Save model information in driver_data so we can simplify the implementation.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of checking, if the work is pending, it is safer to cancel
the pending work, or wait till the scheduled work finishes.
This way we can avoid modifying the variables used by the work
function.
Since we know that no work is pending, we can remove the two additional
checks in POST_PMU, and PRE_PMD for non pending works.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
None of the driver handled by out_drv_event have it's power
bit shifted by 3.
Remove the case for shift 3, and also add comment for the cases.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Headset gain range is 0 - 0xf (4 bit resolution)
The Handsfree gain range is 0 - 0x1d (5 bit resolution,
0x1e, and 0x1f values are invalid)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is a bit overkill to have three (3) separate
workqueue for a single driver.
We can manage things with one workqueue nicely.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link")
changed DAI format flag values and we cannot simply invert anymore e.g.
frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there
is no anymore fixed bit position for bit-clock or frame-sync inversion.
Fix this by relying only on DAI format flag values passed to us and by not
making any assumption on individual bit positions.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No reason to use static variable for channel_index.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The name contains invalid valid character (/), which
causes problems when trying to create the debugfs
directory structure:
ASoC: Failed to create Aux/FM Stereo In debugfs file
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link")
changed DAI format flag values and we cannot simply invert anymore e.g.
frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there
is no anymore fixed bit position for bit-clock or frame-sync inversion.
Fix this by relying only on DAI format flag values passed to us and by not
making any assumption on individual bit positions
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Vaibhav Bedia <vaibhav.bedia@ti.com>
Cc: Sekhar Nori <nsekhar@ti.com>
Cc: Kevin Hilman <khilman@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit a810364a04
ALSA: hda - Handle -1 as invalid position, too
caused a regression on some machines that require the position-buffer
instead of LPIB, e.g. resulting in noises with mic recording with
PulseAudio.
This patch fixes the detection by delaying the test at the timing as
same as 3.0, i.e. doing the position check only when requested in
azx_position_ok().
Reported-and-tested-by: Rocko Requin <rockorequin@hotmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put parentheses around macro argument uses. This avoids pitfalls
for the programmer, where the argument expansion does not give the
expected result, for example:
SAMPLES_TO_US(substream->runtime->rate, dac33->uthr - DAC33_MODE7_MARGIN + 1);
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound_timer.info.name is a 32 character buffer. This function only
has one caller (in sound/oss/ad1848.c) and it passes as 128 character
buffer as "name". I don't know if this is a problem in real life,
and I doubt we're going to add more OSS drivers so it's unlikely to
become an issue. But we may as well take care of it.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the ugly real world, there area really broken devices that don't set
codec SSID correctly. In such a case, the ID can be random, thus the
patching won't work reliably.
For applying the patch forcibly to such a device, the driver will skip
the vendor and/or subsystem ID checks when zero or a negative number is
given in [codec] section.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new option "snoop" for the traffic control of the HD-audio
controller chip. When set to 0, the non-snooping mode is used with
the traffic control bit is set in each stream control register.
This may allow better operations in the low power mode, but the actual
implementation is depending pretty much on the chipset.
As already implemented, more or less each chipset has own snoop-control
register bit. Now this setup refers to the snoop option, too.
Also, a new VIA chipset may require the non-snooping mode when set so
in BIOS. In such a case, the option value is overridden.
As default, it's still set to snoop=1 for keeping the same behavior as
before. In near future, it'll be set to 0 as default after checking
it works in every system well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Export the default mmap function, snd_pcm_lib_default_mmap().
The upcoming non-snooping support in HD-audio driver will use this
to override the mmap method.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For almost all machines the DAI format is a constant, always set to the
same thing. This means that not only should we normally set it on init
rather than in hw_params() (where it has been for historical reasons) we
should also allow users to configure this by setting a variable in the
dai_link structure. The combination of these two will make many machine
drivers even more data driven.
Implement a new dai_fmt field in the dai_link doing just that. Since 0 is
a valid value for many format flags and we need to be able to tell if the
field is actually set also add one to all the values used to configure
formats.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently codec->control_data is not initialized before calling
process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE).
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the case of SND_SOC_DAIFMT_CBS_CFS, adau1373_dai->master should be false.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM9093 is an enhanced version of the WM9093. Add the device ID to
the driver, further patches will add support for the additional features
in the WM9093.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are numerous broken references to Documentation files (in other
Documentation files, in comments, etc.). These broken references are
caused by typo's in the references, and by renames or removals of the
Documentation files. Some broken references are simply odd.
Fix these broken references, sometimes by dropping the irrelevant text
they were part of.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Since really_cleanup_stream() is called from both purity_inactive_streams()
and hda_cleanup_all_streams(), the verbs to clear the PCM channel and
format may be called multiple times unnecessarily.
This patch adds checks to skip these unneeded verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the oscillator is always enabled and the clock output is always
disabled. This patch adds support for controlling the oscillator and clock
output state through snd_soc_dai_set_sysclk. Which makes it possible to
disable or enable them dynamically according to the requirements of the board
on which the CODEC is used.
This patch also slightly modifies the behavior as to when the oscillator is
going to be disabled in low-power states. Previously it would only be disabled
in BIAS_OFF, now it is also going to be disabled in BIAS_STANDBY, since no
components which depend on it should be active in this state.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Set the initial bias level to standby during CODEC probe instead of leaving the
CODEC powered off.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is not used outside this driver so no need to make the symbol global.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Both *socdev and *codec of struct mfld_mc_private are not being used
in this driver, remove it.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the the internal oscillator is powered down when entering BIAS_OFF
state, but not re-enabled when going back to BIAS_STANDBY. As a result the
CODEC will stop working after suspend if the internal oscillator is used to
generate the sysclock signal. This patch fixes it by clearing the appropriate
bit in the power down register when the CODEC is re-enabled.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
There are certain devices that are reportedly so slow that they need
more than 100 ms to handle control transfers. Therefore, increase the
timeout in mixer(_quirks).c to 1000 ms.
