Initial silicon did not have master bias enabled by default, unlike
later HW, so use regmap patch to align with newer defaults.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For a sample rate of 12kHz the bclk was taken from the 44.1kHz table as
we test for a multiple of 8kHz. This patch fixes this issue by testing
for multiples of 4kHz instead.
Signed-off-by: Nikesh Oswal <Nikesh.Oswal@cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Add required tables and the binding document for ACPI and OF matching.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is quite a busy release on the driver front with a lot of new
drivers being added but comparatively quiet on the core side with only
one big change going in and that a fairly straightforward refactoring.
- Conversion of the array of DAI links to a list by Mengdong Lin,
supporting dynamically adding and removing DAI links.
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production. We really need to get to the
point where that can be done.
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come.
- New drivers for a number of Imagination Technologies IPs.
- Lots of new features and cleanups for the Renesas drivers.
- ANC support for WM5110.
- New driver for Atmel class D speaker drivers.
- New drivers for Cirrus CS47L24 and WM1831.
- New driver for Dialog DA7128.
- New drivers for Realtek RT5659 and RT56156.
- New driver for Rockchip RK3036.
- New driver for TI PC3168A
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Merge tag 'asoc-v4.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.5
This is quite a busy release on the driver front with a lot of new
drivers being added but comparatively quiet on the core side with only
one big change going in and that a fairly straightforward refactoring.
- Conversion of the array of DAI links to a list by Mengdong Lin,
supporting dynamically adding and removing DAI links.
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production. We really need to get to the
point where that can be done.
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come.
- New drivers for a number of Imagination Technologies IPs.
- Lots of new features and cleanups for the Renesas drivers.
- ANC support for WM5110.
- New driver for Atmel class D speaker drivers.
- New drivers for Cirrus CS47L24 and WM1831.
- New driver for Dialog DA7128.
- New drivers for Realtek RT5659 and RT56156.
- New driver for Rockchip RK3036.
- New driver for TI PC3168A
A collection of small driver specific fixes here, nothing that'll affect
users who don't have the devices concerned. At least the wm8974 bug
indicates that there's not too many users of some of these devices.
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Merge tag 'asoc-fix-v4.4-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.4
A collection of small driver specific fixes here, nothing that'll affect
users who don't have the devices concerned. At least the wm8974 bug
indicates that there's not too many users of some of these devices.
There is a status bit on RT5677_PLL1_CTRL2 and RT5677_PLL2_CTRL2.
That's why those registers are set volatile. However, the status
bit is currently not used by codec driver. So, it should be no
problem if we set them non-volatile.
The purpose of setting them non-volatile is to restore the setting
after a syspend/resume cycle.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As we want gpio_chip .get() calls to be able to return negative
error codes and propagate to drivers, we need to go over all
drivers and make sure their return values are clamped to [0,1].
We do this by using the ret = !!(val) design pattern.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
As we want gpio_chip .get() calls to be able to return negative
error codes and propagate to drivers, we need to go over all
drivers and make sure their return values are clamped to [0,1].
We do this by using the ret = !!(val) design pattern.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The stream is created whilst the compressed stream is opened and a
buffer is created when the DSP powers up. It is necessary at a point
once both the DSP has powered up and the the stream has been opened to
connect a stream to a buffer on the DSP. This is done in the trigger
callback as this is after the DSP has been powered and obviously the
stream must be open. Note that whilst the connect is currently trivial
it is expected that this will get more complex when support for multiple
buffers/streams per DSP is added.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add code that locates and initialises the buffer of compressed data on
the DSP if the firmware supported compressed data capture. The buffer
struct (wm_adsp_compr_buf) is kept separate from the stream struct
(wm_adsp_compr) this will allow much easier support of multiple
streams of data from the one DSP in the future, although support for
this will not be added in this patch chain.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow user-space to open a compressed stream, although no data will be
passed yet, as part of this adding the ability to define supported
capabilities per firmware and check these match the stream being opened.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Register a platform driver for the CODEC and add DAIs that will be used
to connect a compressed record path for the voice control functionality.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PLL mode based on 32KHz master clock not supported in
AB silicon so remove support from the driver.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HW can provide 1.6V micbias level as well the existing levels
already provided in the driver. This patch adds support for 1.6V
to the DT binding.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In AB silicon, the internal LDO is not supported so remove
DT and driver references to this (digital voltage direct from
'VDD' supply)
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current AB silicon, BIAS_EN field is enabled by default in the
REFERENCES register, so the regmap default value should reflect
this.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If codec probe() function fails after supplies have been enabled
it should really tidy up and disable them again. This patch updates
the probe function to do just that.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Previously Sidetone would operate only when capture to DAI was in
progress, due to DAPM path configuration. There is no reason why
this should not operate without DAI capture, so this patch updates
the DAPM path accordingly.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
fsl_ssi uses different stream names ("AC97 Playback" / "AC97 Capture")
in AC'97 mode so in this case fsl-asoc-card route map should
also be using them.
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add 8kHz, 11.025kHz, 16kHz, 22.05kHz output sample rate support.
According referance menual, "Limited support for the case when
output sampling rates is between 8kHz and 30kHz. The limitation
is the supported ratio (Fsin/Fsout) range as between 1/24 to 8."
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sometimes the audio play can not be resumed after it is
suspended. Add snd_soc_pm_ops to execute power management
operations, then this issue is fixed.
