove runtime field from
DAI"), but was, presumably by accident, brought back in commit f0fba2ad1 ("ASoC:
multi-component - ASoC Multi-Component Support"). But has never been
initialized to anything but NULL ever since. This commit removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The regular state before we execute SNDRV_PCM_TRIGGER_SUSPEND should be
SNDRV_PCM_TRIGGER_START, not SNDRV_PCM_TRIGGER_STOP. Thus fix it.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Machine specific trigger callback allows to do final stream start/stop
related operations in a machine driver after setting up the codec, DMA and
DAI.
One example could be clock management for linked streams case where machine
driver can start/stop synchronously the linked streams.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Stefan Roese <sr@denx.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replace rtd_get_codec_widget() and rtd_get_cpu_widget() by a simple
dai_get_widget() in preparation for DAI-multicodec support, per Lars
suggestion.
No functional change.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:
- Lots of cleanups from Takashi for enumerations; the original API for
these was error prone so he's refactored lots of code to use more
modern APIs which avoid issues.
- Elimination of the ASoC level wrappers for I2C and SPI moving us
closer to converting to regmap completely and avoiding some
randconfig hassle.
- Provide both manually and transparently locked DAPM APIs rather than
a mix of the two fixing some concurrency issues.
- Start converting CODEC drivers to use separate bus interface drivers
rather than having them all in one file helping avoid dependency
issues.
- DPCM support for Intel Haswell and Bay Trail platforms.
- Lots of work on improvements for simple-card, DaVinci and the Renesas
rcar drivers.
- New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
CSR SiRF SoC.
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Merge tag 'asoc-v3.15' into asoc-next
ASoC: Updates for v3.15
Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:
- Lots of cleanups from Takashi for enumerations; the original API for
these was error prone so he's refactored lots of code to use more
modern APIs which avoid issues.
- Elimination of the ASoC level wrappers for I2C and SPI moving us
closer to converting to regmap completely and avoiding some
randconfig hassle.
- Provide both manually and transparently locked DAPM APIs rather than
a mix of the two fixing some concurrency issues.
- Start converting CODEC drivers to use separate bus interface drivers
rather than having them all in one file helping avoid dependency
issues.
- DPCM support for Intel Haswell and Bay Trail platforms.
- Lots of work on improvements for simple-card, DaVinci and the Renesas
rcar drivers.
- New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
CSR SiRF SoC.
# gpg: Signature made Wed 12 Mar 2014 23:05:45 GMT using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
The changes in "ASoC: pcm: free path list before exiting from error
conditions" actually introduced both double frees (in case where the
path list was allocated but empty) and frees of unallocated memory (in
cases where the error being handled was -ENOMEM. Drop the commit for
now.
Fixes: e4ad1accb (ASoC: pcm: free path list before exiting from error conditions)
Reported-by: Ben Hutchings <ben@decadent.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
In preparation for componentization move the ignore_pmdown_time field from the
snd_soc_codec struct to the snd_soc_component struct. Set it to true for non
CODEC components for now.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is no reason why active count tracking should only be done for CODECs but
not for other components. Moving the active count from the snd_soc_codec struct
to the snd_soc_component struct reduces the differences between CODECs and other
components and will eventually allow component to component DAI links (Which is
a prerequisite for converting CODECs to components).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
For CODEC to CODEC links we need to make sure to also manage the 'active' field
of the cpu_dai CODEC.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have the same code that increments and decrements the active field of the
various PCM runtime components (all with the same bugs). Factor this out into
common helper functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
For CODEC to CODEC links we should only immediately power down if both CODECs
are configured to ignore the power down delay. Factor the logic for this
into a helper function that can be used for both compressed and normal PCMs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
dpcm_path_get() allocates dynamic memory to hold path list.
Corresponding dpcm_path_put() must be called to free the memory.
dpcm_path_put() is not called under several error conditions.
This leads to memory leak.
