This commit adds common PCM constraints according to current firewire-lib
implementation.
1.Maximum width for each sample is limited by 24.
AM824 in IEC 61883-6 can deliver 24bit data.
2. Minimum time for period is 5msec.
Apply the old value. For shorter latency, further works are needed.
3. In blocking mode, frames per period/buffer is aligned to 32.
Each packet can include some frames depending on its sampling rate. In
blocking mode, the number equals to SYT_INTERVAL. Currently firewire-lib
can schedule snd_pcm_period_elapsed() for each packet. So, for accurate
PCM interrupt, the number of frames per period/buffer should be aligned
to SYT_INTERVAL.
Currently firewire-lib is lack of better rules to achieve this. So LCM of
each value of SYT_INTERVALs (=32) is applied. This can be improved for
further work.
[Fixed the compile error due to the missing "&" by tiwai]
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, AMDTP functionality in firewire-lib supports mapping
for PCM data channels. With this mapping, firewire-lib can obsolete
a flag, CIP_HI_DUALWIRE, but Dice driver still keeps dual wire mode.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices arrange the position of PCM/MIDI data channel in AMDTP packet.
This commit allows drivers to set channel mapping.
To be simple, the mapping table is an array with fixed length. Then the number
of channels for PCM is restricted by 64 channels.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Generally, the devices can synchronize to handle 'presentation timestamp'
in CIP packets. This commit adds functionality to pick up this timestamp from
in-packets transmitted by the device, then use it for out packets.
In current implementation, this module generated the timestamp by itself. This
is 'SYT Match' mode. Then drivers with this module acts as synchronization
master. This commit allows this module to act as synchronization slave.
This commit restricts this mechanism is only available in blocking mode because
handling the timestamp in non-blocking mode is more complicated than in
blocking mode.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For duplex streams with synchronization, drivers should pick up
'presentation timestamp' from in-packets and use the timestamp for
out-packets. This commit is preparation for this.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For capturing/playbacking MIDI messages, this commit adds one MIDI conformant
data channel. This data channel has multiplexed 8 MIDI data streams. So this
data channel can transfer messages from/to 8 MIDI ports.
And this commit allows to set PCM format even if AMDTP streams already start.
I suppose the case that PCM substreams are going to be joined into AMDTP
streams when AMDTP streams are already started for MIDI substreams. Each
driver must count how many PCM/MIDI substreams use AMDTP streams to stop
AMDTP streams.
There are differences between specifications about MIDI conformant data.
About the multiplexing, IEC 61883-6:2002, itself, has no information. It
describes labels and bytes for MIDI messages and refers to MMA/AMEI RP-027
for 'successfull implementation'. MMA/AMEI RP-027 describes 8 MPX-MIDI data
streams for one MIDI conformant data channel. IEC 61883-6:2005 adds
'sequence multiplexing' and apply this way and describe incompatibility
between 2002 and 2005.
So this commit applies IEC 61883-6:2005. When we find some devices compliant
to IEC 61883-6:2002, then this difference should be handles as device quirk
in additional work.
About the number of bytes in an MIDI conformant data, IEC 61883-6:2002 describe
0,1,2,3 bytes. MMA/AMEI RP-027 describes 'MIDI1.0-1x-SPEED', 'MIDI1.0-2x-SPEED',
'MIDI1.0-3x-SPEED' modes and the maximum bytes for each mode corresponds to 1,
2, 3 bytes. The 'MIDI1.0-2x/3x-SPEED' modes are accompanied with 'negotiation
procedure' and 'encapsulation details' but there is no specifications for them.
So this commit implements 'MIDI1.0-1x-SPEED' mode for playback, but allows
to pick up 1-3 bytes for capturing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For capturing PCM, this commit adds the functionality to handle in-stream.
This is also applied for dual-wire mode.
Currently, capturing 32bit samples are supported.
When the sequence of in-packet has discontinuity of dbc, in-stream isn't handled
and amdtp_streaming_error() returns true.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some codes can be reused to handle in-stream. This commit adds new functions.
This commit also renames some functions to keep naming consistency.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds 'direction' member to amdtp_stream structure to indicate its
direction. This patch also adds 'direction' argument to amdtp_stream_init()
function to determine its direction.
The amdtp_stream_init() function is exported and used by firewire-speakers and
dice so this patch also affects them.
This patch just add them. Actual implementation will be done by followed
patches.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds some macros instead of fixed value for AMDTP according to
IEC 61883-1/6. These macros will also be used by followed patches.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch renames some functions, a structure and its member to reuse them
in both AMDTP in/out stream.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just two members in the alc269_pin_fixup_tbl[] can cover more than
10 Dell laptop models.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A lot a machine have the same codec, but they have different default
pinconf setting just because the def association and sequence is
different, as a result they can't share a hda_pintbl[], to overcome
it, we don't compare def association and sequence in the pinconf
matching.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Normally, we match on pci ssid only. This works but needs new code
for every machine. To catch more machines in the same quirk, let's add
a new type of quirk, where we match on
1) PCI Subvendor ID (i e, not device, just vendor)
2) Codec ID
3) Pin configuration default
If all these three match, we could be reasonably certain that the
quirk should apply to the machine even though it might not be the
exact same device.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "fixup_forced" flag will indicate whether a specific fixup
(or nofixup) has been set by the user, to override the driver's
default.
This flag will help future patches.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If kmalloc() fails, warnings will be loud enough. We can safely just
return -ENOMEM in such cases.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent Intel H97/Z97 chipsets need the similar setups like other
Intel chipsets for snooping, etc. Especially without snooping, the
audio playback stutters or gets corrupted. This fix patch just adds
the corresponding PCI ID entry with the proper flags.
Reported-and-tested-by: Arthur Borsboom <arthurborsboom@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ensure i2s->op_clk is not used when clk_get() for this clock fails.