The timeout parameter of snd_usb_ctl_msg() is now constant, so we can
drop it.
Reported-by: Felipe Balbi <balbi@ti.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Handsfree gain have 2dB steps all the way, so there is no
reason to have different delays as we approaching to the
end of the scale.
The comment was also wrong, since we have 0dB at 0x3 raw, at 16 the gain
is -26dB.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Headset gain have 2dB steps all the way, so there is no
reason to have different delays as we approaching to the
end of the scale.
The comment was also wrong, since we have 0dB at 0x0 raw at
one end of the range, and not in the middle.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The delayed_work named 'delayed_work' is for the headset detection,
so move it to the twl6040_jack_data struct.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The delayed works for the output can be moved within the
twl6040_output struct (from the twl6040_data) to be better
organized.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We can manage with one set of get, and put function for the gain
controls we need to handle with custom code due to the shadowing
of the register.
For both get, and put function we can call decide based on the
mc->rreg value, if we need to call the volsw, or the vlosw_2r
variant (in 2r case rreg is not 0).
Handling of the shadow values are the same for both type of
controls.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This event handler is used with the OUT_DRV widgets.
The name pga_event was misleading.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Based on the values from twl6040 codec (HSOTRIM L/R) we can configure
the McPDM offset cancellation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The offset cancellation values can be different from board to board, even
on the same HW platform.
Provide a way for the machine drivers to configure the McPDM offset
cancellation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide API to fetch the TRIM values (for machine drivers)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update the reg_cache with values from chip regarding to TRIM.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Before clearing the probing flag in the error exit path, check that the
chip pointer is not NULL.
Signed-off-by: Thomas Pfaff <tpfaff@gmx.net>
Cc: <stable@kernel.org> [2.6.39+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The spec->autocfg.line_out_pins[] may contain the same pins as hp_pins[]
depending on the configuration. When they are identical, detecting the
line_jack_present flag screws up the auto-mute because alc_line_automute()
is called unconditionally at initialization while it won't be triggered
by unsol events, thus the old line_jack_present flag is kept for the
whole run.
For fixing this buggy behavior, the driver needs to check whether the
line-outs are really individual, and skip if same as headphone jacks.
Reference: https://bugzilla.novell.com/show_bug.cgi?id=716104
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the headphone pin is assigned as primary output to line_out_pins[],
the automatic HP-pin assignment by ASSID must be suppressed. Otherwise
a wrong pin might be assigned to the headphone and breaks the auto-mute.
Reference: https://bugzilla.novell.com/show_bug.cgi?id=716104
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
ad1980_dai is not used outside this driver,
thus drop exporting it.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sn95031_get_mic_bias() is not used outside this driver
and it is a static function now.
Thus drop exporting sn95031_get_mic_bias.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes the following compile time warning:
omap-mcbsp.c:519: warning: suggest explicit braces to avoid ambiguous ‘else’
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd-aloop driver is virtual and has no need for allocating contiguous
pages. It'll be more system-friendly to use vmalloc buffers.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is necessary to make internal speakers work on this chip.
Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/854468
Tested-by: Alex Wolfson <alex.wolfson@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM1250 EV1 is functionally digital in a system (the analogue I/O
is either ground referenced or always powered) so flag it as idle_bias_off.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM1250 EV1 has some GPIOs which can be used to control the behaviour
at runtime. Request them all if supplied and add a set_bias_level()
function to start and stop the clocks.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently we force all devices in the system to be at the same bias level.
This is due to concerns about power or pop/click impacts from either
ramping VMID or mismatching VMID on the analogue I/O lines between
connected devices but does mean we power devices up more often than we
really need to.
If a device flags idle_bias_off this will usually mean that it's either
all digital or ground referenced (in which case the idle and powered bias
levels are identical) so this concern does not apply and we can save some
power by leaving it off when not needed itself.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A supply widget is generally clearer than a MICBIAS widget and a mic bias
is just a type of supply so use a supply widget for the MICBIAS. This also
avoids confusion with the routing when connected to multiple inputs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As we've only got one audio interface and it is symmetric we can just set
SYSCLK based on the sample rate requested by the application layer. Provide
a default so bypass paths work before audio playback.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Terratec Aureon 5.1 USB sound card support is broken since kernel
2.6.39.
2.6.39 introduced power management support for USB sound cards that added
a probing flag in struct snd_usb_audio.
During the probe of the card it gives following error message :
usb 7-2: new full speed USB device number 2 using uhci_hcd
cannot find UAC_HEADER
snd-usb-audio: probe of 7-2:1.3 failed with error -5
input: USB Audio as
/devices/pci0000:00/0000:00:1d.1/usb7/7-2/7-2:1.3/input/input6
generic-usb 0003:0CCD:0028.0001: input: USB HID v1.00 Device [USB Audio]
on usb-0000:00:1d.1-2/input3
I can not comment about that "cannot find UAC_HEADER" error, but until
2.6.38 the card worked anyway.
With 2.6.39 chip->probing remains 1 on error exit, and any later ioctl
stops in snd_usb_autoresume with -ENODEV.
Signed-off-by: Thomas Pfaff <tpfaff@gmx.net>
Cc: <stable@kernel.org> [2.6.39+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add "AD198x Headphone" playback device for independent headphone playback
while playing 7.1 surround using rear panel audio jacks.