Signed-off-by: Songjun Wu <songjun.wu@atmel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add NULL test on call to devm_kzalloc.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x;
identifier fld;
@@
* x = devm_kzalloc(...);
... when != x == NULL
x->fld
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add NULL test on call to devm_kzalloc.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x;
identifier fld;
@@
* x = devm_kzalloc(...);
... when != x == NULL
x->fld
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add NULL test on call to devm_kzalloc.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x;
@@
* x = devm_kzalloc(...);
... when != x == NULL
*x
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The return type "unsigned int" was used by the ssm2518_lookup_mcs()
function even though it will eventually return a negative error code.
Improve this implementation detail by deletion of the type modifier then.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds Multi channel support on Renesas R-Car sound.
This patch is tested on Salvator-X board, but it can't use
Multi channel, because supported format is different between
codec chip and R-Car.
Thus, it was tested on board which doesn't mount codec chip,
with oscilloscope.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to the codec driver to handle mic level
detect related IRQs, and report these to user-space using a uevent
variable.
The uevent variable string "EVENT=MIC_LEVEL_DETECT" is sent to
user-space, if the mic level detect feature is enabled, and the
audio captured at the chosen mic(s) is above a certain threshold.
User-space can then handle the event accordingly (e.g. process
audio capture stream).
This method was chosen over ALSA control notification for a couple
of reasons:
1) There's no requirement here for a control to read state from.
The event is the only thing that's required and of interest.
2) tinyalsa support for control notifications does not exist so on
platforms using this over alsa-lib there is a need to add code
to support this event handling.
Another possible option would be to use the standard Jack reporting
framework but this really does not fit for this kind of event.
Finally, use of the input device framework is not being encouraged,
due to difficulties in enabling apps to access input devices, so
this has also been avoided.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
An external amp (if any) is connected to the external outputs of the SoC
of course, rather then directly to the internal amp.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a device tree match table. This serves to make the driver's support
of device tree more explicit.
Signed-off-by: Caesar Wang <wxt@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AFE is actually allowed to be turn on before configuration of DAIs
since each DAI has its own enabling control. Turn on/off AFE in
runtime resume/suspend to avoid AFE being shut down when closing a DAI
while other DAIs are still active.
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As long as I investigate SCS.1m, this model reports to transfer/receive
PCM data channels/MIDI conformant data channels in tx/rx AMDTP packet.
There's a contradiction that this model actually has no analog/digital
capture port for PCM frames and no physical MIDI ports.
I guess that SCS.1d also has the contradiction. This model has no
analog/digital ports for PCM frames and no physical MIDI ports, thus it
requires no streaming functionality.
This commit adds some modification codes to handle the contradiction,
as much as possible. Unfortunately, this module adds one PCM playback
substream for SCS.1d so as SCS.1m.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now ALSA oxfw driver gains functionalities which scs1x module has.
This commit obsoletes the scs1x module, and adds a line of MODULE_ALIAS
to load oxfw module instead of scs1x module.
In scs1x module, the name of 'shortname' field is fixed as 'SCS1x'. This
field is used to name MIDI ports for both of SCS.1m and SCS.1d. This is
not good because typically some SCS.1m and SCS.1d are used in the same
system. It's better to distinguish them according to name of the ports.
This commit applies model name in config ROM to the 'shortname'.
For the name of 'driver' and 'longname', this commit uses the same way
applied to the other models. This change may not bring disadvantages to
users because userspace applications use ALSA rawmidi or seq interface
and these interfaces are not influenced by them directly.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit copies some functions of asynchronous transactions for MIDI
playback, to merge scs1x module. The features of payload in asynchronous
transaction are the same as captured MIDI messages.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit copies some functions of asynchronous transactions for MIDI
capture, to merge scs1x module. The features of payload in asynchronous
transaction are:
* System exclusive messages for SCS.1 are encoded without ID data. In
this encoding scheme, 4 bits in LSB are available. The bits are squashed
in payload byte. Thus, one payload byte transfers two MIDI messages.
* The first byte of payload byte means:
* 0x00: depending on second payload byte
* 0xf9: including escaped system exclusive messages for SCS.1, up to
3 byte (= 6 MIDI messages)
* the others: including MIDI 1.0 messages
* the others: including escaped system exclusive messages for SCS.1, up
to 64 bytes
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When physical controls on SCS.1 models are operated, the models transfer
MIDI messages in asynchronous transactions on IEEE 1394 bus. The models
have a register to have an address for the transactions, and drivers
can register own address for this purpose.
This commit keeps a region of address, registers it and adds a handler for
the transactions.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stanton Controllers and Systems 1 (SCS.1) series is supported by ALSA
scs1x driver. This driver just supports MIDI functionality. On the other
hand, models in this series are based on OXFW971 and ALSA OXFW driver can
support them.
SCS.1 series has MIDI functionality to control its surface state such as
LED lighting. When operating physical knobs and faders, the models
generate MIDI messages. These MIDI messages are transferred by asynchronous
transactions. These transactions are really model-specific and ALSA OXFW
driver requires the functionality so as scs1x module implements.
This commit adds scs1x layer as a preparation to merge scs1x driver to
oxfw driver.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commits, some model-specific members are split from the
structure. The structure is just to keep names for compatibility to old
drivers.
This commit arranges name of the structure and localize it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, some members are moved from 'struct snd_oxfw' because
they're model-specific. There are also the other model-specific parameters
in 'struct device_info'.
This commit moves these members to model-specific structure.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, 'struct snd_oxfw' has some members for models supported by old
firewire-speakers driver, while these members are useless to the other
models.
This commit allocates new memory block and moves these members to
model-specific structure.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA oxfw driver should have backward compatibility to old
firewire-speakers driver. Additionally, in future commit, scs1x driver
will be merged. It's nice to add a pointer to have a memory block for
model-specific structures.