This prevents working with an incorrectly configured clock in some
conditions.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Lots of cleanup work going on in the core this release but very little
visible to external users except for the new drivers that have been
added.
- Support for specifying aux CODECs in DT.
- Removal of the deprecated mux and enum macros.
- More moves towards full componentisation.
- Removal of some unused I/O code.
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers.
- Several drivers exposed directly in Kconfig for use with simple-card.
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
ST STA350.
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Merge tag 'asoc-v3.16' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.16
Lots of cleanup work going on in the core this release but very little
visible to external users except for the new drivers that have been
added.
- Support for specifying aux CODECs in DT.
- Removal of the deprecated mux and enum macros.
- More moves towards full componentisation.
- Removal of some unused I/O code.
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers.
- Several drivers exposed directly in Kconfig for use with simple-card.
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
ST STA350.
We should prefer `const struct pci_device_id` over
`DEFINE_PCI_DEVICE_TABLE` to meet kernel coding style guidelines.
This issue was reported by checkpatch.
A simplified version of the semantic patch that makes this change is as
follows (http://coccinelle.lip6.fr/):
// <smpl>
@@
identifier i;
declarer name DEFINE_PCI_DEVICE_TABLE;
initializer z;
@@
- DEFINE_PCI_DEVICE_TABLE(i)
+ const struct pci_device_id i[]
= z;
// </smpl>
It has been tested by compilation.
Signed-off-by: Benoit Taine <benoit.taine@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
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Merge tag 'asoc-v3.15-rc5-intel' into asoc-linus
ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
# gpg: Signature made Wed 14 May 2014 12:40:27 BST 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>"
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
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Merge tag 'asoc-v3.15-rc5-drivers' into asoc-linus
ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
# gpg: Signature made Wed 14 May 2014 12:49:57 BST 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>"
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
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Merge tag 'asoc-v3.15-rc5-core' into asoc-linus
ASoC: Core fixes for v3.15
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
# gpg: Signature made Wed 14 May 2014 12:59:19 BST 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>"
clk_get in probe function can be safely replaced with devm_clk_get.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replaced snd_soc_register_component with its devres equivalent,
devm_snd_soc_register_component.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replaced snd_soc_register_platform with devm_snd_soc_register_platform
in samsung_asoc_dma_platform_register(). This makes the function
samsung_asoc_dma_platform_unregister() redundant. This is removed and
all its users are updated.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replace snd_soc_register_card with devm_snd_soc_register_card.
With this change, we can delete the empty remove functions.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The analog-loopback causes the speaker noises even if it's set to zero
volume. As a simple workaround, just get rid of the loopback mixer.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=873704
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Disable the AA-loopback path to get rid of the constant white noise
that can be heard when headphones are used.
Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DAPM routes for this board are complete, hence we can let the core take care
of disconnecting non-connected pins rather than doing it manually.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for setting source clock of the "Core CLK"
of the IIS Multi Audio Interface.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As we are moving the mmp platform towards multiplatform support,
we have to stop including platform header files.
This changes the pxa-ssp sound driver file to no longer depend
on mach/hardware.h and mach/dma.h. The code using the definitions
from those headers is actually gone already, the only thing
that was still being used was the pxa_dma_desc typedef, which
we can easily work around by using the normal 'struct pxa_dma_desc'
name.
The pxa2xx-dma driver still uses this header, so we include it
explicitly there, which is ok because that is only used on pxa,
not on mmp.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let the ALC5623 codec be instantiated from DT. Add a simple binding
for the additional control register and the jack detect register.
Also, add a prompt to the Kconfig entry for this CODEC, so that it can
be selected. Since kirkwood-t5325.c will no longer be used, we need to
be able to enable the CODEC in the mvebu_v5_defconfig etc.
Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Acked-by: Jason Cooper <jason@lakedaemon.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
fsl_ssi_set_dai_sysclk will be called from fsl_ssi_hw_params in the
next patch. Move up to avoid forward declaration and to keep the next patch
more readable. No functional change.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the fsl-ssi unit is used in i2s slave mode, it is possible that the
SSI unit starts transmitting data on the wrong channel. This happens
because the SSI does not synchronize with the left-right-clock by
default.
This patch enables transmit enable synchronization.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There are some variables defined in struct fsl_ssi_private that describe
states that are also described by other variables.
This patch adds some helper functions that return exactly the same
information based on available variables. This helps to clean up struct
fsl_ssi_private and remove them from the probe function.
It also removes some not really used variables (new_binding, name).
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reorder the probe function to be able to move the second imx-specific
block to the seperate imx probe function. The patch also removes some
comments/variables/code that are not used anymore or could be simply
replaced by other variables.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Simplify dma DT property handling. fsl,ssi-dma-events is not used
anymore. It passes invalid data to imx_pcm_dma_params_init_data() which
copies some data into an imx dma struct. This struct is never used in
imx-dma or imx-sdma because of generic OF DMA handling. The
"fsl,ssi-dma-events" is not used anywhere in dts files.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move imx specific probe code to a seperate function. It reduces the
size of the probe() function and makes the code and error handling
easier to understand.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Instead of creating a name using string manipulation functions, we can
simply use the device name for the DAI driver struct.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move all code that is only used for debugging to a seperate file. This
makes it easier to see what functions are used for debugging only.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The bits we have to clear when disabling are different when the other
stream is still active.
This patch fixes the calculation of new register values after disabling
one stream. It also adds a more detailed description of the new register
value calculation.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using a proper do-while loop here instead of a open-coded goto loop is both
cleaner and shorter.
Also fixes the following warnings from smatch:
sound/soc/codecs/ad1980.c:213 ad1980_reset() info: loop could be replaced with if statement.
sound/soc/codecs/ad1980.c:212 ad1980_reset() info: ignoring unreachable code.
sound/soc/codecs/ad1980.c:215 ad1980_reset() info: ignoring unreachable code.
