The previous enable flow:
1, Enable TE&RE (SAI starts to consume tx FIFO and feed rx FIFO)
2, Mask IRQ of Tx/Rx to enable its interrupt.
3, Enable DMA request of Tx/Rx.
As this flow would enable DMA request later than TERE, the Tx FIFO
would be easily emptied into underrun while Rx FIFO would be easily
stuffed into overrun due to the delayed DMA transfering.
This issue happened merely occational before the patch 'ASoC: fsl_sai:
Reset FIFOs after disabling TE/RE' because there were useless data
remaining in the FIFO for the gap. However, it manifested after FIFO
reset's implemented.
After this patch, the new flow:
1, Enable DMA request of Tx/Rx.
2, Enable TE&RE (SAI starts to consume tx FIFO and feed rx FIFO)
3, Mask IRQ of Tx/Rx to enable its interrupt.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
TE/RE bit of T/RCSR will remain set untill the current frame is physically
finished. The FIFO reset operation should wait this bit's totally cleared
rather than ignoring its status which might cause TE/RE disabling failed.
This patch adds delay and timeout to wait for its completion before FIFO
reset.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For trigger start, we don't need to check if it's the first time to
enable TE/RE or second time. It doesn't hurt to enable them any way,
which in the meantime can reduce race condition for TE/RE enabling.
For trigger stop, we will definitely clear FRDE of current direction.
Thus the driver only needs to read the opposite one's.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
rsnd driver is using SSI/SRC/DVC which are
using "mod" base operation.
These "mod" are supporting "probe" and "remove" callbacks.
Current rsnd_probe should call "remove" if "probe" was failed,
since "probe" might be having DMAEngine handle.
Some mod's "remove" callback might be called without calling
"probe", but it is no problem. because "remove" do nothing
in such case.
So, all mod's "remove" should be called when error case
of rsnd_probe() and rsnd_remove().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In order to show the option under 'CODEC drivers' we need to have text in
Kconfig.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The codec can be configured via I2C and using regmap_i2c.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Changes to the AIF configuration registers only take
effect when the AIF is disabled. If the configuration
is being changed from the previous setup, temporarily
disable the AIF.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Different playback and capture bits-per-sample
are not supported on the AIFs
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Different playback and capture bits-per-sample
are not supported on the AIFs
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Different playback and capture bits-per-sample
are not supported on the AIFs
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have now everything in place to actual let a component register controls. Add
a function which allows to do so.
Also update snd_soc_add_codec_controls() and snd_soc_platform_controls() to use
this new function internally. And while we are at it also change the
num_controls parameter of those two functions from int to unsigned int.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Both the snd_soc_codec and snd_soc_platform struct do have a pointer to the
parent card and both handle this pointer in mostly the same way. This patch
moves the card field to the component level which will allow further code
consolidation between platforms and CODECS.
Since there are only a handful of users of the snd_soc_codec struct's card field
(and none of the snd_soc_platform's) these are update in this patch as well,
which allows it to be removed 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>
Signed-off-by: Mark Brown <broonie@linaro.org>
The platform_dev_list was added in commit f0fba2ad1b ("ASoC: multi-component -
ASoC Multi-Component Support") and while platforms are added and remove from
that list it is otherwise unused. This patch removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Dell XPS 15 (1028:05fe) gives the bad feedback noise from analog
loopback line even if the channels are muted. The similar problem has
been seen on XPS 13, and it was fixed by simply disabling loopback.
The same fixup was confirmed to be applicable and fix the problem on
XPS 15, too.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=80821
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a supplement to my previous patch.
http://mailman.alsa-project.org/pipermail/alsa-devel/2014-July/079190.html
The special_clk_ctl_put() still returns 0 in error handling case. It should
return -EINVAL.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is for correction of my misunderstanding about return value of
.put callback in ALSA Control interface.
According to 'Writing ALSA Driver' (*1), return value of the callback has
three patterns; 1: changed, 0: not changed, an negative value: fatal error.
But I misunderstood that it's boolean; zero or nonzero.
*1: Writing an ALSA Driver (2005, Takashi Iwai)
http://www.alsa-project.org/main/index.php/ALSA_Driver_Documentation
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit uses different labels for control elements of digital input/output
interfaces to correct my misunderstanding about M-Audio Firewire 1814 and
ProjectMix I/O.
According to user manuals for these two models, they have two modes for
digital input; one is S/PDIF in both of optical and coaxial interfaces,
another is ADAT in optical interface only.
But in current implementation, a control element for it reduced labels which
a control element for digital output uses because of my misunderstanding
that optical interface is not available for digital input with S/PDIF mode.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In error handling case, special_clk_ctl_put() returns without unlock_mutex(),
therefore the mutex is still locked. This commit moves mutex_lock() after
the error handling case.
This commit is my solution for this post.
[PATCH -next] ALSA: bebob: Fix missing unlock on error in special_clk_ctl_put()
https://lkml.org/lkml/2014/7/20/12
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new quirk table to make sure that pin quirks have a higher
priority than quirks that apply to an entire vendor.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previously, calling one quirk function first and another later
would make the latter one take priority, this is now changed
to make the former take priority.
By adding two special values for fixup_id we can also get rid of the
fixup_forced flag.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Gigabyte BRIX BXBT-2707 is a mini-PC with Realtek ALC283 HDA,
exposing a single headset jack.
However, the default pin config information only suggests
that one pin is connected: a HP out jack (pin 0x21, default
config 0x04211010).