This commit adds a member to 'struct snd_oxfw' for this aim. Deallocation
is done at freeing ALSA card structure.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For better readability, use list_for_each_entry_continue_reverse()
in have_dup_chmap().
Signed-off-by: Geliang Tang <geliangtang@163.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this patch, internal speaker and line-out work,
but front headphone output jack stays silent on the
Mac Pro 4,1.
This code path also gets executed on the MacPro 5,1 due
to identical codec SSID, but i don't know if it has any
positive or adverse effects there or not.
(v2) Implement feedback from Takashi Iwai: Reuse
alc889_fixup_mbp_vref and just add a new nid
0x19 for the MacPro 4,1.
Signed-off-by: Mario Kleiner <mario.kleiner.de@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The suspend / resume cycle resets the settings of the FM tuner. Restore
frequency settings on resume.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In symmetry we save context first before suspend and restore it last after
resume.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case of tuner only card there is no need to take care of the codec which is
anyway absent.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If user does not supply tea575x_tuner parameter the driver tries to detect the
tuner type. The failed codec initialization is considered as FM-only card
present, however the driver still registers an IRQ handler for it.
Move codec detection earlier to set tea575x_tuner parameter before check.
Here the following functions are introduced
reset_coded() resets AC97 codec
snd_fm801_chip_multichannel_init() initializes cards with multichannel support
Fixes: 5618955c42 (ALSA: fm801: move to pcim_* and devm_* functions)
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit d7ba858a7f (ALSA: fm801: implement TEA575x tuner autodetection)
brings autodetection to the driver. However the autodetection algorithm misses
the TUNER_ONLY bit if it is supplied by the user.
Thus, user gets weird messages and no card registered.
snd_fm801 0000:0d:01.0: detected TEA575x radio type SF64-PCR
snd_fm801 0000:0d:01.0: AC'97 interface is busy (1)
snd_fm801 0000:0d:01.0: AC'97 interface is busy (1)
...
snd_fm801 0000:0d:01.0: AC'97 0 does not respond - RESET
snd_fm801 0000:0d:01.0: AC'97 interface is busy (1)
snd_fm801 0000:0d:01.0: AC'97 interface is busy (1)
snd_fm801 0000:0d:01.0: AC'97 0 access is not valid [0x0], removing mixer.
snd_fm801: probe of 0000:0d:01.0 failed with error -5
Do a copy of TUNER_ONLY bit to be applied after autodetection is done.
Fixes: d7ba858a7f (ALSA: fm801: implement TEA575x tuner autodetection)
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Cc: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no need to store struct pci_dev in struct fm801. Generic struct device
can be easily translated to struct pci_dev whenever it's needed, in particular
for one user for now.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The compiler complains on unused condition as follows
sound/pci/fm801.c: In function ‘snd_fm801_interrupt’:
sound/pci/fm801.c:585:3: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
Put the curly braces around empty body as suggested.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch introduces two new helpers fm801_iowrite16() and fm801_ioread16() to
write and read the registers by offset. Previously similar was done to access
the hardware registers by their names.
Signed-off-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise we will have a warning on ->remove() since device is a PCI one.
WARNING: CPU: 4 PID: 1411 at /home/andy/prj/linux/fs/proc/generic.c:575 remove_proc_entry+0x137/0x160()
remove_proc_entry: removing non-empty directory 'irq/21', leaking at least 'snd_fm801'
Fixes: 5618955c42 (ALSA: fm801: move to pcim_* and devm_* functions)
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is an API consolidation only. The use of kmalloc + memset to 0
is equivalent to kzalloc.
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The upstreamed code modified the control names from Mute to
Switch without changing the logic. To get audio working the Switch
needs to be off which isn't aligned with normal ALSA conventions.
Inverting the logic now so that Switch Off means mute and Switch On
means active audio using the specific volume setting.
Signed-off-by: Sebastien Guiriec <sebastien.guiriec@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
the fields channels_min, channels_max, rate and formats are
irrelevant for compressed playback, they will depend on the
content. This was probably a copy-paste mistake to have
them in the first place
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add dai links to enable additional playback stream with deeper
buffer for lower power consumption.
The normal and DEEP_buffer streams are not mutually exclusive,
content will be mixed by the DSP.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add definitions for MERR_DPCM_DEEP_BUFFER AND PIPE_MEDIA3_IN
Add relevant cpu-dai and dai link names
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
All the functionality was merged in DPCM-based driver,
keep older driver to avoid breaking userspace but
tag it as unsupported/deprecated
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge DMI quirks for various machines such as Asus T100
and clean-up code
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
first renaming and reducing delta with byt-rt5640 code before
dmi-based quirks are enabled
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
initial cleanup to use same pins
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Using the hw_fixup function in order to overwrite the default SSP
setting for Audio DSP port connected to the codec. Instead of
TDM 4ch use I2S 2ch 24 bits.
Signed-off-by: Sebastien Guiriec <sebastien.guiriec@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When pipeline is deleted, set the pipeline state to invalid state.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds clean up routine to clear the stream registers and
calls this routine before setting up stream registers.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After several open/close sai test with ctrl+c, there will be
I/O error. The SAI can't work anymore, can't recover. There
will be no frame clock. With adding the software reset in
trigger stop, the issue can be fixed.
This is a hardware bug/errata and reset is the only option.
According to the reference manual, the software reset doesn't
reset any control register but only internal hardware logics
such as bit clock generator, status flags, and FIFO pointers.