While we are at it also change retry_cnt to unsigned int, using u16 for a
on-stack loop counter doesn't make that much sense.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
I missed a rename during the review process. Fix the
MODULE_DEVICE_TABLE to match the structure.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a driver for the HDA block in Tegra SoCs. The HDA bus is
used to communicate with the HDMI codec on Tegra124.
Most of the code is re-used from the Intel/PCI HDA driver. It brings
over only two of the module params, power_save and probe_mask.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In some cases the pcm stream is closed while context has been
scheduled to be restored, causing a null pointer deref panic.
Cancel work to ensure stream does not get freed while work is
still active/pending.
Also, restoring the pcm context can be safely skipped after the
stream has been stopped. Check if pcm stream is still running
before restoring stream context to help pending work finish
more quickly in stream close path.
Signed-off-by: Kevin Strasser <kevin.strasser@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This looks like a copy-paste bug, fix it.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ALSA SoC core marks widgets as connected by default when they are
initialized in snd_soc_dapm_new_control() so there is no need to call
snd_soc_dapm_enable_pin() from machine driver init functions.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ALSA SoC core takes care of calling snd_soc_dapm_sync() at the end
snd_soc_instantiate_card() so there is no need to call it from machine
driver init functions.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
rtd->codec does not necessarily point to the CODEC instance for which the
callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use
dai->codec instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
rtd->codec does not necessarily point to the CODEC instance for which the
callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use
dai->codec instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Just move field descriptions to the struct description in the kernel-doc
format. There is no functional change.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently snd_dmaengine_pcm_trigger() calls dmaengine_pause()
unconditinally during device suspend. In case where DMA controller
doesn't support PAUSE/RESUME functionality, this call is not able
to stop the DMA controller. In this scenario, audio playback doesn't
resume after device resume.
Calling dmaengine_pause/dmaengine_terminate_all conditionally fixes
the issue.
It has been tested with audio playback on Samsung platform having
PL330 DMA controller which doesn't support PAUSE/RESUME.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Delete unnecessary local variable whose value is always 0 and that hides
the fact that the result is always 0.
A simplified version of the semantic patch that fixes this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@r exists@
local idexpression ret;
expression e;
position p;
@@
-ret = 0;
... when != ret = e
return
- ret
+ 0
;
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Microphone bias level configuration register can configure voltage between
2.2 V and 2.8 V but doesn't manage is voltage on or off. Microphone bias
on/off state is controlled by "MICBIAS" DAPM widget.
Therefore there is no need to update bias voltage conditionally depending on
jack state each time when codec goes to SND_SOC_BIAS_ON state and setting
can be moved to max98090_probe() as driver currently doesn't support other
levels than 2.8 V.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
BCL is being configured in two places producing a warning message.
Make sure we only configure BCLK once and when we are master.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add ACPI ID for MAX98090 and ACPI 5 I2C device probing support.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure the cache is fully flushed at resume time.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure we reset codec and clear any IRQs on resume. This matches
the init sequence in probe.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reset needs to wait 20ms before other codec IO is performed. This wait
was not being performed. Fix this by making sure the reset register is not
restored with the cache, but use the manual reset method in resume with
the wait.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
TLV for digital sidetone volume is wrong, this fix matches it to the
datasheet.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As we already have a memeber struct snd_sst_params.codec to fill this.
so removing duplicate instance
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@linaro.org>
This patch adds the support to implement drain_notify in Intels mfld driver
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@linaro.org>
This reverts commit 0cac6fc3eb.
This comiit was dropped from rev2 and would not be required as it renames the
platform ops as well which is not required.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
I managed to drop a change to stream ID setting from commit 49fee17816
("ASoC: Intel: Only export one Baytrail DAI") leading to non-working
simultaneous capture-playback since after one DAI conversion
rtd->cpu_dai->id + 1 will be the same for both playback and capture.
Use substream->stream + 1 like it was in original Liam's patch.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use the %pap printk specifier to print resource_size_t variables. This
fixes warnings on platforms where resource_size_t has a different size
than int.
Signed-off-by: Laurent Pinchart <laurent.pinchart+renesas@ideasonboard.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We don't need more than one DAI for Baytrail SST. Usage becomes also more
straightforward by grouping playback and capture streams under the same PCM
device.
[Jarkko: I made Liam's sst-baytrail-pcm.c change a few lines smaller and
squashed together with my byt-rt5640.c change]
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Prepare for single Baytrail DAI playback/capture link by accessing PCM data
using stream ID instead of rtd->dev. Now rtd->dev is unique for playback
and capture since they are exported as separate DAIs but not once converted
to single DAI.
[Jarkko: Separated from another commit with updated commit log]
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The return statement was not indented correctly. I lined up the
condition a bit as well.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The if condition here was supposed to return on error but the return
statement is missing. The effect is that the ->mixername is set to
"???" instead of "DT019X".
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
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Merge tag 'asoc-v3.15-rc5-intel' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
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Merge tag 'asoc-v3.15-rc5-drivers' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
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Merge tag 'asoc-v3.15-rc5-core' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Core fixes for v3.15
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
Adding missing pm ops so that audio playback works across
suspend and resume cycle.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Additionally to the ak4642 pll frequencies the ak4648 also supports 13MHz,
19.2MHz and 26MHz. This adds support for these frequencies.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently unused, this is done to let the driver distinguish between
the different supported codec types in later patches.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
ALC and ALC Zero crossing detection has been enabled unconditionally.
Add controls for this.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds DVC (Digital Volume Controller)
support which is member of CMD unit.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
rsnd_mod_ops would like to come to use multi parameter.
modify macro to enable it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now, it can get rsnd_dai_stream pointer from rsnd_mod.
Remove duplicate parameter from rsnd_mod_ops
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
SRC module needs ADINR register settings,
but, it has many similar xxx_ADINR register,
and needs same settings.
This patch adds rsnd_get_adinr() to sharing code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current R-Car sound supports only SRC/SSI,
but, other module will be supported.