The microphone input is behind pin 0x19, which has default
config 0x411111f0 (i.e. unused), so it does not show up in
userspace, and no microphone input is possible via the headset.
Override the pin config so that the headset mic can be used.
[rearranged the fixup entry position by tiwai]
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This makes the mute LED work on a HP 15 touchsmart machine.
BugLink: https://bugs.launchpad.net/bugs/1334950
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the pm suspend/resume it is better
to disable the GPIO after the regmap_cache
setting calls so that if the call is interrupted
the new reg values will be cached and set on resume.
Also add pm_runtime_put in the remove call.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For controlling the new fields more strictly, add sw_params.proto
field indicating the protocol version of the user-space. User-space
should fill the SNDRV_PCM_VERSION value it's built with, then kernel
can know whether the new fields should be evaluated or not.
And now tstamp_type field is evaluated only when the valid value is
set there. This avoids the wrong override of tstamp_type to zero,
which is SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Quite a few build coverage fixes in here among the usual small driver
fixes includling the sigmadsp change from Lars - moving the driver to
separate modules per bus (which is basically just code motion) avoids
issues with some combinations of buses being enabled.
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Merge tag 'asoc-v3.16-rc1' into asoc-linus
ASoC: Fixes for v3.16
Quite a few build coverage fixes in here among the usual small driver
fixes includling the sigmadsp change from Lars - moving the driver to
separate modules per bus (which is basically just code motion) avoids
issues with some combinations of buses being enabled.
# gpg: Signature made Thu 19 Jun 2014 11:57:31 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>"
Remove unnecessary bit shifts.
Correct min value to match datasheet.
Num steps = number of steps between min and max.
Reported-by: Ryan Harvey <ryan.harvey@cirrus.com>
Signed-off-by: Ryan Harvey <ryan.harvey@cirrus.com>
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Stereo Headphone and LineOut volume mixers are now attached to HPA+HPB, LOA+LOB.
Reported-by: Ryan Harvey <ryan.harvey@cirrus.com>
Signed-off-by: Ryan Harvey <ryan.harvey@cirrus.com>
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In the rx irq handling part, we should clear the flags in RCSR not TCSR.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
SAI will not clear their FIFOs after disabling TE/RE. Therfore, the driver
should take care the task so as not to let useless data remain in the FIFO.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Support the TI TAS2552 Class D amplifier.
The TAS2552 is a high efficiency Class-D audio
power amplifier with advanced battery current
management and an integrated Class-G boost
The device constantly measures the
current and voltage across the load and provides a
digital stream of this information.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The third parameter for snd_pcm_format_set_silence needs the number
of samples instead of sample bytes.
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
It was accidentally added in commit 44ba2641 ("ASoC: dapm: Add support for DAI
multicodec").
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add support for Broadwell based machines with SST DSP audio.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jie Yang <yang.jie@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.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>
Multiple codecs does not make sense in the case of compressed
stream.
Exit with error if it happens.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
DAI link assumes a one to one mapping between CPU DAI and CODEC. In
some cases, the same CPU DAI can be connected to several codecs.
This is the case for example, if you connect two mono codecs to the
same I2S link in order to have a stereo card.
The current ASoC implementation does not allow such setup.
Add support for DAI links composed of a single CPU DAI and multiple
CODECs. Sound cards have to pass the CODECs array in the corresponding
DAI link through a new 'snd_soc_dai_link_component' struct. Each CODEC in
this array is described in the same manner single CODEC DAIs are
(either DT/OF node or codec_name).
Multi-codec links are not supported in the case of CODEC to CODEC links.
Just print a warning if it happens.
Based on an original code done by Misael.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Fabien Parent <fparent@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>
ALSA supports arbitrary length TLVs for each kcontrol that can be used
to pass metadata about the control (e.g. volumes, enum information). The
same transport mechanism is now used for arbitrary length data by
defining a new helper.
Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Update FW version readback. IPC_GLB_GET_FW_VERSION reads back
the ABI version whilst the release version is in the mailbox.
Update to use mailbox version for info logging.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
If a machine driver provides an BCLK divider to the McASP driver, skip the
automatic calculation.
This fixes machines on which the physical sample transport always works
in 32 bits, even though not all of them are actually used.
snd_soc_params_to_bclk() will fail to address such cases properly.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use the edma-pcm with AM335x and AM437x SoCs.
Keep using the davinci-pcm for daVinci devices, they can be switched to use
the dmaengine based driver later when they are verified to work correctly.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
So drivers mixing with other platform drivers, like davinci-mcasp do not
need to fiddle with CONFIG_SND_EDMA_SOC in their code.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The SNDRV_PCM_INFO_BATCH will be added back based on the dmaengine driver's
capabilities.
Patches for edma dmaengine driver has been prepared to suppress the
interrupts. We can add this flag right away and have the benefit of not
having interrupts during audio activity when the edma and ASoC patches
are in the same tree.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Things seem to calm down so far, just a small few HD-audio fixes
(regression fixes and a new codec ID addition) popping up.
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Merge tag 'sound-3.16-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Things seem to calm down so far, just a small few HD-audio fixes
(regression fixes and a new codec ID addition) popping up"
* tag 'sound-3.16-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix broken PM due to incomplete i915 initialization
ALSA: hda - Revert stream assignment order for Intel controllers
ALSA: hda - Add new GPU codec ID 0x10de0070 to snd-hda
ALSA: hda: Fix build warning
I previously added an integer overflow check here but looking at it now,
it's still buggy.