(Our purpose is just to reset the clock generator while the
software reset is the only way to do that.)
Since slave mode doesn't use the clock generator, only apply
the reset procedure to the master mode.
For asynchronous mode, TX will not be reset when RX is still
running. In this case, i can't reproduce this issue.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd driver is using complex macro to parse DAI connection.
This patch adds new rsnd_parse_connect_common() and replace current
macro to it.
This is prepare for multi channel support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
TDM will use 6 or 8 slots on 1 SSI, and Multi channel will use
6 or 8 slots on few SSI (each SSI uses 2 slots).
Thus, this adds new slot control functions which can be prepare
for Multi channel support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_get_slot_rdai() returns total slots (it returns 6 if total 6
channels) , and rsnd_get_slot_extend() returns extended SSI width
(it returns 8 if total 6 channels). This will be used on SSI multi
channel support too (It will return 2 if total 6 channels with 3 SSI).
But, it is using confusable naming.
This patch changes rsnd_get_slot_rdai() -> rsnd_get_slot(),
rsnd_get_slot_extend() -> rsnd_get_slot_width()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current Renesas sound driver is using rsnd_get_slot_runtime(), but
it is same as runtime->channels. This patch removes
rsnd_get_slot_runtime()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current SSI/SSIU are using rsnd_get_slot_runtime() to check TDM,
but using rsnd_get_slot_extend() is more sane.
This patch fix it up
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It can't output corrent dma name *before* rsnd_mod_init().
It goes to *after* rsnd_mod_init() by this patch
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound driver has rsnd_get_adinr_bit/chan() functions.
It is assuming _bit() returns ADINR :: OTBL,
and _chan() returns ADINR :: CHNUM.
Current _bit() returns both OTBL and CHNUM. This patch fixup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SSIU should be controlled after SSI. This patch fix up it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It takes three minutes to enter into hibernation on some OEM SKL
machines and we see many codec spurious response after thaw() opertion.
This is because HDA is still in D0 state after freeze() call and
pci_pm_freeze/pci_pm_freeze_noirq() don't set D3 hot in pci_bus driver.
It seems bios still access HDA when system enter into freeze state,
HDA will receive codec response interrupt immediately after thaw() call.
Because of this unexpected interrupt, HDA enter into a abnormal
state and slow down the system enter into hibernation.
In this patch, we put HDA into D3 hot state in azx_freeze_noirq() and
put HDA into D0 state in azx_thaw_noirq().
V2: Only apply this fix to SKL+
Fix compile error when CONFIG_PM_SLEEP isn't defined
[Yet another fix for CONFIG_PM_SLEEP ifdef and the additional comment
by tiwai]
Signed-off-by: Xiong Zhang <xiong.y.zhang@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This machine supports HDMI/DP ports so add these ports and its FE and BE
DAIlinks
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have WOV module which should act as DAPM sink, so add that and
its links.
Also rename the refcap to "Wake On Voice" as some user expect to
find this name
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have specific constraints for FE device (48KHz, stereo, 16
bits) and fixups for BE DMIC links (2 or 4 ch), so add those.
Also add one more FE DAIlink for dmiccap
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't support ignore suspend on few devices so remove that.
Also since we support ignore susend on PDM DMIC, add that
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DAPM map for DMIC and SSP was not properly done, so fix that up.
Also mark machine as fully routed
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds Skylake I2S machine driver which uses NAU88L25 as anlog codec and
MAX98357A as speakers
Signed-off-by: Rohit Ainapure <rohit.m.ainapure@intel.com>
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the NAU88L25 + MAX98357A machine driver entry into
the machine table
Signed-off-by: Rohit Ainapure <rohit.m.ainapure@intel.com>
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds devicetree support to the wm8974 codec driver.
With a DT-based kernel, there is no board-specific setting
to select the driver so allow it to be manually chosen.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adding ACPI ID "MX98357A" for the MAXIM 98357A amp.
Signed-off-by: Rohit Ainapure <rohit.m.ainapure@intel.com>
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add driver for the Pulse Density Modulation Interface
Controller. It comes with digitallly controlled gain,
a High-Pass and a SINCC filter.
Signed-off-by: Songjun Wu <songjun.wu@atmel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsrc-card is DPCM supported version of simple-card. Thus it has similar
DT format. OTOH, snd_soc_dai_link requests cpu/codec, but one of them
will be snd-soc-dummy in DPCM case, and DPCM requests frontend/backend
dai_link. This means it might have multi backend/codec.
And, SND_SOC_DAIFMT_xxx is based on "codec". Because of these
difference, current rsrc card can't detect correct dai_fmt.
This patch detect correct dai fmt from 1st "codec".
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
1efb53a220 ("ASoC: simple-card: Remove support for setting differing
DAI formats") removed set_fmt support from simple-card.
rsrc-card follows same style, because it is based on simple-card.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound driver will use tdm slot on TDM Multi Mode support.
This patch enables tdm slot on rsrc card driver on DT.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When power up, a "pop" is heard on line-in and mic-in.
An analysis of the PCM shows it lasts ~400ms
and looks like a filter response.
VAG power up should be delayed by 400ms as VAG power down is.
Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Older firmwares don't specify access flags for the controls,
unfortunately the usage of some of these firmware relies on being able
to read back values from the DSP. The current control code will only do
this for volatile controls. This patch will read the control from the
hardware if no flags are specified and the control is currently
enabled, which should cover these legacy use-cases.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AZX_DCAPS_POSFIX_VIA is coupled always with AZX_DRIVER_VIA type, so we
don't have to keep this bit in dcaps. Save one more!