- Remove "6stack-dig-fp" model since "Headphone Playback Volume" control using
DAC0 instead of DAC1 (HDA_FRONT) was already added to all models.
- Add "Independent HP" switch to enable/disable this playback device.
When the switch is OFF, headphone use "copy front" mode to get the front
channel as the green jack.
When the switch is ON, you can play stereo sound through "AD198x Headphone"
device to headphone while playing 7.1 surround sound through "AD198x Analog"
device.
The switch cannot be changed when either "AD198x Headphone" or "AD198X Analog"
is open.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since f0fba2ad "ASoC: multi-component - ASoC Multi-Component Support",
snd_soc_register_codec() now does all the codec list and mutex init.
Thus don't need to call mutex_init(&codec->mutex) in wl1273_probe() any more.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since our irq handler has been called, it is granted, that
the reason was either PLUGINT, or UNPLUGINT.
The INTID register has been checked in the MFD part of
twl6040 driver (twl6040-irq.c).
We have no reason to read from chip again here.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
OMAP4 McPDM supports 5 downlink (playback), and
3 uplink (capture) channels.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Attempt to change McBSP CLKS source while another stream is active is not
safe after commit d135865 ("OMAP: McBSP: implement functional clock
switching via clock framework") in 2.6.37.
CLKS parent clock switching using clock framework have to idle the McBSP
before switching and then activate it again. This short break can cause a
DMA transaction error to already running stream which halts and recovers
only by closing and restarting the stream.
This goes more fatal after commit e2fa61d ("OMAP3: l3: Introduce
l3-interconnect error handling driver") in 2.6.39 where l3 driver detects a
severe timeout error and does BUG_ON().
Fix this by not changing any configuration in omap_mcbsp_dai_set_dai_sysclk
if the McBSP is already active. This test should have been here just from
the beginning anyway.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Smatch has a new check for Rosenberg type information leaks where
structs are copied to the user with uninitialized stack data in them.
The status struct has a hole in it, and on some paths not all the
members were initialized.
struct hdspm_status {
unsigned char card_type; /* 0 1 */
/* XXX 3 bytes hole, try to pack */
enum hdspm_syncsource autosync_source; /* 4 4 */
long long unsigned int card_clock; /* 8 8 */
The hdspm_version struct had holes in it as well.
struct hdspm_version {
unsigned char card_type; /* 0 1 */
char cardname[20]; /* 1 20 */
/* XXX 3 bytes hole, try to pack */
unsigned int serial; /* 24 4 */
short unsigned int firmware_rev; /* 28 2 */
/* XXX 2 bytes hole, try to pack */
int addons; /* 32 4 */
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The number of times we look at a potentially connected neighbour is just
as important as the number of times we actually recurse into looking at
that neighbour so also collect that statistic.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the DBVDD2 and DBVDD3 rails to be powered down when idle, helping
fully power down connected devices when idle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use one set of defines for the HS bits, since they are identical in both
control register.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the legacy DAI name from "twl6040-hifi" to "twl6040-legacy" to
be more intuitive.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AUX L/R outputs can be driver from the Handsfree PGA output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use "Headset XYZ" for user visible controls, while the internal DAPM
widgets can use "HS XYZ".
In this way we can group the Headset related controls in UI
(alsamixer for example).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use "Handsfree XYZ" for user visible controls, while the internal DAPM
widgets can use "HF XYZ".
In this way we can group the Handsfree related controls in UI
(alsamixer for example).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the DAPM routing for the earphone path.
Convert the DAPM_SWITCH_E to DAPM_OUT_DRV_E, so we can have correct
power up, and down sequence for EP.
Introduce mute control (Earphone Playback Switch) for users to
enable/disable the EP path.
Note: the EP does not have it's own dedicated DAC. EP is connected to
HSL DAC.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Software only shadow register to be used by the driver.
For example Earpiece path will need this shadow register.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace the string with plain NULL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the register name strings in the comments for the
twl6040_reg table, so it is easier to search for specific
register.
This is cosmetic change.
Before we had for example:
TWL6040_REG_HSLCTL as register definition.
At the register table we had:
TWL6040_HSLCTL
Searching for TWL6040_HSLCTL resulted no hits.
While if we look for REG_HSLCTL, we can find the places
the register has been used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The default gains on outputs/inputs are set to 0dB.
This is fixing the pop noise issue at the first playback, which
caused by the wrong starting point of the ramp code.
The ramp code for the outputs expects the gains to be in
their lowest configuration in order to be effective.
After the playback stops, the ramp code takes care of
ramping down the gains to their minimum.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 9676001559
("ALSA: fm801: add error handling if auto-detect fails") seems to
break systems that were previously working without a tuner.
As a bonus, this should fix init and cleanup for the case where the
tuner is explicitly disabled.
Reported-and-tested-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946
Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
Cc: stable@kernel.org [v3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9676001559
("ALSA: fm801: add error handling if auto-detect fails") added
incorrect error handling.
Once we have successfully called snd_device_new(), the cleanup
function fm801_free() will automatically be called by snd_card_free()
and we must *not* also call fm801_free() directly.
Reported-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946
Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
Cc: stable@kernel.org [v3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit [e58aa3d2: genirq: Run irq handlers with interrupts disabled],
We run all interrupt handlers with interrupts disabled
and we even check and yell when an interrupt handler
returns with interrupts enabled (see commit [b738a50a:
genirq: Warn when handler enables interrupts]).
So now this flag is a NOOP and can be removed.
Signed-off-by: Yong Zhang <yong.zhang0@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reasons for the replacement:
The current driver for McPDM was developed to support the legacy mode only.
In preparation for the ABE support the current driver stack need the be
replaced.