This patch adds rsnd_path_parse() macro to share code
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The register CLASS_D_CONTROL_1 is marked as volatile because it contains
a bit, DAC_MUTE, which is also mirrored in the ADC_DAC_CONTROL_1
register. This causes problems for the "Speaker Switch" control, which
will report an error if the CODEC is suspended because it relies on a
volatile register.
To resolve this issue mark CLASS_D_CONTROL_1 as non-volatile and
manually keep the register cache in sync by updating both bits when
changing the mute status.
Reported-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Commit 10df350977 ("ASoC: Intel: Fix Audio DSP usage when IOMMU is
enabled.") caused following regression in Baytrail SST:
baytrail-pcm-audio baytrail-pcm-audio: error: DMA alloc failed
baytrail-pcm-audio baytrail-pcm-audio: error: failed to load firmware
Fix this by calling dma_coerce_mask_and_coherent() in sst_byt_init() with
the same dma_dev device what is now used in sst_fw_new() when allocating the
DMA buffer.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This follows the same idea than commit 10df350977
("ASoC: Intel: Fix Audio DSP usage when IOMMU is enabled.") by using only
ACPI device for all DMA allocations. Since DMA masking is already done in
firmware loading it can be removed from here.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Broadwell display controller has 3 stream DMA engines. DMA0 cannot update DMA
postion buffer properly while DMA1 and DMA2 can work well. So this patch masks
the buggy DMA0 by keeping it as opened.
This is a tentative workaround, so keep the change small as Takashi suggested.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vendor ID 0x10de0071 is used by a yet-to-be-named GPU chip.
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We don't need to change those dividers if bclk and mclk remains the same
directions and values.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
According to Reference Manual -- ESAI Initialization chapter, as the
standard procedure of ESAI personal reset, the PCRC and PRRC registers
should be remained in its reset value and then configured after T/RCCR
and T/RCR configurations's done but before TE/RE's enabling.
So this patch moves PCRC and PRRC settings to the end of hw_params().
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ESAI can only output EXTAL clock source directly. But for FSYS clock source,
ESAI can not output it without getting through PSR PM dividers.
So this patch adds an extra check in the code.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The range here from 1 to 16 is confined to FP divider only while the
sck_div indicates if the calculation contains PSR and PM dividers. So
for the case using PSR and PM since the sck_div is true, the range of
ratio would simply become bigger than 16.
So this patch fixes the condition here and adds one line comments to
make the purpose here clear.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We may read an unexpected value when detemining which codec is attached.
In that case, either a unsupported codec is attached or something wrong
with I2C. The driver will not work properly on both cases. So we return
an error for that.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.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>
Suspend/resume support for the atmel-pcm-pdc driver was broken in commit
f0fba2ad1 ("ASoC: multi-component - ASoC Multi-Component Support"). It
essentially reverted the modifications done in commit 10cab262 ("ASoC: Change
how suspend and resume obtain the PCM runtime"). The suspend and resume handlers
at the beginning check if dai->runtime is not NULL, but dai->runtime is always
NULL, hence the code never runs. Considering that nobody noticed any problems in
the last 4 years since the code was broken and that the driver does not set
SNDRV_PCM_INFO_RESUME, which means applications are expected to stop and restart
the audio stream during suspend/resume, it is probably safe to assume that his
code is not needed and can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently when the DAPM context bias level is SUSPEND and the target bias level
is OFF dapm_pre_sequence_async() will first transition to PREPARE and
dapm_post_sequence_async() will then transition back from PREPARE to STANDBY and
then to OFF.
This patch makes sure that dapm_pre_sequence_async() only transitions to PREPARE
when either going to ON or away from ON. This avoids the extra unnecessary
transitions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since there is no support for compressed audio in Baytrail ADSP firmware
there is no need to leave it on during suspend since ALSA PCM buffers are
too small for leaving ADSP on for playing or recording.
Implement PM callbacks to Baytrail byt-rt5640.c machine driver that call
snd_soc_suspend and snd_soc_resume functions and unset the ignore_suspend
fields in DAI links.
This makes soc-core and ALSA core gracefully suspend and resume active
stream and call sst_byt_pcm_trigger() during suspend-resume cycle.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add suspend and resume support to Baytrail SST DSP. This is implemented by
unloading firmware modules and putting DSP into reset prior suspend and
restarting DSP again in normal boot state after resume.
Context restore for running streams is implemented by scheduling a work from
sst_byt_pcm_trigger() that will allocate a stream with existing parameters
and start it from last known buffer position before suspend.
[Jarkko: Squashed together 5 WIP patches from Liam and 1 from me]
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Suspend/resume requires reloading FW to boot state so we need to also make
sure that the driver matches the FW state at boot.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have to save the physical address of extended firmware block in the
beginning of mailbox every time when we boot the DSP firmware since that
mailbox address is re-used after DSP firmware is running. Otherwise DSP
firmware will get bogus extended firmware block address during next DSP
boot.
Currently this is not problem but becomes when DSP runtime rebooting is
implemented. Prepare for that by moving extended firmware address saving
from sst_byt_init() to sst_byt_boot().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Stream start position will be needed in resume code. Prepare for it by
adding start offset argument to sst_byt_stream_start().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Baytrail ADSP stream IPC simplifies a little by moving IPC_IA_START_STREAM
construction and sending directly into sst_byt_stream_start() from
sst_byt_stream_operations(). This is because IPC_IA_START_STREAM is only
stream IPC with extra message data so this move saves a few code lines.
Main motivation for this is to prepare for passing stream start position
to sst_byt_stream_start() which will be needed in resume code.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is for preparing suspend/resume support but can give also more
safeguard against concurrent timestamp structure access between DSP firmware
and host.
Now DSP DMA pointer is sampled in each pcm pointer callback in
sst_byt_pcm_pointer() but that is unneeded since DSP updates the timestamp
period basis and can potentially be racy if sst_byt_pcm_pointer() is called
when DSP is updating the timestamp.