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AZX_DCAPS_RIRB_DELAY is dedicated only for Nvidia and its purpose is
just to set a flag in bus. So it's better to be set in the toplevel
driver, either hda_intel.c or hda_tegra.c, instead of the common
hda_controller.c. This also allows us to strip this flag from dcaps,
so save one more bit there.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AZX_DCAPS_RIRB_PRE_DELAY is always tied with AZX_DCAPS_CTX_WORKAROUND,
which is Creative's XFi specific. So, we can replace it and reduce
one more bit free for DCAPS.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"data" is a u32 pointer so this copies the information to wrong place
entirely.
Fixes: 140adfba52 ('ASoC: Intel: Skylake: Add tlv byte kcontrols')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sometimes PLL1 stops working if the codec loses power
during suspend (when pow-ldo2 or reset gpio is used).
MX-7Bh(RT5677_PLL1_CTRL2) is cleared and won't be restored
by regcache since it's volatile. MX-7Bh has one status bit
and M code for PLL1. rt5677_set_dai_pll doesn't reconfigure
PLL1 after resume because it thinks the PLL params are not
changed.
This patch clears the cached PLL params at resume so that
rt5677_set_dai_pll can reconfigure the PLL after resume.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch rearranges the switch statement in arizona_calc_fratio so
that older codecs are the special cases, with the default case
applying to newer codecs (WM8998 and later). This is preferable
because it avoids having to patch new cases in every time a new
codec is added.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
These need to be signed because they hold negative error codes.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sst_memcpy32() only copied bytes/4 32bits, which means it dropped
the remaining bytes%4 bytes wrongly.
Here add copying those missing bytes, first to a 32bits tmp, and
then write the tmp to 32bits iomem.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We use ret as the return value from the rsnd_mix_probe() but if there
are no child nodes and no errors then we will never initialize ret leading
to build warnings. Ensure ret is initialized before we iterate over the
child nodes to avoid this.
Signed-off-by: Mark Brown <broonie@kernel.org>
We use ret as the return value from the rsnd_dvc_probe() but if there
are no child nodes and no errors then we will never initialize ret leading
to build warnings. Ensure ret is initialized before we iterate over the
child nodes to avoid this.
Signed-off-by: Mark Brown <broonie@kernel.org>
We use ret as the return value from the rsnd_ctu_probe() but if there
are no child nodes and no errors then we will never initialize ret leading
to build warnings. Ensure ret is initialized before we iterate over the
child nodes to avoid this.
Signed-off-by: Mark Brown <broonie@kernel.org>
Adding control elements is just for models supported by old
firewire-speakers modules. The processing should be in a function to add
model-dependent quirk.
This commit moves the codes to the function. As a result, the function
should handle error state, thus this commit also changes prototype of
the function.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, assignment to model-dependent quirk is corresponding to
asynchronous transactions on IEEE 1394 bus. This is also achieved with
device entry.
This commit changes the processing of model-dependent quirk with the
entry. As a result, the transactions are sent only for Loud models.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA OXFW driver uses AV/C Audio Subunit commands to control some models.
The commands get/set the state of Feature function block of the subunit.
The commands are not specific to OXFW, thus there's a possibility to use
them in the other drivers.
Currently, helper functions for the commands require 'struct snd_oxfw',
although, it's not necessarily required. It's better to change prototype
of the functions without the structure for future use.
This commit changes the prototype.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit renames local functions with prefix 'spkr_', so that they're
for firewire-speakers.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In ALSA firewire stack, drivers basically has no control elements. This
is due to the fact that each model has own functionality even if they use
the same communication chipset. Implementing all of the functionalities in
kernel space unreasonably increases our efforts to maintain the stack. In
most case, these functionalities can be implemented in userspace via Linux
fw character devices.
However, ALSA OXFW driver has control elements comes from old
firewire-speakers driver. Adding the elements is in a file names as
'oxfw-control.c', while the elements are really model-specific. The
name is confusing because it gives an idea to handle control elements
for all of OXFW-based models.
This commit renames the file so that it's just for models supported by
old firewire-speakers driver.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apply the same fixup for Thinkpad with dock to Thinkpad X1 Carbon 2nd,
too. This reduces the annoying loud cracking noise problem, as well
as the support of missing docking port.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=958439
Reported-and-tested-by: Benjamin Poirier <bpoirier@suse.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo Thinkpads with Realtek codecs may still have some loud
crackling noises at reboot/shutdown even though a few previous fixes
have been applied. It's because the previous fix (disabling the
default shutup callback) takes effect only at transition of the codec
power state. Meanwhile, at reboot or shutdown, we don't take down the
codec power as default, thus it triggers the same problem unless the
codec is powered down casually by runtime PM.
This patch tries to address the issue. It gives two things:
- implement the separate reboot_notify hook to struct alc_spec, and
call it optionally if defined.
- turn off the codec to D3 for Thinkpad models via this new callback
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=958439
Reported-and-tested-by: Benjamin Poirier <bpoirier@suse.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that a workaround for Thinkpad T440s crackling noise can be
applied generically to all Thinkpad models: namely, disabling the
default alc269 shutup callback. This patch moves it to the existing
alc_fixup_tpt440_dock() while also replacing the rest code with
another existing alc_fixup_disable_aamix(). It resulted in a good
code reduction.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=958439
Reported-and-tested-by: Benjamin Poirier <bpoirier@suse.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These laptops support both headphone, headset and mic modes
for the 3.5mm jack.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1526330
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Intel Atom processors seem to have a problem at recording when
bdl_pos_adj is set to an odd value. When a value like 1 is used, it
may drop the samples unexpectedly. Actually, for the old Atoms, we
used to set AZX_DRIVER_SCH type, and this assigns 32 as default.