The bug happens in snd_compr_allocate_buffer(). We multiply
".fragments" and ".fragment_size" and that doesn't overflow but then we
save it in an unsigned int so it truncates the high bits away and we
allocate a smaller than expected size.
Fixes: b35cc82258 ('ALSA: compress_core: integer overflow in snd_compr_allocate_buffer()')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The sst-haswell-dsp.c is an ACPI independent file, this patch removes
ACPI header files for it.
Signed-off-by: Lv Zheng <lv.zheng@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since init_failed flag was moved to struct hda_intel, its access in
the commit [4da63c6f: ALSA: hda - Fix broken PM due to incomplete
i915 initialization] is also replaced with hda->init_failed
appropriately.
When the initialization of Intel HDMI controller fails due to missing
i915 kernel symbols (e.g. HD-audio is built in while i915 is module),
the driver discontinues the probe. However, since the probe was done
asynchronously, the driver object still remains, thus the relevant PM
ops are still called at suspend/resume. This results in the bad access
to the incomplete audio card object, eventually leads to Oops or stall
at PM.
This patch adds the missing checks of chip->init_failed flag at each
PM callback in order to fix the problem above.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=79561
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The s3c_dma_client structures and the 'ch' and 'ops' members in
s3c_dma_params were only used by the legacy DMA driver and serve
no function any more. This removes any reference to them.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit ae602456e8 ("ASoC: samsung: drop support for legacy
S3C24XX DMA API") removed the old code for the samsung specific
DMA interfaces, now that everybody can use dmaengine.
This picks up the few remaining pieces left over by that patch:
The most important one is the removal of the dma_data->ops->started()
calls in ac97. My understanding is that these are only required
for drivers that do not support cyclic transfers, which the new dma
engine driver now does, so we can simply remove them. This would also
fix at least one bug in the ac97 driver on newer machines, which
currently gives us a NULL pointer dereference from trying to call
dma_data->ops->started().
Further, we must no longer 'select' S3C2410_DMA, which conflicts
with the dmaengine driver. The SND_S3C_DMA symbol is now
useless, because it is always selected, so we can remove it
and build the dmaengine support unconditionally.
Finally, we should not 'select' S3C24XX_DMAC or S3C64XX_PL080,
which may have additional dependencies. This replaces it with
'depends on', to be more conservative.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This interface will be used by subsequent patches to set/get parameters from DSP
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
DSP returns error codes for IPC return so add them in driver
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
the shim registers start and end can be useful while parsing the shim addresses,
so add these
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Merge D-SRAM0 D-SRAM1 D-SRAM2 to D-SRAM, for wild cat point ADSP mem regions.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add dummy read after each block enable, to workaround
SRAM write missing bytes issue.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This change removes unsupported formats from System,
Capture and Loopback FE DAIs.
Also it fixes S24_LE support on all DAIs.
While at this fix 24 bit flag for BYT as well.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
This patch removes s5p64x0 related WM8580 because of removing support
for s5p64x0 SoCs.
Signed-off-by: Kukjin Kim <kgene.kim@samsung.com>
[pebolle@tiscali.nl: reported missing dependency for smdk6450]
Reported-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Mark Brown <broonie@linaro.org>
I got another build error from SND_SOC_SMARTQ selecting
SND_SOC_WM8750 without ensuring that I2C is enabled,
in this example, i2c is a loadable module:
sound/built-in.o: In function `wm8750_i2c_probe':
:(.text+0x3e6c0): undefined reference to `devm_regmap_init_i2c'
sound/built-in.o: In function `wm8750_exit':
:(.exit.text+0x3f4): undefined reference to `i2c_del_driver'
sound/built-in.o: In function `wm8750_modinit':
:(.init.text+0x13d0): undefined reference to `i2c_register_driver'
This changes the samsund ASoC Kconfig to have explicit
I2C dependencies for each driver that uses an i2c bus and
SPI dependencies for the ones using those.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch introduces the use of managed interfaces like devm_clk_get
and does away with the clk_puts in the probe and remove functions. A
label is also done away with.
Signed-off-by: Himangi Saraogi <himangi774@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This resulted from an unfortunate copy/paste to similar DEFINE name.
Detected automagically by kbuild.
Signed-off-by: xujianqun <xjq@rock-chips.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We got a regression report for 3.15.x kernels, and this turned out to
be triggered by the fix for stream assignment order. On reporter's
machine with Intel controller (8086:1e20) + VIA VT1802 codec, the
first playback slot can't work with speaker outputs.
But the original commit was actually a fix for AMD controllers where
no proper GCAP value is returned, we shouldn't revert the whole
commit. Instead, in this patch, a new flag is introduced to determine
the stream assignment order, and follow the old behavior for Intel
controllers.
Fixes: dcb32ecd9a ('ALSA: hda - Do not assign streams in reverse order')
Reported-and-tested-by: Steven Newbury <steve@snewbury.org.uk>
Cc: <stable@vger.kernel.org> [v3.15+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull slave-dmaengine fixes from Vinod Koul:
"We have two small fixes. First one from Daniel to handle 0-length
packets for usb cppi dma. Second by Russell for imx-sdam cyclic
residue reporting"
* 'fixes' of git://git.infradead.org/users/vkoul/slave-dma:
Update imx-sdma cyclic handling to report residue
dma: cppi41: handle 0-length packets
Currently configuration of the CDCLK pad is being overwritten in
the i2s_shutdown() callback in order to gate the SoC output clock.