By taking DSP DMA pointer only after period elapsed IPC messages in
byt_notify_pointer() and returning stored hw pointer in
sst_byt_pcm_pointer() there is less risk for concurrent access.
The same stored hw pointer can be also used in suspend/resume code for
restarting the stream at the same position.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit commit de9ba98b6d ("ASoC: dapm: Make widget power register settings more
flexible") added generic support for on_val/off_val in the DAPM core. With this
in place there is no need anymore for having a special event callback for
SND_SOC_DAPM_REG() widgets.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
If we find a widget who's stream name matches the name of a DAI widget then
thats the one it should be connected to. Based on the widget id we can say in
which direction the path should be. No need to go back to the DAI and check the
stream names.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We already know the widgets we want to connect, so use snd_soc_dapm_add_path()
instead of snd_soc_dapm_add_route() in snd_soc_dapm_new_pcm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We already know which two widgets should be connected, so use
snd_soc_dapm_add_path() instead of snd_soc_dapm_add_route() in
snd_soc_dapm_connect_dai_link_widgets().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This reverts commit bd23c5b661.
The patch claims that the patch is necessary to avoid double prefix addition
when calling snd_soc_dapm_add_route() from snd_soc_dapm_connect_dai_link_widgets().
But snd_soc_dapm_add_route() is called with the card's DAPM context, which does
not have a prefix, which means there is no prefix that could be added a second
time.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
For CODEC to CODEC DAI links the paths are created in snd_soc_dapm_new_pcm().
Also for CODEC to CODEC links the widgets are connected cross-over via a DAI
link widget, meaning that the capture widget of one CODEC will be connected to
the playback widget of the other and vice versa. Whereas
snd_soc_dapm_connect_dai_link_widgets() directly connects the playback widget of
the CPU DAI to the playback widget of the CODEC DAI and the capture widget of
the CPU DAI to the capture widget of the CODEC DAI. So not skipping
CODEC<->CODEC links in snd_soc_dapm_connect_dai_link_widgets() will create
incorrect connections between the two CODECs which will cause DAPM to detect
active paths where there are none and unnecessarily power up widgets.
Fixes: b893ea5 ("ASoC: sapm: Automatically connect DAI link widgets in DAPM graph.")
Cc: <stable@vger.kernel.org> (for 3.14+)
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>
Since commit 4b68b4e1c5 (ASoC: Intel: split the pcm and compress to
different files) the compressed ops haven't been built causing link
failures on allyesconfig and making the driver unbuildable. Add the
object to the Makefile to fix that.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by Vinod Koul <vinod.koul@intel.com>
When we plug a 3-ring headset on the Dell machines (VID: 0x10ec0255,
SID: 0x1028065c; VID: 0x10ec0255, SID: 0x10280680; VID: 0x10ec0292,
SID: 0x10280684), the headset mic can't be detected, after apply this
patch, the headset mic can work well.
And on the machine with SID 0x10280684, and the Lineout and external
microphone should be routed to docking, this patch also fix this
problem.
BugLink: https://bugs.launchpad.net/bugs/1297581
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>
this is for further updates to driver which supports DPCM :)
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
With DPCM we have media dai used and no seperate headset and speaker dai so
remove the speaker dai
The vibra is no longer supported thru audio, so remove
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For manging them and adding support for more platforms
Code move only
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
as this will be used in compressed split file in subsequent patch
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
to sst-mfld-platform-pcm.c so that we can split pcm and compress to different
files for upcoming changes to support more platforms
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add some SST API calls to unload and reload firmware modules. This can be used
by PM code to restore state and also allow modular FW to unload and release
memory blocks.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now, all platform is using new style rsnd_dai_platform_info.
Keeping compatibility is no longer needed.
We can cleanup code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
All platform which used old style was
switched to new style.
R-Car sound can remove old style clock support,
use device dependent clock now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Return statements in functions returning bool should use
true/false instead of 1/0.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch moves the private register settings from probe() to reg_default
struct.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This include file is about to disapear. In addition it is
useless for this code. So it is time to remove it.
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Mark Brown <broonie@linaro.org>
The headphone and mic jacks on Thinkpad T440 are assigned to pins NID
0x16 and 0x19, respectively. These need to be set up manually by a
fixup.
Reported-and-tested-by: Joschi Brauchle <joschi.brauchle@tum.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit d191bd8de8 ("ASoC: snd_soc_codec includes snd_soc_component") removed the
last user of the num_dai field. Also remove the field itself.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The global card list was removed in commit b19e6e7b7 ("ASoC: core: Use driver
core probe deferral"). The 'list' field of the snd_soc_card struct has been
unused since then. This patch removes the field.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit f0fba2ad1 ("ASoC: multi-component - ASoC Multi-Component Support") added
a per card list that keeps track of all the DAIs that have been registered with
the card, but the list has never been used. This patch removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Block offset calculations are done in the contiguous allocator so
are not required here.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Cirrus Logic Low Power Stereo I2C CODEC
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add support for RPDNEN, NSHHPEN, BRIDGOFF, CPWMEN and PNDLSL, and add DT
bindings to access them.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch fixes the following dereference check ordering.