The new driver is much simpler, easier to extend, and it also fixes some of the
issues with the old stack.
Main changes:
- single file for omap-mcpdm (mcpdm.c/h removed)
- Define names for registers, bits cleaned up, prefixed
- Full-duplex audio operation (arecord | aplay) has been fixed
- Less code
McPDM need to be turned off after all streams has been stopped.
This might cause pop noise on the output, if the codec's DAC is
still powered at this time.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Sebastien Guiriec <s-guiriec@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to probe, and operate correctly, the OMAP McPDM driver needs to
be converted to use hwmod.
The device name has been changed to probe the driver.
Replace the clk_* with pm_runtime_* calls to manage the clocks correctly.
Missing request_mem_region/release_mem_region added.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
DMA packet_size must be configured based on the McPDM FIFO threshold
value, number of channels.
Due to the FIFO operation the DMA muse be configured differently for
playback, and capture.
At the same time fix the McPDM threshold values used for playback, and
capture to avoid broken code.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The model_id is no longer needed within the platform_data
for the TPA driver since the model of TPA specified
with the device name (tpa6130a2/tpa6140a2).
Also update rx51 (the only affected user) to use the device name rather
than platform data.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the device name and driver_data to identify
the TPA model supported by the driver.
Board files should use either "tpa6130a2" or
"tpa6140a2" as device name to specify the model
in used on the specific board.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The orginal code does not cover the case that one DAI such as codec
may be shared between other two DAIs(CPU).
When do symmetry checking, altough the codec DAI requires symmetry,
the two CPU DAIs may still be configured to run on different rates.
We change to check each DAI's state separately instead of only checking
the dai link to prevent this issue.
Signed-off-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reset the codec according to the Audio power-up delay errata for the 88PM8607.
Signed-off-by: Bas Vermeulen <bas.vermeulen@novero.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce a I2S CLK supply so playback and capture can operate independently.
Signed-off-by: Bas Vermeulen <bas.vermeulen@novero.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is not used outside this driver so no need to make the symbol global.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is not used outside this driver so no need to make the symbol global.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is not used outside this driver so no need to make the symbol global.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is not used outside this driver so no need to make the symbol global.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 873bd4c (ASoC: Don't set invalid name string to snd_card->driver
field) broke generation of a driver name for all ASoC cards relying on the
automatic generation of one. Fix this by using the old default with spaces
replaced by underscores.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
The indentation is getting a little deep. Should be straight code motion,
no functional changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the longest standing areas for improvement in ASoC has been the
DAPM algorithm - it repeats the same checks many times whenever it is run
and makes no effort to limit the areas of the graph it checks meaning we
do an awful lot of walks over the full graph. This has never mattered too
much as the size of the graph has generally been small in relation to the
size of the devices supported and the speed of CPUs but it is annoying.
In preparation for work on improving this insert a trace point after the
graph walk has been done. This gives us specific timing information for
the walk, and in order to give quantifiable (non-benchmark) numbers also
count every time we check a link or check the power for a widget and report
those numbers. Substantial changes in the algorithm may require tweaks to
the stats but they should be useful for simpler things.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since modern HDMI cards often have more than one output pin and thus
input device, we need to know which one has actually been plugged in.
This patch adds a name hint that indicates which PCM device is connected
to which pin.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Increase readability and understandability in the automute code.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM5100 is a highly integrated low power audio subsystem with advanced
digital signal processing capabilities including effects, speech clarity
enhancement and active noise cancellation. This initial driver provides
support for basic audio paths, further patches will provide more
complete functionality.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The driver assumes that control_data points to the drivers i2c_client struct,
but this is no longer the case since the ASoC core has switched to regmap.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Call platform_device_put() instead of platform_device_unregister() if
platform_device_add() fails.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This functionality is now subsumed within the bias management, using the
standard cache management functionality, without assuming the cache type.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
For digital only paths we need to make sure the bandgap is enabled prior
to starting the FLL which isn't tied into DAPM.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Add the PCM rule to allow disabling the PCM playback SRC.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the PCM rule to allow disabling the PCM playback SRC.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the PCM rules to allow disabling the PCM playback and capture SRCs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a helper function to allow drivers to disable hardware resampling
when the application has specified the SNDRV_PCM_HW_PARAMS_NORESAMPLE
flag.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The installation of the minimum period size constraint in the PCM open
callbacks was not checked for errors. Add this check, and move the call
to the beginning of the function to avoid having to do any cleanups in
the error case.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
twl6040 supports 5 playback, and 2 capture channels
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only mono audio can be used for vibra (DL4 channel).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reset the twl6040_reg array to hold the chip default values.
The only changed values were for the microphone input selection.
Select no input for the microphones in the twl6040_init_chip function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to write to the vio registers at probe time, since most
them either read only, or shared with MFD or not used.
On the other hand it is a good idea to updated the ASICREV register in
the cache at this time.
After power up we need to restore some registers. Clean up the list to
contain only the registers we are going to restore.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1. add different clkmux mode handling
SAIF can use two instances to implement full duplex (playback &
recording) and record saif may work on EXTMASTER mode which is
using other saif's BITCLK&LRCLK.
The clkmux mode could be set in pdata->init() in mach-specific code.
For generic saif driver, it only needs to know who is his master
and the master id is also provided in mach-specific code.
2. support playback and capture simutaneously however the sample
rates can not be different due to hw limitation.
Signed-off-by: Dong Aisheng <b29396@freescale.com>
Acked-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Vinod Koul <vinod.koul@linux.intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8996 can measure the impedance of accessories connected to the
headphone output. Implement initial support for this, measuring the
left channel impedance when an accessory is detected and using this
to distinguish between a line load and a headphone load.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Rather than managing the bandgap in the bias level control use a supply
widget as we only actually need to enable it for analogue paths, not
fully digital ones.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When the system has only the headphone and the line-out jacks without
speakers, the current auto-mute code doesn't work. It's because the
spec->automute_lines flag is wrongly referred in update_speakers().