However if an ASoC machine driver doesn't restore that clock
settings each time after opening the sound device this results
in the CDCLK pin being permanently configured into input mode.
I.e. the output clock will always stay disabled.
Fix that by saving the CDCLKCON bit state in i2s_shutdown() and
and restoring it in the i2s_startup() callback.
Signed-off-by: Chen Zhen <zhen1.chen@samsung.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Missing module.h in linux directory will cause compile error with randconfig,
like as:
sound/soc/rockchip/rockchip_i2s.c:526:20: error: expected declaration
specifiers or ‘...’ before string constant
MODULE_DESCRIPTION("ROCKCHIP IIS ASoC Interface");
Signed-off-by: xujianqun <xjq@rock-chips.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For allowing adjusting the timestamp type on the fly, add it to
sw_params. The existing ioctl is still kept for compatibility.
Along with this, increment the PCM protocol version.
The extension was suggested by Clemens Ladisch.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No functional change.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, the driver places no restrictions on changes that can be
applied to an active FLL. However, it is only possible to change the
input for an active FLL, to change the output the FLL should be stopped
and then recofigured. This patch disallows changes in output frequency
and adds some additional handling to ensure the output remains
consistent across an input transition.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The FLL freerun is only required whilst we disable the FLL not the
entire time the FLL is disabled. This patch moves the FLL freerun
disable from the enable sequence to the disable sequence.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
arizona_is_enabled_fll currently returns a bool, but can throw an error.
The error will be basically ignored and we will treat the FLL as already
on. This patch changes the return to be an int and adds error code to
propagate the error up to the callback.
Reported-by: Anil Kumar <anil.kumar@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When configuring the FLL we must ensure that the reference clock passed
to the FLL is under a certain limit. This limit was specified
incorrectly in the current code, this patch corrects this. Although the
error will only be encountered in some edge cases.
Reported-by: Ryo Tsutsui <ryo.tsutsui@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The check to ensure the Fref frequency is within the bounds for the
current ratio, was placed in the wrong loop. The initial configuration
will always be valid and the loop lowering the ratio will only reinforce
this validity. The check should be on the loop increasing the ratio.
This could on occasion cause an invalid ratio/Fref combination to be
selected.
Reported-by: Ryo Tsutsui <ryo.tsutsui@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Zero is not a valid FRATIO for the FLL, as such we shouldn't test it
whilst refining the FRATIO. This patch does just that.
Reported-by: Ryo Tsutsui <ryo.tsutsui@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the sound subsystem driver for Odroid-X2 and
Odroid-U3 boards. The codec works in I2S master mode; there
are two separate audio routing paths defined, as there are
differences in the signal routing between the X2 and U3 boards,
i.e. U3 uses single jack for headphones and microphone.
Signed-off-by: Chen Zhen <zhen1.chen@samsung.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
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: Manish Badarkhe <badarkhe.manish@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
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: Manish Badarkhe <badarkhe.manish@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using the sgtl5000 codec driver as a module and trying to remove it causes the
followig kernel oops:
root@freescale /$ rmmod snd-soc-imx-sgtl5000
[ 117.122920] ------------[ cut here ]------------
[ 117.127609] WARNING: CPU: 0 PID: 631 at drivers/regulator/core.c:3604 regula)
[ 117.137046] Modules linked in: snd_soc_imx_sgtl5000(-) snd_soc_sgtl5000 evbug
[ 117.144315] CPU: 0 PID: 631 Comm: rmmod Not tainted 3.16.0-rc3-next-201407043
[ 117.153366] Backtrace:
[ 117.155865] [<80011e5c>] (dump_backtrace) from [<80011ff8>] (show_stack+0x18)
[ 117.163484] r6:802fcc48 r5:00000000 r4:00000000 r3:00000000
[ 117.169228] [<80011fe0>] (show_stack) from [<80668cc0>] (dump_stack+0x88/0xa)
[ 117.176508] [<80668c38>] (dump_stack) from [<80029a38>] (warn_slowpath_commo)
[ 117.184696] r5:00000009 r4:00000000
[ 117.188322] [<800299c8>] (warn_slowpath_common) from [<80029a80>] (warn_slow)
[ 117.197150] r8:dd60d600 r7:ddfa6d00 r6:dd5d9064 r5:dd5e0f90 r4:dd5d9400
[ 117.203983] [<80029a5c>] (warn_slowpath_null) from [<802fcc48>] (regulator_u)
[ 117.212828] [<802fcb74>] (regulator_unregister) from [<7f0047c4>] (ldo_regul)
[ 117.223475] r4:dd59e300 r3:dd5e0f90
[ 117.227100] [<7f00479c>] (ldo_regulator_remove [snd_soc_sgtl5000]) from [<7f)
[ 117.238959] r4:dd5d8000 r3:ddd51420
[ 117.242623] [<7f0047dc>] (sgtl5000_remove [snd_soc_sgtl5000]) from [<804e5b1)
[ 117.252489] r5:00000000 r4:dd5d8000
[ 117.256111] [<804e5af8>] (soc_remove_codec) from [<804e5ed4>] (soc_remove_da)
[ 117.264933] r4:ddfb640c r3:00000000
[ 117.268555] [<804e5c10>] (soc_remove_dai_links) from [<804e5fbc>] (snd_soc_u)
[ 117.277810] r10:80359e48 r9:dd684000 r8:dd5ca800 r7:dd685e60 r6:00000001 r5c
[ 117.285761] r4:dd5d9064