sound/soc/intel/sst-haswell-pcm.c:749 hsw_pcm_probe() warn: variable dereferenced before check 'pdata' (see line 746)
git remote add asoc git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
git remote update asoc
git checkout 0b708c87f6
vim +/pdata +749 sound/soc/intel/sst-haswell-pcm.c
a4b12990 Mark Brown 2014-03-12 740 };
a4b12990 Mark Brown 2014-03-12 741
a4b12990 Mark Brown 2014-03-12 742 static int hsw_pcm_probe(struct snd_soc_platform *platform)
a4b12990 Mark Brown 2014-03-12 743 {
a4b12990 Mark Brown 2014-03-12 744 struct sst_pdata *pdata = dev_get_platdata(platform->dev);
a4b12990 Mark Brown 2014-03-12 745 struct hsw_priv_data *priv_data;
0b708c87 Liam Girdwood 2014-05-02 @746 struct device *dma_dev = pdata->dma_dev;
0b708c87 Liam Girdwood 2014-05-02 747 int i, ret = 0;
a4b12990 Mark Brown 2014-03-12 748
a4b12990 Mark Brown 2014-03-12 @749 if (!pdata)
a4b12990 Mark Brown 2014-03-12 750 return -ENODEV;
a4b12990 Mark Brown 2014-03-12 751
a4b12990 Mark Brown 2014-03-12 752 priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL);
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When using auto-muted controls it may happen that the register value will not
change when changing a control from enabled to disabled (since the control might
be physically disabled due to the auto-muting). We have to make sure to still
update the DAPM graph and disconnect the mixer input.
Fixes: commit 5729507 ("ASoC: dapm: Implement mixer input auto-disable")
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The function is only used locally, make it static.
Fixes the following warning from sparse:
sound/soc/soc-core.c:1644:22: warning: symbol 'soc_find_matching_codec' was not declared. Should it be static?
Fixes: 3ca041ed ("ASoC: dt: Allow Aux Codecs to be specified using DT")
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-By: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Other people would clearly understand each member and improve if they want.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
People would simply know what the driver gets the best for the current
sample rate playback.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The sysclk is one the clock sources that could be selected to derive
tx clock. But the route for sysclk is a bit different that it does
not only contain txclk df divider but also have an extra sysclk df.
So this patch mainly adds syclk df configuration support so as to
let the driver be able to get clock from sysclk.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We should have used _df by following the reference manual at the beginning.
So this patch just renames them.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The clock mux for the Freescale S/PDIF controller has eight clock sources
while most of them are from other moudles and even system clocks that do
not allow a rate-changing operation.
So we here only allow the clk_set_rate() and clk_round_rate() happened to
spdif root clock, the private clock for S/PDIF controller.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently infoframe contents and channel mapping are only set when a
sink (monitor) is present.
However, this does not make much sense, since
1) We can make a very reasonable guess on CA after 18e391862c ("ALSA:
hda - hdmi: Fallback to ALSA allocation when selecting CA") or by
relying on a previously valid ELD (or we may be using a
user-specified channel map).
2) Not setting infoframe contents and channel count simply means they
are left at a possibly incorrect state - playback is still allowed
to proceed (with missing or wrongly mapped channels).
Reasons for monitor_present being 0 include disconnected cable, video
driver issues, or codec not being spec-compliant. Note that in
actual disconnected-cable case it should not matter if these settings
are wrong as they will be re-set after jack detection, though.
Change the behavior to allow the infoframe contents and the channel
mapping to be set even without a sink/monitor, either based on the
previous valid ELD contents, if any, or based on sensible defaults
(standard channel layouts or provided custom map, sink type HDMI).
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Tested-by: Stephan Raue <stephan@openelec.tv>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit 1df5a06a ("ALSA: hda - hdmi: Fix programmed active channel
count") channel count is no longer being set if monitor_present is 0.
This is because setting the count was moved after the CA value is
determined, which is only after the monitor_present check in
hdmi_setup_audio_infoframe().
Unfortunately, in some cases, such as with a non-spec-compliant codec or
with a problematic video driver, monitor_present is always 0. As a
specific example, this seems to happen with gen1 ATV (SiI1390 codec),
causing left-channel-only stereo playback (multi-channel playback has
apparently never worked with this codec despite it reporting 8 channels,
reason unknown).
Simply setting converter channel count without setting the pin infoframe
and channel mapping as well does not theoretically make much sense as
this will just mean they are out-of-sync and multichannel playback will
have a wrong channel mapping.
However, adding back just setting the converter channel count even in
no-monitor case is the safest change which at least fixes the stereo
playback regression on SiI1390 codec. Do that.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: Stephan Raue <stephan@openelec.tv>
Tested-by: Stephan Raue <stephan@openelec.tv>
Cc: <stable@vger.kernel.org> # 3.12+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the Realtek ALC5645 codec driver. It is the base
version that because the jack detect function is not implemented to
it, the headphone and AMIC1 are not workable. We will fill up the
further functions later.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
we need _EXT version for SND_SOC_BYTES so that DSPs can use this to pass data
for DSP modules
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Read the stream offset and presentation position from DSP memory rather
than using the old estimated position. This fixes timing issues with
pulseaudio.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
hw_params() can be called multiple times. Make sure we release the DSP
stream that was allocated on previous hw_params() calls before allocating
a new DSP stream.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The Intel IOMMU requires that the ACPI device is used to allocate all
DMA memory buffers. This means we need to pass the DMA device pointer into child
component devices that allocate DMA memory.
We also only set the DMA mask for the ACPI device now instead of for each
component device.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fix page table creation on Haswell and Broadwell to remove unsafe
virt_to_phys mappings and use more portable SG buffer. Use audio buffer
APIs to allocate DMA buffers.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure we add the allocated blocks to the modules list of blocks.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure we dont alloc blocks twice with requests spanning more
than one block.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that INPUT is required for the CS42L52 and WM8962 we can remove the
IS_ENABLED(INPUT) check in the drivers.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The TEAC UD-H01 firmware sends wrong feedback frequency values, thus
causing the PC to send the samples at a wrong rate, which results in
clicks and crackles in the output.
Add a workaround to detect and fix the corruption.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
[mick37@gmx.de: use sender->udh01_fb_quirk rather than
ep->udh01_fb_quirk in snd_usb_handle_sync_urb()]
Reported-and-tested-by: Mick <mick37@gmx.de>
Reported-and-tested-by: Andrea Messa <andr.messa@tiscali.it>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent addition of the USB audio mixer suspend/resume may lead to
deadlocks when the driver tries to call usb_autopm_get_interface()
recursively, since the function tries to sync with the finish of the
other calls. For avoiding it, introduce a flag indicating the resume
operation and avoids the recursive usb_autopm_get_interface() calls
during the resume.