This flag must be meaningless when spec->automute_hp_lo isn't set, thus
they should be always coupled.
The patch fixes the problem and add a comment to indicate the
relationship briefly.
BugLink: http://bugs.launchpad.net/bugs/851697
Reported-by: David Henningsson <david.henningsson@canonical.com>
Tested-By: Jayne Han <jayne.han@canonical.com>
Cc: stable@kernel.org (3.0)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8776 supports a continuous range of sample rates rather than
discrete values and supports a wider range of sample rates on the
playback path than is currently supported. Update the constraints on
the DAIs to reflect this.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The i2c core will clear the clientdata pointer automatically,
we don't have to set the `data' field to NULL in remove() or
if probe() failed anymore.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ambiguously named variable 'link' is used as a temporary throughout
davinci-pcm -- its presence makes grepping (and groking) the code
difficult.
Replace link with the value of link in almost all sites. The exception
is a couple places where the last-assigned link/chan needs to be
returned by a function -- in these cases, rename to last_link.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current method for machine driver to register with the ASoC core is to use
snd_soc_register_card() instead of creating a "soc-audio" platform device.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current method for machine driver to register with the ASoC core is to
use snd_soc_register_card() instead of creating a "soc-audio" platform device.
In addition we use platform_device_register_simple() to create a platform
device for the codec. This function will handle putting and deleting the
device automatically which simplifies the error handling in the machine
driver.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
To get the PCM module loaded automatically by udev et al. we need to add a
proper MODULE_ALIAS.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current method for machine driver to register with the ASoC core is to use
snd_soc_register_card() instead of creating a "soc-audio" platform device.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_pcm_format_width() to determine the sample size, instead of
checking specify sample formats and assuming that those are the only
valid format.
This change adds support for big-endian architectures (which use the _BE
formats) and the packed 24-bit format (SNDRV_PCM_FORMAT_S24_3xE).
[Fixed single letter variable name legibility problem -- broonie]
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Roland UM-ONE midi usb interface differs from Roland UM-1.
Signed-off-by: Daniele Guerrieri <d.guerrieri@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We need to count the timeout down.
Reported-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Fix below build warning:
sound/soc/blackfin/bf5xx-ad73311.c: warning: initialization from incompatible pointer type
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
clk_get() returns a pointer to the struct clk or an ERR_PTR().
This patch also use PTR_ERR() for return value.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Freescale SSI audio controller supports "synchronous" and "asynchronous"
modes. In synchronous mode, playback and capture use the same input clock,
so sample rates must be the same during simultaneous playback and capture.
Unfortunately, the code which supports asynchronous mode is just broken in
various ways. In particular, it was constraining sample sizes as well as
the sample rate.
The fix also allows us to simplify the code by eliminating the 'asynchronous',
'playback', and 'capture' variables that were used to keep track of playback
and capture streams.
Unfortunately, it turns out that simulataneous playback and record does not
actually work on the only platform that supports asynchronous mode: the
Freescale P1022DS reference board. If a second stream is started, the SSI
grinds to halt for both streams. This is true even if the P1022 is configured
for synchronous mode, so it's likely a hardware problem that needs to be
worked around.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
wait_for_avail() in pcm_lib.c has a race in it (observed in practice by an
Intel validation group).
The function is supposed to return once space in the buffer has become
available, or if some timeout happens. The entity that creates space (irq
handler of sound driver and some such) will do a wake up on a waitqueue
that this function registers for.
However there are two races in the existing code
1) If space became available between the caller noticing there was no
space and this function actually sleeping, the wakeup is missed and the
timeout condition will happen instead
2) If a wakeup happened but not sufficient space became available, the
code will loop again and wait for more space. However, if the second
wake comes in prior to hitting the schedule_timeout_interruptible(), it
will be missed, and potentially you'll wait out until the timeout
happens.
The fix consists of using more careful setting of the current state (so
that if a wakeup happens in the main loop window, the schedule_timeout()
falls through) and by checking for available space prior to going into the
schedule_timeout() loop, but after being on the waitqueue and having the
state set to interruptible.
[tiwai: the following changes have been added to Arjan's original patch:
- merged akpm's fix for waitqueue adding order into a single patch
- reduction of duplicated code of avail check
]
Signed-off-by: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No code altered at this point, simply preparing for upcoming
refactorizations.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move code from endpoint.c into a new file called stream.c and rename
functions so that their names actually reflect what they're doing.
This way, endpoint.c will be available to functions that hold all the
endpoint logic.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sort its entries in alphabetical order.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes "Surround Speaker Playback Volume" being cut off.
(Commit b4dabfc452 was probably meant to fix this, but it fixed
only the "Switch" name, not the "Volume" name.)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AIF1 channels are numbered from zero than one; do the same thing for
AIF2 too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The WM8996 only requires CPVDD when the charge pump is active so control
it separately to the other supplies, only enabling it when the charge pump
is active. This will result in a small power saving on systems which are
able to provide independent software control of the supply.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The semantics of snd_mpu401_uart_new()'s interrupt parameters are
somewhat counterintuitive: To prevent the function from allocating its
own interrupt, either the irq number must be invalid, or the irq_flags
parameter must be zero. At the same time, the irq parameter being
invalid specifies that the mpu401 code has to work without an interrupt
allocated by the caller. This implies that, if there is an interrupt
and it is allocated by the caller, the irq parameter must be set to
a valid-looking number which then isn't actually used.