Meanwhile the newer chips, Baytrail and Braswell, are set as
AZX_DRIVER_PCH, and the lower default value, 1, is assigned.
This patch changes the default values for these chipsets to a safer
default, 32, again. Since changing the driver type (AZX_DRIVER_XXX)
leads to the rename of the driver string, it would result in a
possible regression. So, we can't change the type. Instead, in this
patch, manual (ugly) PCI ID checks are added on top.
A drawback by this increase is the slight increase of the latency, but
it's a sub-ms order in normal situations, so mostly negligible.
Reported-and-tested-by: Jochen Henneberg <jh@henneberg-systemdesign.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just a minor cleanup; instead of passing an array, pass the assigned
bdl_pos_adj option value directory in struct azx. Also split the code
to get the default bdl_pos_adj value for the change that will follow
after this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If driver received a message that it can't handle, it won't
clear the corresponding bit and unmask interrupt, this may
lock the IRQ and DSP can't send message anymore.
To fix the issue, we should Always update IMRX after IPC.
Here we always clear the DONE/BUSY bit and unmask the IRQ
source, even when IPC failures have occurred previously.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Modified-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Avoid getting sample rate on AudioQuest DragonFly as it is unsupported
and causes noisy "cannot get freq at ep 0x1" messages when playback
starts.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AudioQuest DragonFly DAC reports a volume control range of 0..50
(0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which
is obviously incorrect and would cause software using the dB information
in e.g. volume sliders to have a massive volume difference in 100..102%
range.
Commit 2d1cb7f658 ("ALSA: usb-audio: add dB range mapping for some
devices") added a dB range mapping for it with range 0..50 dB.
However, the actual volume mapping seems to be neither linear volume nor
linear dB scale, but instead quite close to the cubic mapping e.g.
alsamixer uses, with a range of approx. -53...0 dB.
Replace the previous quirk with a custom dB mapping based on some basic
output measurements, using a 10-item range TLV (which will still fit in
alsa-lib MAX_TLV_RANGE_SIZE).
Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the
range is 0..50, so if this gets fixed/changed in later HW revisions it
will no longer be applied.
v2: incorporated Takashi Iwai's suggestion for the quirk application
method
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for PA gpio pin for controlling an external amplifier as used
on some Allwinner boards.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Rename the codec dapm widgets and routes with a _codec prefix. This is
a preparation patch for adding card dapm widgets and routes.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DPCM does not fully support symmetry attributes. soc_pcm_apply_symmetry()
is skipped in soc_pcm_open() for DPCM, without being applied elsewhere.
So HW parameters cannot be correctly limited, and user space can do
playback/capture at different rates while HW actually does not support it.
soc_pcm_params_symmetry() will return error and the second stream stops.
This patch adds soc_pcm_apply_symmetry() for FE, BE, and codec DAIs
in DPCM path that was skipped in soc_pcm_open().
Signed-off-by: PC Liao <pc.liao@mediatek.com>
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add code to unregister codec in probe function,
when the error occurs after the codec is registered.
Signed-off-by: Songjun Wu <songjun.wu@atmel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Locking is currently missing from the DSP firmware controls, which can
lead to some race conditions if the controls are accessed as the DSP
powers up or down. This patch adds them to the new power lock.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should hold the DSP power lock whilst changing the firmware since we
need to check if it is running first.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Most events around the DSP just need to be locked to ensure that the DSP
can't change power state whilst they are happening. This includes the
debugfs entries and this will make sorting the rest of the locking
simpler.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add driver for Texas Instruments pcm3168a codec
Signed-off-by: Damien.Horsley <Damien.Horsley@imgtec.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Explicitly set the transmit data level on the transceiver to 16 samples
rather then the default 0. This matches both the level set in the vendor
kernel and the (seemingly very similar) i2s engine. This fixes audio
glitches when playing back at 192k rate.
At the same time, fix a trivial typo in the TDL mask definition
Signed-off-by: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
Attempting to use this codec driver triggers a BUG() in regcache_sync()
since no cache type is set. The register map of this device is fairly
small and has few holes so a flat cache is suitable.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
These are all off by one; the playback and bypass switches are the top
two bits of the registers, which are at shifts 7 and 6 not 8 and 7.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The usb_protocol_ops structures are never modified, so declare them as
const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The condition for checking for XDAT being cleared was not correct.
Fixes: 36bcecd0a7 ("ASoC: davinci-mcasp: Correct TX start sequence")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
A couple of i915_audio_component ops have been added and accessed
directly from patch_hdmi.c. Ideally all these should be factored out
into hdac_i915.c.
This patch does it, adds two new helper functions for setting N/CTS
and fetching ELD bytes. One bonus is that the hackish widget vs port
mapping is also moved to hdac_i915.c, so that it can be fixed /
enhanced more cleanly.
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we have a new audio component ops to fetch the current ELD and
state now, we can reduce the usage of unsol event of HDMI/DP pins.
The unsol event isn't only unreliable, but it also needs the power
up/down of the codec and link at each time, which is a significant
power and time loss.
In this patch, the jack creation and unsol/jack event handling are
modified to use the audio component for the dedicated Intel chips.
The jack handling got slightly more codes than a simple usage of
hda_jack layer since we need to deal directly with snd_jack object;
the hda_jack layer is basically designed for the pin sense read and
unsol events, both of which aren't used any longer in our case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent commit [e90247f9fc: ALSA: hda - Split ELD update code
from hdmi_present_sense()] rewrote the HDMI jack handling code, but a
slight behavior change sneaked in unexpectedly. When the jack isn't
connected, it tries repoll unnecessarily.