Reported-and-tested-by: Bryan Quigley <gquigs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The suspend callback of usb-audio driver may be called multiple times
per suspend when multiple USB interfaces are bound to a single sound
card instance. In such a case, it's superfluous to save the mixer
values multiple times. This patch fixes it by checking the counter.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This (widely used) construction:
if(printk_ratelimit())
dev_dbg()
Causes the ratelimiting to spam the kernel log with the "callbacks suppressed"
message below, even while the dev_dbg it is supposed to rate limit wouldn't
print anything because DEBUG is not defined for this device.
[ 533.803964] retire_playback_urb: 852 callbacks suppressed
[ 538.807930] retire_playback_urb: 852 callbacks suppressed
[ 543.811897] retire_playback_urb: 852 callbacks suppressed
[ 548.815745] retire_playback_urb: 852 callbacks suppressed
[ 553.819826] retire_playback_urb: 852 callbacks suppressed
So use dev_dbg_ratelimited() instead of this construction.
Signed-off-by: Sander Eikelenboom <linux@eikelenboom.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit 204dec93ea "ASoC: fsl: Allow to select individual common
options", it is possible to enable SND_SOC_FSL_SSI and SND_SOC_FSL_SPDIF
manually, either as loadable modules or built-in. This unfortunately
leads to a link error if one or both of them are built-in, while
the imx-pcm-dma framework is a loadable module:
sound/built-in.o: In function `fsl_ssi_probe':
:(.text+0x51fb8): undefined reference to `imx_pcm_dma_init'
sound/built-in.o: In function `fsl_spdif_probe':
:(.text+0x52e20): undefined reference to `imx_pcm_dma_init'
This changes Kconfig to prevent this case by using 'select' to turn
on the imx-pcm-dma code from both drivers. For consistency, we also
turn on the imx-pcm-fiq code, which is an alternative to the dma
implementation.
Note that imx-pcm-fiq is platform dependent, so we must not enable
that unless we are building a kernel for i.MX. Note also the
"if SND_IMX_SOC != n" syntax as opposed to the normal "if SND_IMX_SOC".
This is needed to avoid turning on the options as 'm' if 'SND_IMX_SOC'
is a module.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The cx20442 codec driver used here requires the TTY layer to
be enabled, or we get a link error:
sound/built-in.o: In function `cx20442_codec_remove':
cx20442.c:398: undefined reference to `tty_hangup'
sound/built-in.o: In function `ams_delta_remove':
ams-delta.c:613: undefined reference to `tty_unregister_ldisc'
sound/built-in.o: In function `ams_delta_cx20442_init':
ams-delta.c:559: undefined reference to `tty_register_ldisc'
This adds the missing dependency in the E3 configuration, there
was already one for the codec.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The WM8978 driver needs I2C to be enabled, so the
SND_SIU_MIGOR option also requires this.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This codec requires I2C to be enabled, so any other option
that selects it should also depend on I2C.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The WM8904 codec driver needs I2C to be enabled, so the
SND_ATMEL_SOC_WM8904 option also requires this.
Found using randconfig build testing.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This will be useful for out-of-tree drivers since in-tree drivers
could select it automatically.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This will be useful for out-of-tree drivers since in-tree drivers
could select it automatically.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The missing dependency can lead to build errors, so
make it explicit in Kconfig.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The UDA1380 driver needs I2C to be enabled, so
SND_SOC_SAMSUNG_H1940_UDA1380 and
SND_SOC_SAMSUNG_RX1950_UDA1380 also
require this.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The codec requires I2C to be enabled, so any other option
that selects it should also depend on I2C.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds device tree support to the Nokia N900 audio driver and
adds documentation for the DT binding.
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for specifying auxiliary codecs and
codec configuration via device tree phandles.
This change adds new fields to snd_soc_aux_dev and snd_soc_codec_conf
and adds support for the changes to SoC core methods.
Signed-off-by: Pavel Machek <pavel@ucw.cz>
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add more error messages making it easier to identify problems.
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Update the driver to get GPIO numbers from the
devm gpiod API instead of requesting hardcoded
GPIO numbers.
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is a preparation for DT based booting where the McBSP id
is set to -1 for all McBSP instances.
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_regulator_bulk_get() can make the code cleaner and smaller as we
do not need to call regulator_bulk_free() in the error and remove paths.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The symbol "nuc900_ac97_data" is used by the nuc900_pcm driver,
which may be a loadable module, so we should export it.
If one tries to build SND_SOC_NUC900 without SND_SOC_NUC900_AC97,
the kernel fails to link because of the reference to nuc900_ac97_data.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
dma_addr_t may be 64 bit wide, which causes a build failure
when doing a division on it. Here it is safe to cast to an
u32 type, which avoids the problem.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Tested-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This adds a missing dependency for SND_SOC_SMDK_WM8580_PCM to
require REGMAP_I2C to be enabled, avoiding possible build
erorrs.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This codec requires I2C to be enabled, so any other option
that selects it should also depend on I2C.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Building ARM randconfig got into a situation where CONFIG_INPUT
is turned off and SND_SOC_ALL_CODECS is turned on, which failed
for two codecs trying to use the input subsystem. Some other
drivers also select one of these codecs and consequently need an
explicit dependency added.