With the removal of IRQF_DISABLED, zero becomes a valid irq_flags value,
which forces us to handle the parameters differently.
This patch introduces a new flag MPU401_INFO_IRQ_HOOK for when the
device interrupt is handled by the caller, and makes the allocation of
the interrupt to depend only on the irq parameter. As suggested by
Takashi, the irq_flags parameter was dropped because, when used, it had
the constant value IRQF_DISABLED.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds to support the DMA PL330 driver that uses
DMA generic API. Samsung sound driver uses DMA generic API
if architecture supports it. Otherwise, use samsung specific
S3C-PL330 API driver to transfer PCM data.
Signed-off-by: Boojin Kim <boojin.kim@samsung.com>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Cc: Jassi Brar <jassisinghbrar@gmail.com>
Cc: Liam Girdwood <lrg@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
[kgene.kim@samsung.com: removed useless variable]
Signed-off-by: Kukjin Kim <kgene.kim@samsung.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
The recursive search of widget connections in snd_hda_get_conn_index()
must be terminated at the pin and the audio-out widgets. Otherwise
you'll get "too deep connection" warnings unnecessarily.
Reported-by: Francis Moreau <francis.moro@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A fix merged in 3.1-rc2 introduced a small regression, this should get it
to build again.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The i2c core will clear the clientdata pointer automatically.
We don't have to set the `data' field to NULL in remove() or
if probe() failed anymore.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Reviewed-by: Wolfram Sang <w.sang@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The i2c core will clear the clientdata pointer automatically.
We don't have to set the `data' field to NULL in remove() or
if probe() failed anymore.
Also remove a unneeded NULL checking for kfree.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Reviewed-by: Wolfram Sang <w.sang@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- use DAC0 instead of DAC1 for Port-A Headphone
- assign 0x03 to spec->multiout.hp_nid except model="6stack-dig-fp"
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
YDSXGR_LEGACYOUTVOL is a Playback Volume control for OPL3 FM Synth.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Existing code only updates the audio delay when URBs were
submitted/retired. This can introduce an uncertainty of 8ms
on the number of samples played out with the default settings,
and a lot more when URBs convey more packets to reduce the
interrupt rate and power consumption.
This patch relies on the USB frame counter to reduce the
uncertainty to less than 2ms worst-case. The delay information
essentially becomes independent of the URB size and number of
packets. This should help applications like PulseAudio which
require accurate audio timing. Clemens Ladisch reported
a decrease of mplayer's A-V difference from nrpacks down to at
most 1ms.
Thanks to Clemens for also pointing out that the implementation
of frame counters varies between different HCDs. Only the
8 lowest-bits are used to estimate the delay.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
[clemens: changed debug code]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently register read-back for the ad193x is broken, because it expects bit 0
of the upper byte to be set to indicate a read operation, while the regmap
default for SPI is to use bit 7.
This patch also addresses another oddity of the device. There are SPI and I2C
versions of this codec. In both cases the registers are 8-bit wide and numbered
from 0x0 to 0x10, but in the SPI case there is also a so called
'global address' which is prefixed in-front of the register address. The global
address mimics I2C behaviour and includes a static device address the and the
read/write flag. This basically extends the register address to an 16-bit value
numbered from 0x800 to 0x810. These are the register numbers which are
currently used by the driver. This works, because I2C will ignore the upper
8 bits of the register, but it is still a bit confusing, as there are no such
register numbers in the I2C case.
The approach taken by this patch is to number the registers from 0x00 to 0x10
and encode the global address for SPI mode into the read and write flag masks.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Return proper error instead of 0 if clk_get fails.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_pcm_hw_constraint_integer() could return -1, in this case, sst platform is
not opened successfully. However the corresponding close callback isn't able
to be called later on to release these two allocated memories, thus resulting
in memory leak.
This patch moves the check for hardware contraints earlier, thus resolving this
issue.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the builtin snd_soc_set_runtime_hwparams() instead of assigning it by
myself.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Return -ENODEV instead of 0 if vendor id mismatch.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We don't use the step size so there's no need to work it out.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
We can directly read the FLL lock status on WM8996 so even if we don't
have an interrupt wired up we can still verify that the FLL started
successfully.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the codec_name field of the dai_link to match the actual device name
of the codec. Otherwise the card won't be instantiated.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
We need to report the entire jack state to the core jack code, not just
the bits that were being updated by the caller, otherwise the status
reported by other detection methods will be omitted from the state seen
by userspace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
In the case of no free channel available,
current implementation returns 0 instead of negative errno.
This patch fixes the logic to return -EINVAL if no free channel available.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Vinod Koul <vinod.koul@linux.intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Similarly to PLLs/FLLs some modern CODECs provide selectable system clock
sources. When the clock is the clock for a DAI we do not usually need to
identify which clock is being configured so can use clk_id for the source
clock but with CODEC wide system clocks we will need to specify both the
clock being configured and the source.
Add a source argument to the CODEC driver set_sysclk() operation to
reflect this. As this operation is not as widely used as the DAI
set_sysclk() operation the change is not very invasive. We probably
ought to go and make the same alternation for DAIs at some point.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes the following warning:
CC sound/soc/imx/imx-pcm-fiq.o
sound/soc/imx/imx-pcm-fiq.c: In function 'imx_pcm_fiq_new':
sound/soc/imx/imx-pcm-fiq.c:243: warning: unused variable 'card'
CC sound/soc/imx/imx-pcm-dma-mx2.o
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_codec_readable_register instead of open-coding it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the condition for these WARN_ONs is reversed and they are placed
before the actual check whether we are going to write to that register. So if
the codec implements the register_writable callback we'll get a warning for each
writable register when syncing the register cache.