This patch addresses the flaw, to the right behavior as before.
Fixes: e90247f9fc ('ALSA: hda - Split ELD update code from hdmi_present_sense()')
Reported-and-tested-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the recent change, HDA controller driver for Intel PCH tries to
bind i915 audio component always at the probe time no matter whether
HDMI/DP codec is found. This is, however, superflulous for old
chipsets (e.g. on IVB) where they don't have always the HDMI/DP codecs
but often have only a discrete GPU instead.
For the newer chipsets, we need already the i915 binding from the
beginning due to power well control. Meanwhile, for older chipsets
where we don't need power well, we don't need the i915 binding at the
controller level.
This patch removes again the i915 binding in the HDA controller driver
for old Intel PCHs, but adds the binding in HDMI/DP codec driver
instead. This allows still the use of the direct notification from
the graphics driver while we can avoid the unnecessary load of i915
driver for machines only with another GPU.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The audio component is enabled only when CONFIG_SND_HDA_I915 is set.
Give a dummy macro for allowing the compiler optimize out the relevant
codes when this Kconfig isn't set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is using completely the wrong mask and value when updating the
register. Since the correct values are already defined in the header,
switch to using a table with explicit constants rather than shifting the
array index.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
The previous commit removed the only use of these variables.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If we only clear the tx/rx state when both are disabled it is not
possible to start/stop one multiple times while the other is running.
Since the two are independently controlled, treat them as such and
remove the false dependency between capture and playback.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Alienware 17 (2015) has the same card and pin configuration of the
Alienware 15, so the same quirks must be applied.
Signed-off-by: Gabriele Martino <g.martino@gmx.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo Thinkpad T440s suffers from constant background noises, and it
seems to be a generic hardware issue on this model:
https://forums.lenovo.com/t5/ThinkPad-T400-T500-and-newer-T/T440s-speaker-noise/td-p/1339883
As the noise comes from the analog loopback path, disabling the path
is the easy workaround.
Also, the machine gives significant cracking noises at PM suspend. A
workaround found by trial-and-error is to disable the shutup callback
currently used for ALC269-variant.
This patch addresses these noise issues by introducing a new fixup
chain. Although the same workaround might be applicable to other
Thinkpad models, it's applied only to T440s (17aa:220c) in this patch,
so far, just to be safe (you chicken!). As a compromise, a new model
option string "tp440" is provided now, though, so that owners of other
Thinkpad models can test it more easily.
Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=958504
Reported-and-tested-by: Tim Hardeck <thardeck@suse.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The OSS sound drivers used to rely on virt_to_bus(), but don't any more,
so we can remove the Kconfig dependency.
As a lot of architectures don't provide VIRT_TO_BUS any more, removing
the dependency in sounds/oss/ would make the deprecated drivers appear
there, which we probably don't want. Instead I'm replacing the
simple dependency with 'VIRT_TO_BUS || RPC || NETWINDER' so we can
still build these sound drivers for the platforms that need them,
but don't change anything on other architectures.
As a follow-up, we can remove the virt_to_bus() implementation
and Kconfig symbol in the ARM architecture.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd-hda-intel driver tries to bind with i915 audio component always
when AZX_DCAPS_I915_POWERWELL is set in the driver caps. This was
mostly OK in the past, as the flag was applied only to a limited set
of devices, namely, Haswell and Broadwell. On these machines, i915
graphics is almost mandatory as long as HDMI/DP is concerned.
Recently the application of i915 binding was widened to more Intel
chips. On these chips, the chance of a kernel without i915 graphics
is much higher, and such user would hit an error like:
snd_hda_intel 0000:00:1b.0: failed to add i915 component master (-19)
Although the error itself is harmless, it's certainly superfluous even
to try binding with i915, if we already know that there isn't any.
This patch fixes it by simply defining AZX_DCAPS_I915_POWERWELL as 0
in the case without i915. Then all codes referring to this flag will
be optimized out by the compiler.
Fixes: 6603249dcd ('ALSA: hda - Enable audio component for old Intel PCH devices')
Reported-by: kernel test robot <ying.huang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent commit [6603249dcd: ALSA: hda - Enable audio component
for old Intel PCH devices] enabled the i915 binding for HDMI/DP on old
Intel PCHs. But many boards are without HDMI/DP, and they actually
don't need i915 binding, and yet the driver has a check of i915
binding and complains like
Haswell must be built with CONFIG_SND_HDA_I915
This error is false-positive, and it should be put only for HSW/BDW,
instead of all devices that may be bound with i915.
This patch fixes the condition to check, as well as rephrasing the
message specific to HSW/BDW HDMI/DP.
Fixes: 6603249dcd ('ALSA: hda - Enable audio component for old Intel PCH devices')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When building the da7218 driver on a 64-bit architecture, we get
a harmless warning:
sound/soc/codecs/da7218.c: In function 'da7218_of_get_id':
sound/soc/codecs/da7218.c:2261:10: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
This changes the code to use uintptr_t to ensure we have an integer
type of the same size as a pointer and won't get a warning on any
architecture.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Fixes: 4d50934abd ("ASoC: da7218: Add da7218 codec driver")
Signed-off-by: Mark Brown <broonie@kernel.org>
We get a warning for the imx-pcm-fiq driver when CONFIG_LPAE
is enabled on ARM, because dma_addr_t is 64-bit then:
sound/soc/fsl/imx-pcm-fiq.c: In function 'snd_imx_pcm_mmap':
sound/soc/fsl/imx-pcm-fiq.c:223:107: warning: format '%x' expects argument of type 'unsigned int', but argument 6 has type 'dma_addr_t {aka long long unsigned int}' [-Wformat=]
This changes the printk to use the correct format string for
printing a dma_addr_t.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Probing components can bring new DAI or DAI links based on the topology
info. This patch finds the unbound DAI links and bind them.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This function will return success immediately for a bound DAI link.