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
The ASoC compressed code needs to call the internal DPCM APIs in order to
dynamically route compressed data to different DAIs.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Instead of open-coding the intersecting of two rate masks (and getting slightly
wrong for some of the corner cases) use the new snd_pcm_rate_mask_intersect()
helper function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
If none of the components (CODEC or CPU DAI) sets a maximum sample rate we'll
end up with the rate_max field of the runtime hardware set to 0. (Note that it
is still possible for the components to constrain the supported sample rates
using other methods, e.g. setting a list constraint) If rate_max is 0 this means
that the sound card doesn't support any rates at all, which is not the desired
result. So initialize rate_max to UINT_MAX. For symmetry reasons also set
rate_min to 0.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some DMA cores might add additional restrictions on which in memory audio
formats can be supported by the compound sound card. If the PCM component
specifies a set of formats it supports (by setting the formats field to non 0)
take these into account when calculating the format set for the compound sound
card.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some BE DAIs can be "dummy" (when the DSP is controlling the DAI) and as such
wont have set a minimum number of playback or capture channels required for BE
DAI registration (to establish supported stream directions).
Force machine drivers to explicitly set whether they support playback and capture
stream directions for every BE DAIs.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the platform driver has no ops, the platform function
bespoke_trigger() is no more called.
The problem was introduced by the commit c5914b0aae
"ASoC: pcm: Check for ops before deferencing them"
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
Allow PCMs that do not impose any restrictions on the supported formats to set
the formats field to 0, Instead of assuming that this means that the PCM does
not support any formats (which doesn't make much sense), assume that it supports
all formats. This brings the behavior of DPCM closer to that of non-DPCM.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have the same code for initializing the runtime pcm on both the playback and
the capture path. Factor this out into a common helper function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The snd_soc_dai_digital_mute() here will be never executed because we only
decrease codec->active in snd_soc_close(). Thus correct it.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch removed the redundant snd_soc_dai_digital_mute() in close() since
it's better to mute in hw_free() which's slightly earlier and symmetrical for
the case of reconfiguration: 'aplay 44k1.wav 48k.wav', for example, will be
open()->hw_params()->prepare(unmute)->playi1ng->hw_free(mute)->hw_params()->
parepare(unmute)->playing->hw_free(mute)->close()
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
If there are symmetry constraints between the playback and the capture channel
set the SNDRV_PCM_INFO_JOINT_DUPLEX flag to let userspace know about this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_pcm_limit_hw_rates() will initialize the minimum and maximum sample rate for
the PCM stream based on the rates specified in the rates field. Since we call
snd_pcm_limit_hw_rates() after soc_pcm_init_runtime_hw() it will essentially
overwrite the min and max rate set in soc_pcm_init_runtime_hw(). This may cause
the minimum or maximum rate to be set to a value outside the range of one of the
components if one of the components sets either SNDRV_PCM_RATE_CONTINUOUS or
SNDRV_PCM_RATE_KNOT and the other component specified a discrete rate via
SNDRV_PCM_RATE_[0-9]* that is outside of the first component's rate range. To
fix this first calculate the minimum and maximum rates using
snd_pcm_limit_hw_rates() and then on top of that apply the contraints specified
in the snd_soc_pcm_stream structs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Takashi iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In order to make sure that the sample rate is in the supported range of both
components the maximum rate of the card should be the minimum of the maximum
rate of each components. There is one special case to consider though, if
max_rate is set to 0 this means there is no maximum specified, so use
min_not_zero() macro which will give use the desired result.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Takashi iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We're now applying soc_hw_params_symmetry() to reject unmatched parameters
while we clear parameters in soc_pcm_close(). So here's a use case might be
broken by this mechanism: aplay -Dhw:0 44100.wav 48000.wav 32000.wav
In this case, we call soc_pcm_open()->soc_pcm_hw_params()->soc_pcm_hw_free()
->soc_pcm_hw_params()->soc_pcm_hw_free()->soc_pcm_close() in order. As we
only clear parameters in soc_pcm_close(). The parameters would be remained
in the system even if the playback of 44100.wav is finished.
Thus, this patch is trying to move parameters cleaning into hw_free() so that
the system can continue to serve this kind of use case.
Also, since we set them in hw_params(), it should be better to clear them in
hw_free() for symmetry.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some SoCs can only work in mono or stereo mode at one time. So if
we let them capture a mono stream while playing a stereo stream,
there might be a problem occur to one of these two streams: double
paced or slowed down.