Appending to the dependency list seems the easiest way out,
since this is not a practical limitation. If anyone really
needs to build these codecs for a kernel with no input support,
a more sophisticated solution can be implemented.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
49101a25ac "ASoC: cq93vc: Remove the set_cache_io() entirely from
ASoC probe" introduced the cq93vc_get_regmap function that has an
obvious build error referring to the 'codec' variable that is not
declared anywhere"
sound/soc/codecs/cq93vc.c: In function 'cq93vc_get_regmap':
sound/soc/codecs/cq93vc.c:157:34: error: 'codec' undeclared (first use in this function)
struct davinci_vc *davinci_vc = codec->dev->platform_data;
^
This changes the code to compile again, presumably in the way it was
intended. Not tested.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch correct rt5640's device ID
Signed-off-by: Bard Liao <bardliao@realtek.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255,
SID: 0x1028067e), the headset mic can't be detected, after apply this
patch, the headset mic can work well.
BugLink: https://bugs.launchpad.net/bugs/1297581
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>
This patch adds devicetree support for mc13783-codec.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Acked-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch converts the rx51 ASoC module to use
devm_snd_soc_register_card.
Signed-off-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add module alias to support driver autoloading.
Signed-off-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Mark the array and the string const by using "static const char * const
foo[]" instead of "static const char* foo[]".
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Added machine driver to instantiate I2S based sound card on Snow
board. It has MAX98095 audio codec on board.
There are some other variants for Snow board which have MAX98090
audio codec. Hence support for MAX98090 is also added to this
driver.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The rxclk rate actually uses sysclk, ipg clock for example, as its
reference clock to calculate it. But the driver currently doesn't
pass a correct clock source. So 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>
This fixes module unload regressions introduced by commits 98e639fb8a
("ASoC: Track which components have been registered with
snd_soc_register_component()") and b37f1d123c ("ASoC: Let snd_soc_platform
subclass snd_soc_component").
First commit causes component_list to be corrupted when removing codec and
second when removing platform. Reason for both is that components associated
with platform or codec are never removed from the list because for them
registered_as_component field in struct snd_soc_component is always false.
Now list becomes corrupted when snd_soc_unregister_platform() or
snd_soc_unregister_codec() frees the platform or codec structure and where
the associated struct snd_soc_component is embedded.
Fix these by moving component unregistration and cleanup to a new local
function __snd_soc_unregister_component() that takes component as its
argument.
Since component is known for platforms and codecs the
__snd_soc_unregister_component() can be called directly and
snd_soc_unregister_component() takes care to find and unregister only
components that were registered using snd_soc_register_component().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The recent commit (ca460f8652) changed the CORB RP reset procedure to
follow the specification with a couple of sanity checks.
Unfortunately, Nvidia controller chips seem not following this way,
and spew the warning messages like:
snd_hda_intel 0000:00:10.1: CORB reset timeout#1, CORBRP = 0
This patch adds the workaround for such chips. It just skips the new
reset procedure for the known broken chips.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In soc_dapm_stream_event() we have the same code twice, once for the codec_dai
and once for the cpu_dai. This patch factors the duplicated code out into a
separate function. This will make it easier to modify the implementation (since
there is only one place that needs to be updated) and also easier to add support
for more than two DAIs per DAI link.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The introduced functios check AC97 if it's ready for communication and
read data is valid.
Signed-off-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It will help to maintain HW accessors and, for example, switch from the
direct I/O to MMIO which is more convenient for PCI devices.
Signed-off-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix format string mismatch in snd_seq_midisynth_register_port().
Argument type of p is unsigned int.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
New codec ALC293/ALC3235 support multifunction jacks.
It used for menual select the input device.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255,
SID: 0x10280674), the headset mic can't be detected, after apply this
patch, the headset mic can work well.
BugLink: https://bugs.launchpad.net/bugs/1297581
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>
The patch adds the rt5639 support to the OF match table.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This was initially removed in commit 6479f15ad ("ASoC: Remove name_prefix unset
during DAI link init hack"), but was brought back in commit 503ae5e0 ("ASoC:
core: Add helpers for dai link and aux dev init") by accident. This patch
removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_regulator_bulk_get() can make the code cleaner and smaller as we
do not need to call regulator_bulk_free() in the error and remove paths.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_regulator_bulk_get() can make the code cleaner and smaller as we
do not need to call regulator_bulk_free() in the error and remove paths.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_regulator_bulk_get() can make the code cleaner and smaller as we
do not need to call regulator_bulk_free() in the error and remove paths.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_regulator_bulk_get() can make the code cleaner and smaller as we
do not need to call regulator_bulk_free() in the error and remove paths.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_regulator_bulk_get() can make the code cleaner and smaller as we
do not need to call regulator_bulk_free() in the error and remove paths.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Restore correct parsing of dai-link subnodes with more explicit
implementation for applying the "simple-audio-card,"-prefix to
dai-link property and subnode names.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
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>
Separate DAI link and aux dev initialization in preparation for
DAI multicodec support.
Since aux dev will remain using single codecs but DAI links
will be able to support multiple codecs.
No functional change.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
[fparent@baylibre.com: Adapt to 3.14+]
Signed-off-by: Fabien Parent <fparent@baylibre.com>
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The current simple-card driver limits the DT node name to "sound".
Any of other names is forbidden while actually we should allow DT
to pass other node names.
And if this function is being called, the node must already have
the compatible "simple-audio-card" in DTB. So there should be no
need to check the name here.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now ASoC core is getting the intersection of supported formats not only
from CPU and CODEC dai's but also from DMA's. However, there should be
no specific width limitation from SDMA side.
So drop it. Otherwise, we would only support S16_LE format for all i.MX
platforms.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Regmap is able to enable/disable the core clock automatically each time
it's going to access the registers. But for DMA cases during playback or
recording, it's totally beyong control of regmap. So we have to open the
clock manually.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
I suppose there is a possibility that hsw_notification_work() may run after
sst_hsw_stream_free() which can lead to a kernel crash since struct
sst_hsw_stream is freed at that point and
stream = container_of(work, struct sst_hsw_stream, notify_work) is not valid
when hsw_notification_work() is run.
Reported-by: Derek Basehore <dbasehore@chromium.org>
Reported-by: Wenkai Du <wenkai.du@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Takashi Sakamoto