[ 117.288329] [<804e5f28>] (snd_soc_unregister_card) from [<804f29f0>] (devm_c)
[ 117.297324] r6:00000004 r5:ddd0fc10 r4:dd5a7980 r3:804f29dc
[ 117.303095] [<804f29dc>] (devm_card_release) from [<8035a418>] (release_node)
[ 117.311369] [<8035a2a8>] (release_nodes) from [<8035aab4>] (devres_release_a)
[ 117.319583] r10:00000000 r9:dd684000 r8:8000ed64 r7:00000081 r6:ddd0fc44 r54
[ 117.327543] r4:ddd0fc10
[ 117.330108] [<8035aa7c>] (devres_release_all) from [<803571c8>] (__device_re)
[ 117.339214] r4:ddd0fc10 r3:dd5d9010
[ 117.342989] [<80357148>] (__device_release_driver) from [<80357a48>] (driver)
[ 117.351607] r5:7f00c9e4 r4:ddd0fc10
[ 117.355285] [<8035798c>] (driver_detach) from [<80357030>] (bus_remove_drive)
[ 117.363454] r6:00000880 r5:00000000 r4:7f00c9e4 r3:dd6c3a80
[ 117.369195] [<80356fdc>] (bus_remove_driver) from [<803580b8>] (driver_unreg)
[ 117.377683] r4:7f00c9e4 r3:dd60d480
[ 117.381305] [<80358088>] (driver_unregister) from [<80358bb4>] (platform_dri)
[ 117.390642] r4:7f00ca28 r3:7f00c35c
[ 117.394309] [<80358ba0>] (platform_driver_unregister) from [<7f00c370>] (imx)
[ 117.406188] [<7f00c35c>] (imx_sgtl5000_driver_exit [snd_soc_imx_sgtl5000]) f)
[ 117.417452] [<8008c2b8>] (SyS_delete_module) from [<8000eba0>] (ret_fast_sys)
[ 117.425753] r6:5f636f73 r5:5f646e73 r4:00016f40
[ 117.430428] ---[ end trace 8fd8a5cb39e46d0e ]---
This problem is well explained by Russell King:
"The sgtl5000 uses a two-stage initialisation process. The first stage
is when the platform driver is probed, where some resources are found
and initialised. The second stage is via the codec driver's probe
function, where regulators are found and initialised using the managed
resource support.
The problem here is that this works fine until the codec driver is
unbound. When this occurs, sgtl5000_remove() is called which is a no-op
as far as the managed resource code is concerned. The regulators remain
allocated, and their pointers in sgtl5000_priv remain valid.
If the codec is now re-probed, it will again try and find the regulators,
which will now be busy. This will fail.
That's not the only problem - if using the LDO regulator, the regulator
is unregistered from the regulator core code, but it still has a user.
When the user is cleaned up (eg, by removing the module) it hits the
free'd regulator, and this can oops the kernel.
This bug was originally introduced by 63e54cd9ca ("ASoC: sgtl5000:
Use devm_regulator_bulk_get()")."
This reverts commit 63e54cd9ca.
Tested on a imx53-qsb board.
Reported-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_of_parse_card_name() may be called before card->dev has been
set, which results in a kernel panic.
Unable to handle kernel NULL pointer dereference at virtual address 00000210
PC is at snd_soc_of_parse_card_name+0x18/0x54
LR is at snow_probe+0x5c/0xd4
Add an error check in snd_soc_of_parse_card_name() to take care of this
case and print out a message in case of error.
Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add driver for i2s controller found on rk3066, rk3168 and rk3288
processors from rockchip.
Tested on the RK3288 SDK board.
Signed-off-by: Jianqun Xu <xjq@rock-chips.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The Swap controls for ADC/PCM paths should be in the DAPM domain.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now the led_power_filter() needs to handle 3 situations:
- only mute_led_nid is set
- only cap_mute_led_nid is set
- both mute_led_ind and cap_mute_led_nid are set
BugLink: https://bugs.launchpad.net/bugs/1329580
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: Kailang Yang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vendor ID 0x10de0070 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>
The max98090_{suspend,resume}() functions are used for system sleep and
therefore need to be guarded by CONFIG_PM_SLEEP rather than CONFIG_PM.
Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch fix the issue that the output is almost silent at the
beginning of starting a playback.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The hda_tegra_disable_clocks() function is only used by the suspend and
resume code, so it needs to be included in the #ifdef CONFIG_PM_SLEEP
block to prevent the following warning:
CC sound/pci/hda/hda_tegra.o
sound/pci/hda/hda_tegra.c:238:13: warning: 'hda_tegra_disable_clocks' defined but not used [-Wunused-function]
static void hda_tegra_disable_clocks(struct hda_tegra *data)
^
Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move codec initial settings from asoc probe to i2c probe.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch modifies the voltage level in the BIAS OFF stage
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch removes the redundant definition in head file
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
print_nid_path has a possible buffer overflow if
struct nid_path.path values are > 256.
Avoid this and neaten the output to remove the leading ':'
Neaten debug_badness to always verify arguments.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The s6105-ipcam driver calls i2c_register_board_info() which means that
not only does it need I2C to be enabled to be built but since that
symbol is not exported it means that the driver also can't be linked as
a module. Use a bool instead of a tristate and add a dependency on I2C
to fix these issues.
Commit b5dd60fd26 (ASoC: s6000: Allow to build when COMPILE_TEST is
enabled) made this problem more visible since it enabled build of the
driver on more common architectures but this has always been an issue.