In soc-pcm.c, we have soc_pcm_apply_symmetry() to apply the rate
symmetry. But we don't have one for channels.
Likewise, we can treat symmetric_rate as a solution for those SoCs
or CODECs which can not handle asymmetrical LRCLK. But it's also
impossible for them to handle asymmetrical BCLK. And accodring to
BCLK = LRCLK * channel number * slot size(fixed or sample bits),
sample bits might also be a problem if they are not using a fixed
slot size.
Thus, this patch applys symmetry for channels and sample bits.
Meanwhile, there might be a race between two substreams if starting
simultaneously. Previously, we only added warning to compalin but
still using conservative way to let it carry on. However, this patch
rejects the second stream with any unmatched parameter to make sure
the first existing stream won't be broken.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
It's quite popular that more drivers are using pinctrl PM, for example:
(Documentation/devicetree/bindings/arm/primecell.txt). Just like what
runtime PM does, it would deactivate and activate pin group depending
on whether it's being used or not.
And this pinctrl PM might be also beneficial to cpu dai drivers because
they might have actual pinctrl so as to sleep their pins and wake them
up as needed.
To achieve this goal, this patch sets pins to the default state during
resume or startup; While during suspend and shutdown, it would set pins
to the sleep state.
As pinctrl PM would return zero if there is no such pinctrl sleep state
settings, this patch would not break current ASoC subsystem directly.
[ However, there is still an exception that the patch can not handle,
that is, when cpu dai driver does not have pinctrl property but another
device has it. (The AUDMUX <-> SSI on Freescale i.MX6 series for example.
SSI as a cpu dai doesn't contain pinctrl property while AUDMUX, an Audio
Multiplexer, has it). In this case, this kind of cpu dai driver needs to
find a way to obtain the pinctrl property as its own, by moving property
from AUDMUX to SSI, or creating a pins link/dependency between these two
devices, or using a more decent way after we figure it out. ]
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Avoid oopsing if there is no backend stream associated with a front end
stream.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
Ensure that we always check that an ops structure is present before we
try to use it, improving the robustness of the system.
Reported-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
dev_ prints are already prefixed by ": " before format string so there is no
need for extra spaces.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add 'playback_only' and 'capture_only' fields that can be used for specifying
that a dai_link has a unidirectional capability.
The motivation for this is for the cases of systems, such as Freescale MX28,
that has two unidirectional DAIs.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
soc_dpcm_runtime_update() operates on a ASoC card as a whole. Currently it takes
a snd_soc_dapm_widget as its only parameter though. The widget is then used to
look up the card and is otherwise unused. This patch changes the function to
take a pointer to the card directly. This makes it possible to to call
soc_dpcm_runtime_update() for updates which are not related to one specific
widget.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is no need to use a normal per-CPU workqueue for delayed power downs
as they're not timing or performance critical and waking up a core for them
would defeat some of the point.
Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Viresh Kumar <viresh.kumar@linaro.org>
Even though they are virtual widgets DAI widgets still get counted for the
DAPM context power management so we can't just use the active state to
check if they should be powered as they may not be part of a complete path.
Instead split them into input and output widgets and do the same power
checks as we perform on AIFs.
Reported-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When declaring playback and capture capabilities check for both CODEC
side and CPU side support rather than only checking for CODEC side
support. While it is unusual some CPUs do have unidirectional DAIs.
Reported-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We use the same code to initialize the runtime pcm based on the
snd_soc_pcm_stream struct on both the playback and capture path. Factor this
code into a helper function to make things a bit more tidy.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_set_runtime_hwparams() is the only PCM related function that lives in
soc-core.c. All other PCM related functions live in soc-pcm.c, so move
snd_soc_set_runtime_hwparams() over as well for a bit more consistency.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Help avoid noise from the power up of the capture path propagating through
into the start of the recording (especially noise caused by the ramp of
microphone biases) by keeping the capture muted until after we've finished
powering things up with DAPM in the same manner we do for playback. This
allows us to take advantage of soft mute support in the hardware more
effectively and is more consistent.