While we are at it change the check to use snd_soc_codec_writable_register
instead of open-coding it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the default return value of snd_soc_codec_{readable,writable}_register to
true when no codec specific callback for this function is given. Otherwise all
registers of that codec will neither be readable nor writable, which is most
certainly not what we want.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have dapm_context instead of codec parameter.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace the channels_min check with a check for the relevant substream
being present. Suggested here [1] when mxs implemented the
audio-support.
[1] http://www.spinics.net/lists/arm-kernel/msg133010.html
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the channel is allocated as writecombine, then mmaping it should also
use writecombine. Also, add a proper device for the call. Ported from a
similar fix for mach-mxs.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
My first patch to ASoC ever! If I did something wrong, blame Ian.
Signed-off-by: Susan Gao <sgao@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Although the McASP supports sign-extending samples in RX or TX [1]; the
davinci-mcasp driver does not touch the {R,X}PBIT or {R,X}PAD field of the
{R,X}FMT registers meaning that the McASP will serialize the bytes it is given
regardless of their signedness. So supporting unsigned formats is as simple
as adding them to the metadata of the davinci-mcasp driver.
Update the FMTBITs reported in the snd_soc_dai_driver and also update the case
statements in davinci-mcasp's hw_params() function so that the McASP can be
connected to CODECs that use unsigned values.
[1] http://www.ti.com/lit/ug/sprufm1/sprufm1.pdf
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for Starr Labs USB MIDI devices such as the Z7S, which are
based on an FTDI serial UART chip.
Based on a patch by Daniel Mack.
Signed-off-by: Kristian Amlie <kristian@amlie.name>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PM QoS implementation files are better named
kernel/power/qos.c and include/linux/pm_qos.h.
The PM QoS support is compiled under the CONFIG_PM option.
Signed-off-by: Jean Pihet <j-pihet@ti.com>
Acked-by: markgross <markgross@thegnar.org>
Reviewed-by: Kevin Hilman <khilman@ti.com>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
Headphones has stopped working for the original reported (a regression
compared to 2.6.38). This is because Speaker and Headphones share the
same DAC, in which case no Headphones volume control was created.
This patch fixes so that both Speaker and Headphones volume
controls are created in such scenario.
BugLink: http://bugs.launchpad.net/bugs/817943
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'struct of_device' no longer exists, and its functionality has been merged
into platform_device. Update the MPC5200 audio DMA driver (mpc5200_dma)
accordingly. This fixes a build break.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
When the multi-io jacks are available, parse them first and assign DACs
before parsing speakers and headphones. This allows a better chance of
surround I/O in some desktops and laptops with limited DACs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 23c09b0090
ALSA: hda - Support multiple speakers by Realtek auto-parser
changes the return value from alc_get_line_out_pfx(), and it breaks
the center/LFE mixer split check. The caller must test with a string
"CLFE" now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce the pincfg table to patch_conexant.c for fixing up the extra
pin-configuration for auto-parser. As an example, Lenovo X200 model is
replaced with this new mechanism. (This also fixes the wrong mixer
elements for docking-station I/O in the previous model quirk
automagically.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds Kconfig to support SMDK4212.
SMDK4212 is based on samsung exynos4212 SoC.
And WM8994 is used for audio codec.
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Board file support for Ventana is not yet mainlined, and probably won't
ever be given the move to Device-Tree. Consequently, the Ventana entry
is being removed from arch/arm/tools/mach-types in the next merge window,
since it was registered over a year ago.
This will also remove function machine_is_ventana(), which is used by
the ASoC Tegra WM8903 machine driver. This will cause compilation
failures. Drop Ventana support to resolve this.
Hopefully, in the not-too-distant future, tegra_wm8903.c will be able to
configure itself from Device-Tree, and hence we'll be able to re-instate
Ventana support just by creating a .dts file for the board.
Also note that Aebl support is in a similar boat. However, that board
isn't scheduled for deprecation for at least another 5 months, and
perhaps we will have completely removed non-Device-Tree support from
tegra_wm8903.c by then and/or adjusted mach-types policy.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Joseph Pentland <jp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are references in the code to 256 sources, so I tested it with 256 aplays,
of which the first and last with real data and the rest playing /dev/zero .
Also increase amount of page tables, so the default aplay size works.
Signed-off-by: Maarten Lankhorst <m.b.lankhorst@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Suppose the ALSA card already has a number of MAX_USER_CONTROLS controls, and
the user wants to replace one, it should not fail at this condition check.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly like ALC662 asus-mode* models, rewrite the laptop-amic and
dmic models with the static pin-config tables.
Now we can get rid of all alc269_quirks.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Re-implement the asus-mode[1-8] quirks with the pin-config tables.
They are provided in case where BIOS is broken on the device, so it's
not enabled in PCI SSID lookup table. User needs to specify it via model
option explicitly if the driver doesn't work with the BIOS setup as is.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For supporting both the multiple headphones and the multiple speakers,
add the new field in struct hda_multi_out, and evaluate in the standard
setup functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let's remove the rest of ALC861 and ALC861-VD quirks.
If any breakage is found, it can be fixed easily via the pin-config
table update.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the cleanup by commit 6727b12669,
the specific setups for dallas and hp models, using VREF50 for mic pins,
were lost. Fixed now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... and add a new bit-flags argument to specify the behavior of the
function. The older function is kept as is (as a wrapper).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When multiple headphones or speakers are assigned but no individual
DACs are available, the driver should take the first HP/SPK DAC instead
of another primary output. The patch adds a bit-flag to dac field of
struct pin_dac_pair indicating that it's a slave DAC.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal states, jack_present and line_jack_present should be
updated upon unsolicited events even if no automute is set.