Signed-off-by: Mark Brown <broonie@linaro.org>
Snow sound-card driver supports multiple boards with different
audio codecs. Updating the sound card name per board basis would provide
some more information to the end-user.
Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
max98090_handle_pdata() is not doing anything other than printing
a message if pdata is not valid. This can be removed.
Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
CODEC type (MAX98090/MAX98091) can be specified from device-tree file,
it can also be obtained from the CODEC during runtime.
Add an explicit check to figure out if both are matching, else print
a message warning about the same.
Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We should save/restore relevant I2S registers regardless of
the dai->active flag, otherwise some settings are being lost
after system suspend/resume cycle. E.g. I2S slave mode set only
during dai initialization is not preserved and the device ends
up in master mode after system resume.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
The newly introduced sirf-usp driver defines sirf_usp_pcm_{suspend,resume}
functions only when PM_RUNTIME is enabled, but also uses them when that
is disabled and only PM_SLEEP is turned on, resulting in this error:
../sound/soc/sirf/sirf-usp.c: In function 'sirf_usp_pcm_suspend':
../sound/soc/sirf/sirf-usp.c:308:3: error: implicit declaration of function 'sirf_usp_pcm_runtime_suspend' [-Werror=implicit-function-declaration]
sirf_usp_pcm_runtime_suspend(dev);
^
../sound/soc/sirf/sirf-usp.c: In function 'sirf_usp_pcm_resume':
../sound/soc/sirf/sirf-usp.c:319:3: error: implicit declaration of function 'sirf_usp_pcm_runtime_resume' [-Werror=implicit-function-declaration]
ret = sirf_usp_pcm_runtime_resume(dev);
^
cc1: some warnings being treated as errors
To fix that, this patch changes the #ifdef to CONFIG_PM, which
is enabled when at least one of PM_SLEEP or PM_RUNTIME are enabled.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
All CODEC input and output widgets are either in the DAPM routing table or
manually marked as non-connected. This means the card is fully routed and we can
let the core take care of disconnecting non-connected pins.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Most of the ASoC s6000 code is architecture independent. This patch makes it
possible to select the platform when COMPILE_TEST is enabled.
The only architecture dependent code is the PCM driver which will still only be
selected if XTENSA_VARIANT_S6000 is enabled.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The platform_driver remove callback return type is int not void.
Fixes the following warning:
sound/soc/s6000/s6000-i2s.c:604:19: warning: incorrect type in initializer (different base types)
sound/soc/s6000/s6000-i2s.c:604:19: expected int ( *remove )( ... )
sound/soc/s6000/s6000-i2s.c:604:19: got void ( static [toplevel] *<noident>)( ... )
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In case of _3LE/_3BE formats the samples are stored in 3 consecutive bytes
without padding it to 4 bytes. This means that the DMA needs to be able to
support 3 bytes word length in order to read/write the samples from memory
correctly. Originally the code treated 24 bits physical length samples as
they were 32 bits which leads to corruption when playing or recording audio.
The hw.formats field has already been prepared to exclude formats not
supported by the DMA engine in use, which means that only on platforms where
3 bytes is supported by the DMA will be able to use this format.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Based on the dma_slave_caps's addr_widths queried from the dma driver
prepare the hw.formats mask to include only formats which is supported by
the DMA engine.
In case the dma driver does not implement the slave_caps the default
assumption is that it supports 1, 2 and 4 bytes widths.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
params_physical_width() is available via pcm_params.h
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds Realtek ALC286 codec driver.
ALC286 is a dual mode codec, which can run as HD-A or I2S mode.
It is controlled by HD-A verb commands via I2C protocol.
The following is the I/O difference between ALC286 and general I2S codecs.
1. A HD-A verb command contains three parts, NID, VID, and PID.
And an I2S command contains only two parts: address and data.
2. Not only the register address is written, but the read command also
includes the entire write command.
3. rt286 uses different registers for read and write the same bits.
We map verb command to regmap structure. However, we read most registers from
cache to prevent the asymmetry read/write issue in rt286.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since MODULE_LICENSE is missing the module load fails,
so add this for module.
Signed-off-by: Praveen Diwakar <praveen.diwakar@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
This small patch completes #defines for Control/Status Register,
adds comments for the missing ones there and on the Interrupt Mask
Register and additionally corrects "#define ICE1712_SERR_LEVEL 0x04 -> 0x08",
according to documentation.
Signed-off-by: Konstantinos Tsimpoukas <kostaslinuxxx@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For HSW/BDW display HD-A controller, hda_set_bclk() is defined to set BCLK
by programming the M/N values as per the core display clock (CDCLK) queried from
i915 display driver.
And the audio driver will also set BCLK in azx_first_init() since the display
driver can turn off the shared power in boot phase if only eDP is connected
and M/N values will be lost and must be reprogrammed.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a long standing bug in the read register routing of adau1701.
The bytes arrive in the buffer in big-endian, so the result has to be
shifted before and-ing the bytes in the loop.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
This patch adds ASoC support for SiRF SoCs USP interface.
Features include:
1. Only support slave mode.
2. Support I2S and DSP_A mode.
3. Support S16_LE, S24_LE and S24_3LE formats.
4. Support stereo and mono mode.
5. The biggest Support is 192Khz sample rate.
Signed-off-by: Rongjun Ying <rongjun.ying@csr.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The TAS5086 has two power domains, DVDD and AVDD. Enable them both as
long as the codec is in use.