The core code using the existing digital mute operation is updated to take
advantage of this. Some additional cases in the soc-pcm code and suspend
will need separate handling but these are less practically relevant than
the main runtime stream start/stop case.
Rather than refactor the digital mute function in every single driver a
new operation is added for drivers taking advantage of this functionality,
the old operation should be phased out over time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
I've removed several unreachable returns.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When front-end PCM session is in paused state, back-end
PCM session will be put in paused state as well if given
front-end PCM session is the only client of given back-end.
Then, application closes front-end PCM session, DPCM
framework will not allow back-end enters HW_FREE state
so back-end will never get shutdown completely.
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
pop_wait is used to determine if a deferred playback close
needs to be cancelled when the a PCM is open or if after
the power-down delay expires it needs to run. pop_wait is
associated with the CODEC DAI, so the CODEC DAI must be
unique. This holds true for most CODECs, except for the
dummy CODEC and its DAI.
In DAI links with non-unique dummy CODECs (e.g. front-ends),
pop_wait can be overwritten by another DAI link using also a
dummy CODEC. Failure to cancel a deferred close can cause
mute due to the DAPM STOP event sent in the deferred work.
One scenario where pop_wait is overwritten and causing mute
is below (where hw:0,0 and hw:0,1 are two front-ends with
default pmdown_time = 5 secs):
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1
sleep 1
aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 &
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE
Since CODECs may not be unique, pop_wait is moved to the PCM
runtime structure. Creating separate dummy CODECs for each
DAI link can also solve the problem, but at this point it's
only pop_wait variable in the CODEC DAI that has negative
effects by not being unique.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC has a mixture of message prefixes e.g. "ASoC", "asoc"
or none and message types e.g. pr_debug or dev_dbg.
Make sure all ASoC core messages use the same "ASoC" prefix and
convert any component device specific messages to use dev_dbg
instead of pr_debug.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some userspace will open a PCM device and then configure mixers
for routing before triggering. This patch allows userspace to do
this sequence.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use dev_ style logging throughout soc_new_pcm()
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure that the dpcm_get_be() only returns BE DAI links.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
They pollute the global namespace and cause sparse to complain.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When we instantiate an aux_dev we use a fake rtd as part of the process
which doesn't have a dai_link associated with it. Fix the dpcm startup
code to cope with this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Remove writable debugFS permission, use simple_open() and
fix indentation.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide an ioctl marshaller for ASoC platform drivers.
This will use the default ALSA handler if no platform
handler exists.
This is also required for DPCM BE PCMs as snd_pcm_info()
will call the ioctl as part of stream startup.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some on SoC DSP HW is very tightly coupled with DMA and DAI drivers. It's
necessary to allow some flexability wrt to PCM operations here so that we
can define a bespoke DPCM trigger() PCM operation for such HW.
A bespoke DPCM trigger() allows exact ordering and timing of component
triggering by allowing a component driver to manage the final enable
and disable configurations without adding extra complexity to other
component drivers. e.g. The McPDM DAI and ABE are tightly coupled on
OMAP4 so we have a bespoke trigger to manage the trigger to improve
performance and reduce complexity when triggering new McPDM BEs.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch allows DPCM to dynamically alter the FE to BE PCM links
at runtime based on mixer setting updates. DAPM is looked up after
every mixer update and we perform a DPCM runtime update if the
mixer has a change of value.
This patchs adds/changes the following :-
o Adds DPCM runtime update core.
o Changes soc_dapm_mixer_update_power() and soc_dapm_mux_update_power()
to return if a change has occured rather than 0. No other users check
atm.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add debugFS files for DPCM link management information.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Dynamic PCM core allows digital audio data to be dynamically
routed between different ALSA PCMs and DAI links on SoC CPUs with
on chip DSP devices. e.g. audio data could be played on pcm:0,0 and
routed to any (or all) SoC DAI links.
Dynamic PCM introduces the concept of Front End (FE) PCMs and Back
End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that
they can dynamically route digital audio data to any supported BE
PCM. A BE PCM has no ALSA device, but represents a DAI link and it's
substream and audio HW parameters.
e.g. pcm:0,0 routing digital data to 2 external codecs.
