commit 3c83ac2325 ("ASoC: hdac_hdmi: check error return") fixes
the static checker warning reported by Dan Carpenter:
sound/soc/codecs/hdac_hdmi.c:416 hdac_hdmi_parse_and_map_nid()
warn: unsigned 'hdac->num_nodes' is never less than zero.
But it doesn't fix the issue completely.
It's also a failure if no sub nodes found for an afg node. So modify
the return condition appropriately.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
By default the device latches data on the falling edge of the
BCLK in DSP mode, whereas the expectation for normal BCLK is to
latch on the rising edge. This updates the driver to invert the
BCLK configuration for DSP mode, to align with expected behaviour.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a device would like to use delayed suspending then PM
recommendation is to set ‘power.use_autosuspend’ flag. To allow
users to do so we need to change runtime calls in core to use
autosuspend counterparts.
For user who do not wish to use delayed suspend not setting the
device's ‘power.use_autosuspend’ flag will result in non-delayed
suspend even with these APIs which incidentally is also the default
behaviour, so only users will be impacted who opt in for this.
Signed-off-by: Sanyog Kale <sanyog.r.kale@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
By default the device latches data on the falling edge of the
BCLK in DSP mode, whereas the expectation for normal BCLK is to
latch on the rising edge. This updates the driver to invert the
BCLK configuration for DSP mode, to align with expected behaviour.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The register ASRCFG is volatile, but some bits need to be recovered
after suspend/resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The BIOS for the HP ElitePad 1000 G2 uses an unexpected HID,
(INTCCFFD), add it to the white list of knowns HIDs.
Signed-off-by: Jorge Fernandez Monteagudo <jorgefm@cirsa.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The machine driver is not loaded when the BIOS uses the 10EC5642
_HID. Add it to the white list of known _HIDs, codec_name is
already taken care of by previous commit
Tested on Asus T100TAF.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Codec name is hard-coded in machine driver, pass information
from actual ACPI HID to help support BIOS variations
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
based on bytcr-rt5640 with changes only on codec side
Quirk logic is kept as placeholder.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit 95f0980148
"ASoC: Intel: Move apci find machine routines"
introduced a regression in ACPI probe of the DPCM driver.
Fix by conditionally compiling sst-acpi when the DPCM driver
is not selected
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sound is noisy when using BCLK as reference, enable ASRC in rt5640
codec
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This provide the fix for firmware memory by freeing the pointer in driver
remove where it is safe to do so
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
IRQ reaction time is not immediate when headset putton is pressed.
This patch shortens the reaction time.
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Constify the ACPI device ID array, no need to have it writable at
runtime. Also drop the unused RT5645_INIT_REG_LEN define.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Cc: Bard Liao <bardliao@realtek.com>
Cc: Oder Chiou <oder_chiou@realtek.com>
Cc: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use dev_to_hdac_dev() and to_ehdac_device() instead of open-coding.
Signed-off-by: Geliang Tang <geliangtang@163.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dummy_timer_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>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cs5535audio_dma_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>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The atiixp_dma_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>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
commit 9b8ef9f6b3 ("ASoC: dapm: Add startup & shutdown for dai_links")
Added support for calling startup on CODEC to CODEC links, however this
is called with a NULL runtime pointer. There isn't really a sensible way
to pass a valid runtime pointer to a CODEC to CODEC link at the moment,
so we need to make the startup function safe for NULL runtimes.
This patch returns from the Arizona startup function early if there is no
runtime, this is perfectly safe as all the startup function does is set
the PCM constraints for user-space which arn't relevant to a CODEC to
CODEC link anyway.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Field usrcnt is unsigned so it cannot be lesser than zero.
The patch fixes the check, moves it to the beginning of the function
and changes return value to -EIO in case of usercnt error.
The problem has been detected using proposed semantic patch
scripts/coccinelle/tests/unsigned_lesser_than_zero.cocci [1].
[1]: http://permalink.gmane.org/gmane.linux.kernel/2038576
Signed-off-by: Andrzej Hajda <a.hajda@samsung.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A couple of call sites were missed when the snd_soc_dapm_mutex_lock
function was added this patch fixes those up.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add system clock detection to prevent output DC from SPO.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The IRQ pin will keep high when the headset button is pressed. And
keep low when the headset button is released. So, we need irq trigger
at both edges. However, some platform can't support it. Therefore,
we polling the register to report the button release event once a
button presse event is received.
To support the headset button detection function for those can't
support both edges trigger platforms, we also need to invert the
polarity of jack detection irq since we need to keep the IRQ pin
low in normal case.
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Then users can remap the keycode from userspace. If without the remap,
the input device will pass KEY_MICMUTE to userspace, but in X11 layer,
it uses KEY_F20 rather than KEY_MICMUTE for XF86AudioMicMute. After
adding the keycode map, users can remap the keycode to any value users
want.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo ThinkCenter AIO uses Line2 (NID 0x1b) to implement the
micmute hotkey, here we register an input device and use Line2 unsol
event to collect the hotkey pressing or releasing.
In the meanwhile, the micmute led is controlled by GPIO2, so we
use an existing function alc_fixup_gpio_mic_mute_hook() to control
the led.
[Hui: And there are two places to register the input device, to make
the code simple and clean, move the two same code sections into a
function.]
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