Also, switch on the power to identify the chip at device probe level,
and switch it off again afterwards. The codec level will take care for
power handling later.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There are two call sites for snd_soc_post_component_init(), one passes 0 and the
other 1 for the 'dailess' parameter of snd_soc_post_component_init(). Depending
on whether 'dailess' is 0 or 1 snd_soc_post_component_init() runs different code
at the beginning and the end of the function. The patch moves this conditional
code out of snd_soc_post_component_init() and into the call sites. This removes
the need for snd_soc_post_component_init() to know whether it is called for a
DAI link or a aux dev.
Also do the initialization of rtd->card when the rtd struct is allocated.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently in snd_soc_instantiate_card() we only check if the aux dev exists, but
do not yet assign it to its rtd. This means that we need to lookup the aux dev
again in soc_probe_aux_dev(). This patch changes the behavior to assign the aux
dev to the rtd in soc_check_aux_dev() (and renames it to soc_bind_aux_dev()).
This simplifies the implementation a bit and also removes the need for
soc_post_component_init() to know about the specific CODEC that was assigned to
the rtd. The later is necessary for componentization as the code should work for
all types of components not just CODECs. This new behavior is also more in sync
with how soc_bind_dai_link()/soc_probe_link_dais() works.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
soc_find_matching_codec() works in the same way as soc_find_codec() except that
it only works for auxdevs. It can easily be replaced by the generic
soc_find_codec().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
rtd->codec is already initialized in soc_bind_dai_link(), so there is no need to
do it again in soc_dai_link_init(). Removing the rtd->codec initialization from
soc_dai_link_init() also removes the need for soc_dai_link_init() to know about
the CODEC at all.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a minimum support of Realtek ALC5670 codec.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
R-Car sound uses Audio DMAC and Audio DMAC peri peri.
Audio DMAC peri peri transfers data inside circuit.
DMA transfer needs source / destination address,
and destination address can be set via dmaengine_slave_config().
The source address can be set when starting DMAEngine.
Because Audio DMAC peri peri always ignores its value,
current driver always used same source address for
Audio DMAC / Audio DMAC peri peri
(Audio DMAC peri peri source / destination address
is always fixed value)
But, This is not good match for DT booting.
This patch properly uses DMA start address
for Audio DMAC / Audio DMAC peri peri.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current R-Car sound SSI/SRC/DVC selection has feature limit.
(It is assuming that SSI/SRC are using same index number)
So that enabling SSI/SRC flexible selection,
this patch modifies DMA settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
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>
Since multiple codecs DAI will be usable in the future, remove
explicit unique codec_dai and cpu_dai parameters.
Replace them with snd_soc_pcm_runtime pointer that will contain
every instances.
No functionale change.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
'status' is not used in the function. Remove it.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Tested-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "list" field has been omitted from struct azx, but its
initialization remained mistakenly in hda_tegra.c, which leads to a
compile error:
sound/pci/hda/hda_tegra.c: In function 'hda_tegra_create':
sound/pci/hda/hda_tegra.c:481:22: error: 'struct azx' has no member
named 'list'
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Fixes: 9a34af4a33 ('ALSA: hda - Move more PCI-controller-specific stuff from generic code')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'private_data' is not used in the function. Remove it.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I received a report this morning from one of the Novena developers that
the behaviour of the iMX6 ASoC codec driver (using imx-pcm-dma.c) was
sub-optimal under high system load.
While there are issues relating to system load remaining, upon reviewing
the ASoC imx-pcm-dma.c driver, it was noticed that it not using the
residue support, because SDMA doesn't support it. This has the effect
that SDMA has to make multiple calls into the ASoC and ALSA code, one
for each period.
Since ALSA's snd_pcm_elapsed() does not need to be called multiple times
and it is entirely sufficient to call it once to update ALSA with the
current buffer position via the pointer method, we can do better here.
We can also avoid stopping the DMA entirely, just like real cyclic DMA
implementations behave. While this means that we replay some old samples,
this is a nicer behaviour than having audio stop and restart.
The changes to achieve this are relatively minor - imx-sdma.c can track
where the DMA is to the nearest descriptor boundary - it does this
already when deciding how many callbacks to issue. In doing this,
buf_tail always points at the descriptor which will complete next.
The residue is defined by the bytes remaining to the end of the buffer,
when the buffer is viewed as a single block of memory [start...end].
So, when we start out, there's a full buffer worth of residue, and this
counts down as we approach the end of the buffer, eventually becoming
zero at the end, before returning to the full buffer worth when we
wrap back to the start.
Moving the walking of the descriptors into the interrupt handler means
that we can update the BD_DONE flag at interrupt time, thus avoiding
a delayed tasklet stopping the cyclic DMA.
This means that the residue can be calculated from (total descriptors -
buf_tail) * descriptor size. This is what the change below does. We
update imx-pcm-dma.c to remove the NO_RESIDUE flag since we now provide
the residue.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Peach-pi board has MAX98091 CODEC. Extend snow machine driver to support
this board.
Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Reviewed-by: Doug Anderson <dianders@chromium.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The MAX98091 CODEC is the same as MAX98090 CODEC, but with an extra
microphone. Existing driver for MAX98090 CODEC already has support
for MAX98091 CODEC. Adding proper compatible string so that MAX98091
CODEC can be specified from device tree.