FE pcm:0,0 ----> BE (McBSP.0) ----> CODEC 0
+--> BE (McPDM.0) ----> CODEC 1
e.g. pcm:0,0 and pcm:0,1 routing digital data to 1 external codec.
FE pcm:0,0 ---
+--> BE (McBSP.0) ----> CODEC
FE pcm:0,1 ---
The digital audio routing is controlled by the usual ALSA method
of mixer kcontrols. Dynamic PCM uses a DAPM graph to work out the
routing based upon the mixer settings and configures the BE PCMs
based on routing and the FE HW params.
DPCM is designed so that most ASoC component drivers will need no
modification at all. It's intended that existing CODEC, DAI and
platform drivers can be used in DPCM based audio devices without
any changes. However, there will be some cases where minor changes
are required (e.g. for very tightly coupled HW) and there are
helpers to support this too.
Somethimes the HW params of a FE and BE do not match or are
incompatible, so in these cases the machine driver can reconfigure
any hw_params and make any DSP perform sample rate / format conversion.
This patch adds the core DPCM code and contains :-
o The FE and BE PCM operations.
o FE and BE DAI link support.
o FE and BE PCM creation.
o BE support API.
o BE and FE link management.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently stream events are only perfomed on codec stream widgets only.
There is now a need to be able to perform stream events on platform
widgets too.
e.g. we have the ABE platform driver with several DAI links
to dummy codecs. We need to be able to perform stream events on any
of the dummy codec DAI links.
This patch also removes the snd_soc_dai * parameter since it's already
contained within the rtd * parameter.
Finally makle stream event return void since no one checks it anyway.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to allow us to do something smarter than iterate through widgets
doing strcmp() to work out what to power up for stream events change the
interface used to generate them to be based on the combination of a DAI
and a stream direction rather than just a simple string identifying the
stream.
At some point we'll probably want a set of channels too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Since we've already got logic to special case immediate teardown of the
stream we may as well use it if the pmdown_time has been set to zero by
the application layer instead of scheduling a work item with zero delay.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Use the standard logging macros and use dev_ variants where we can, also
reporting error codes whenever we report an error. These changes (the
error codes in particular) make it noticeably easier to figure out what
went wrong just from the basic dmesg output.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
As per discussion we can safely ignore the 8 and 16 bit sample
sizes when applying the msbits constraint.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Most devices accept data in formats that don't correspond directly to
their internal format. ALSA allows us to set a msbits constraint which
tells userspace about this in case it finds it useful (for example, in
order to avoid wasting effort dithering bits that will be ignored when
raising the sample size of data) so provide a mechanism for drivers to
specify the number of bits that are actually significant on a DAI and
add the appropriate constraints along with all the others.
This is done slightly awkwardly as the constraint is specified per sample
size - we loop over every possible sample size, including ones that the
device doesn't support and including ones that have fewer bits than are
actually used, but this is harmless as the upper layers do the right thing
in these cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The original code does not cover the case that two DAIs(CPU) have different
ASoC core PCM operations(like mmap, pointer...). Currently we have only one
global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different
pointer functions, second DAI's pointer function is set for both first DAI
and second DAI in case of original code.
This patch uses runtime's pcm_ops instead of global pcm_ops for each DAIs. So
each DAIs can have different ASoC core PCM operations. This is needed to
support multiple DAIs.
Signed-off-by: Sangsu Park <sangsu4u.park@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Every device that implements runtime power management for DAIs is doing
it in pretty much the same way: in the startup callback they take a
runtime PM reference and then in the shutdown callback they release that
reference, keeping the device active while the DAI is active. Given the
frequency with which this is done and the obviousness of the need to keep
the device active in this period factor the code out into the core, taking
references on the device for each CPU DAI, CODEC DAI and DMA device in the
core.
As runtime PM is reference counted this shouldn't interfere with any
other reference holding by the drivers, and since (in common with the
existing implementations) we don't check for errors on enabling it
shouldn't matter if the device actually has runtime PM enabled or not.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
There's no point in adding unlikely() annotations outside of hot paths
and on systems using these features the annotation will always be wrong
(as opposed to being something that only comes up once in a while) so
the annotation may even be harmful.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