No need to look for the cpu/codec DAIs again.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A machine driver can register the two ops.
When a DAI link is added or removed by a component's topology, the
ASoC core can call the ops to notify the machine driver for extra
intialization or destruction.
E.g. topology can create FE DAI links from a cpu DAI component, and
the machine driver may define an add_dai_link ops to set machine-specific
.init ops for the DAI link.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Implement a dai link list for the soc card.
Add APIs to add/remove a DAI links dynamically, e.g. by topology.
And a dobj is embedded into the struct snd_soc_dai_link. Topology can
use the dobj to find the links created by it and remove them when the
topology component is unloaded.
The predefined DAI links are reserved to keep backward compatibility.
And they will also be added to the list.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should only add ignore suspend flag for some DAIs and not all.
This patches removes it from the DAIs where we do not support
this
It also marks the endpoints for which ignore_suspend should be
enabled
Signed-off-by: Praveen Diwakar <praveen.diwakar@intel.com>
Signed-off-by: Vunny Sodhi <vunnyx.sodhi@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some of the usecases can be marked as 'ignore_suspend' by
machine. For these on suspend we should keep audio controller
ON by saving the state and not suspending the device
For this we need to maintain a counter for these streams and be
active on suspend when such a stream is opened.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DAPM Machine map for machine was not specifying the paths
correctly.
The correct order should be:
"DMIC01 Rx" (SoC DMIC BE), connected to "DMIC AIF" (DMic Codec
AIF) and then "DMic" (DMic codec Input) connected to "SoC DMIC"
(Machine DMIC MIC Widget)
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since in Skylake we support another DAI for DMIC quad capture,
add a dailink for this as well. Also specify constrains for DMIC
FE devices and fixup for DMIC BEs
Signed-off-by: Dharageswari.R <dharageswari.r@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a module is ON then we should read the module parameters from
DSP rather than driver cached values
Signed-off-by: Omair M Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For messages which have larger payload than mailbox data, we need
to split the payload using set of messages containing mailbox
size as payload.
For sending such payload we already support LARGE_CONFIG_SET
IPCs and now to query such payload add LARGE_CONFIG_GET IPC
Signed-off-by: Mousami Jana <mousami.janax@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SKL actual mailbox size is 0x10000 and initial values were 0x800,
so update these accordingly
Signed-off-by: Omair M Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Module initialization parameter data can be set by
- INIT_INSTANCE IPC by using the default value
- SET_PARAMS immediately after INIT_INSTANCE
- SET_PARAMS data from kcontrol values set
this patch add param type to identify the parameters
has to be sent to DSP.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Mic select is a DSP module which is used to select one or many
inputs to form an output. This is useful to select data
selectively from PDM input and hence the name. This module is of
generic module type.
This patch adds support to add and configure Mic select module in
firmware topology.
Signed-off-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A module can require extra memory for processing, like audio
algorithms. The memory for these modules needs to be represented
in base module configuration and passed to DSP on init, so add
the memory pages as a field in widget data
Signed-off-by: Dharageswari.R <dharageswari.r@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A module is loaded when the path consisting the module is opened.
The module binary(ies) is loaded from file system and cached in
kernel memory for future use. This is downloaded to DSP using DMA
and invoking Load module IPCs
This patch adds support for load/unload module IPCs, DMAing
modules and manging the modules
Signed-off-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Previously we recorded "last interrupt position" and used it in
pointer callback. This is not correct implementation, and it causes
underruns when user space monitors buffer level to decide when to
send next data chunk in low latency application.
Remove position recording in IRQ handler and also hw_ptr in
struct mtk_afe_memif used to record that, and let pointer callback
reports current HW pointer instead.
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Many codecs, typically found on Realtek codecs, have the analog
loopback path merged to the secondary input of the middle of the
output paths. Currently, we don't offer the dynamic switching in such
configuration but let each loopback path mute by itself.
This should work well in theory, but in reality, we often see that
such a dead loopback path causes some background noises even if all
the elements get muted. Such a problem has been fixed by adding the
quirk accordingly to disable aamix, and it's the right fix, per se.
The only problem is that it's not so trivial to achieve it; user needs
to pass a hint string via patch module option or sysfs.
This patch gives a bit improvement on the situation: it adds "Loopback
Mixing" control element for such codecs like other codecs (e.g. IDT or
VIA codecs) with the individual loopback paths. User can turn on/off
the loopback path simply via a mixer app.
For keeping the compatibility, the loopback is still enabled on these
codecs. But user can try to turn it off if experiencing a suspicious
background or click noise on the fly, then build a static fixup later
once after the problem is addressed.
Other than the addition of the loopback enable/disablement control,
there should be no changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fix multiple spelling typos found in
various part of kernel.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
L/R channel data has been treated as inverted on R-Car sound 16bit mode,
Thus, 4689032b1("ASoC: rsnd: tidyup data align position") tidyuped data
align position. But it couldn't care about capture case. This patch
cares both playback/capture
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
An exported function snd_hda_parse_nid_path() is used only inside
hda_generic.c. Let's make it a static local function for a better
code optimization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