Signed-off-by: Wonjoon Lee <woojoo.lee@samsung.com>
Signed-off-by: Doug Anderson <dianders@chromium.org>
Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The implementation of rt5677_pll_calc() has the same logic of rl6231_pll_calc().
The only difference is the lower boundary checking for freq_in.
This patch calls rl6231_pll_calc() instead of open-coded.
The k_bp of struct rt5677_pll_code is always false, thus also remove the
code to check pll_code.k_bp.
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>
In case of S24_LE/U24_LE modes we expect 24bits on the bus while the samples
are stored and transferred in memory on 32bits (lower 3 bytes of the 4
bytes).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The codec need to be configured to 24bit mode in case of S24_LE format.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Correct the hw_params callback to configure the codec correctly in case of
S24_3LE format since in case of S24_3LE the codec has been configured to
16bit format mode.
S24_LE is not defined as supported format for the codec.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The recent commit [6194b99d: ALSA: hda - Kill the rest of snd_print*()
usages] changed the callback map_slaves(), but one call was forgotten
to be replaced due to the cast, which leads to kernel Oops due to
invalid function. This patch replaces it with a proper function.
Fixes: 6194b99de9 ('ALSA: hda - Kill the rest of snd_print*() usages')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cs4265_volatile_register reutrns a bool. The function now returns
true or false vs 1 and 0.
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current DVC can be enabled only when playback,
but, this came from misunderstanding.
It is not correct.
DVC <-> DMA relationship is...
Playback: MEM -> DMAC -> SRC -> DVC -> DMACp -> SSI
Capture: SSI -> DMACp -> SRC -> DVC -> DMAC -> MEM
DVC can be used for both Playback/Capture
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current R-Car sound driver is using DMAEngine directly,
but, ASoC is requesting to use common DMA transfer method,
like snd_dmaengine_pcm_trigger() or dmaengine_pcm_ops.
It is difficult to switch at this point, since Renesas
driver is also supporting PIO transfer.
This patch uses dmaengine_prep_dma_cyclic() instead
of dmaengine_prep_slave_single().
It is used in requested method,
and is good first step to switch over.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Sound data needs to be sent to R-Car sound SSI when playback.
But, there are 2 interfaces for it.
1st is SSITDR/SSIRDR which are mapped on SSI.
2nd is SSIn_BUSIF which are mapped on SSIU.
2nd SSIn_BUSIF is used when DMA transfer,
and it is always used if sound data came from via SRC.
But, we can use it when SSI+DMA case too.
(Current driver is assuming 1st SSITDR/SSIRDR for it)
2nd SSIn_BUSIF can be used as FIFO.
This is very helpful/useful for SSI+DMA.
But DMA address / DMA ID are not same between 1st/2nd cases.
This patch care about these settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current FSI driver is using DMAEngine directly,
but, ASoC is requesting to use common DMA transfer method,
like snd_dmaengine_pcm_trigger() or dmaengine_pcm_ops.
It is difficult to switch at this point, since Renesas
driver is also supporting PIO transfer.
This patch uses dmaengine_prep_dma_cyclic() instead
of dmaengine_prep_slave_single().
It is used in requested method,
and is good first step to switch over.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
fsi PIO/DMA handler are using each own pointer update method,
but these can be share.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current fsi driver is using SNDRV_DMA_TYPE_CONTINUOUS
for snd_pcm_lib_preallocate_pages_for_all().
But, it came from original dma-sh7760.c,
and no longer needed.
This patch exchange its parameter, and removed
original dma mapping and un-needed
dma_sync_single_xxx() from driver.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Avoid creating duplicate directories by prefixing codecs and platforms
with their separate identifiers. This avoids snd-soc-dummy (which can
appear both as a dummy platform and a dummy codec on the same card)
from clashing.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Permit ALSA to run without hardware interrupts from the audio interface.
Instead, ALSA will use a kernel timer to decide when to check the buffer
state, resulting in a lighter workload for the CPU.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is no hardware restriction requiring a minimum of 8 periods, or
a minimum of 2048 bytes in a period. Let's drop these values so that
userspace has more flexibility in choosing these parameters.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
We still see the occasional timeout waiting for busy to clear. As the
spec is contradictory, and we know that the current implementation
doesn't work, try an alternative interpretation from the spec. This
one appears to work - I have yet to find any issue with it during my
testing over several months.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
The spec requires that the mute bits must be set while the channel
is disabled. Ensure that this is the case by providing a helper
which ensures that the appropriate mute bit is set while the enable
bit is clear.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since we wish to disable capture inputs for some formats, we need to
ensure that we clear the enable bits in our cached record control
register. This seems to have been missed, resulting in the register
only accumulating enable bits.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a KIRKWOOD_RECCTL_ENABLE_MASK definition to complement the existing
PLAYCTL definition, and make use of it where we wish to clear both
enable bits.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Also add const to array
text data bss dec hex filename
10946 2904 3528 17378 43e2 sound/soc/codecs/wm0010.o-before
10891 2840 3512 17243 435b sound/soc/codecs/wm0010.o-after
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The similar fixup as T440 is needed for supporting the dock on T540.
Reported-by: Jim Minter <jminter@redhat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another quirk to make the headset mic work on some new Dell machines.
Cc: Hui Wang <hui.wang@canonical.com>
BugLink: https://bugs.launchpad.net/bugs/1297581
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ICH6_ prefix doesn't mean that it's specific to ICH6 chipset but
rather its generic for all HD-audio (or "Azalia") devices.
Use AZX_ prefix instead to align with other constants.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