Otherwise the wrong state is referred when the automute behavior is
changed by the mixer control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the headphone or speaker output has no own DAC, initialize the path
using the primary DAC. Otherwise the path won't be set properly and
can result in the silence.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the test on pdev->id before the kzalloc to avoid requiring kfree when
the test fails. This fix was suggested by Wolfram Sang.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
identifier x;
expression E1!=0,E2,E3,E4;
statement S;
iterator I;
@@
(
if (...) { ... when != kfree(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
kfree(x); ... return ...; }
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Acked-by: Dong Aisheng <b29396@freescale.com>
Reviewed-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Saif needs clear clk gate first before writing registers or the write
will not success.
The original xx_get_mclk function clear clk gate after mclk setting
that may cause the former mclk setting unwork, then the real output
mclk maybe inaccurate.
Placing the clear before setting mclk to avoid such an issue.
We also have to clear clk gate in startup instead of in prepare function.
Signed-off-by: Dong Aisheng <b29396@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The first change is to add an of_node_put, since codec_np has previously
been allocated. The rest of the patch reorganizes the error handling code
so the only code executed is that which is needed.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
identifier x;
expression E1!=0,E2,E3,E4;
statement S;
iterator I;
@@
(
if (...) { ... when != of_node_put(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
of_node_put(x); ... return ...; }
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Acked-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
dma_channel_np has been accessed at this point, so decrease its reference
count before leaving the function.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
identifier x;
expression E1!=0,E2,E3,E4;
statement S;
iterator I;
@@
(
if (...) { ... when != of_node_put(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
of_node_put(x); ... return ...; }
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce a new label that includes kfree and jump to that one.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
identifier x;
expression E1!=0,E2,E3,E4;
statement S;
iterator I;
@@
(
if (...) { ... when != kfree(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
kfree(x); ... return ...; }
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
Reviewed-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adjust the goto to jump to the error handling code that includes kfree.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
identifier x;
expression E1!=0,E2,E3,E4;
statement S;
iterator I;
@@
(
if (...) { ... when != kfree(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
kfree(x); ... return ...; }
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
GFP_ATOMIC is not needed here, use GFP_KERNEL instead.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
of_parse_phandle increments the reference count of np, so this should be
decremented before trying the next possibility.
Since we don't actually use np, we can decrement the reference count
immediately.
Reported-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If devices can unconditionally support idle_bias_off let them flag it in
their driver structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
In case we have a pending completion, for example due to a problem with
the input clock which got corrected after we timed out.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This should never happen with level triggered IRQs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The WM8962 features a DSP providing a number of signal processing
features including HD Bass and Virtual Surround Sound (VSS). Enable
initial support for this, allowing users to enable and disable the
algorithms using the default coefficient sets. Further patches will
add support for runtime configuration of the DSP coefficients.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
We get called once per DAPM context but only need to run once. When DAPM
was serialized this was a series of noops but now it can run in parallel
we need to take proper care.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This occurs frequently if we are in edge triggered mode as we must poll the
interrupt status register until we get no more interrupts so it's worth
the effort - it means we skip writing null acknowledgements to the chip.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
For non-audio uses like accessory detection we can use a lower quality,
unregulated microphone bias, saving a little power. As the hardware can
manually enable and disable the biases we can select regulating mode
automatically with supply widgets connected to the biases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
There's no need for separate widgets for the enables (as the map already
shows).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Make AIFCLK supply the record paths otherwise record will not work unless
there is a simultaneous playback.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The label outnodev is only used when kzalloc has not yet taken place or has
failed, so there is no need for the call for kfree under this label.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
identifier x;
expression E1!=0,E2,E3,E4;
statement S;
iterator I;
@@
(
if (...) { ... when != kfree(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
kfree(x); ... return ...; }
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_get_conn_index() returns a negative value while the current code
stores it in an unsigned int. It must be stored in a signed integer.
Reported-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently HD-audio driver shows the all error ELD byte as an error
in the kernel message. This is annoying when the video driver doesn't
set the correct ELD from the beginning. e.g. radeon sends a zero-byte
data, but we still check ELD with the fixed 128 byte as a workaround
for some broken devices, it spews 128-times errors.
For avoiding this, the driver aborts reading when the first byte is
invalid. In such a case, the whole data is certainly invalid.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch (as1482) adds a macro for testing whether or not a
pm_message value represents an autosuspend or autoresume (i.e., a
runtime PM) event. Encapsulating this notion seems preferable to
open-coding the test all over the place.
Signed-off-by: Alan Stern <stern@rowland.harvard.edu>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
In snd_hda_parse_pin_def_config(), we checked the associated number
of speaker pins and accepts only one number exclusively. But many BIOS
seem to give different assoc number for surround speakers, thus we'd
better to accept all speaker pins no matter which assoc number, and sort
like done for the headphone pins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support of multiple speakers by Realtek auto-parser.
When all speaker pins have individual DACs, create each speaker volume
control. Otherwise, create a bind-volume control for all speaker outs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the initial check of dB-range failed due to the read error, try to
check again at the later read, too. When an invalid dB range is found,
remove TLV flags and notify the mixer info change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for testing dB range at the mixer creation time seems
to cause regressions in some devices. In such devices, reading the dB
info at probing time gives an error, thus both dBmin and dBmax are still
zero, and TLV flag isn't set although the later read of dB info succeeds.
This patch adds a workaround for such a case by assuming that the later
read will succeed. In future, a similar test should be performed in a
case where a wrong dB range is seen even in the later read.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
This patch includes below small fixes:
1. Move sta32x_set_bias_level() from sta32x_i2c_remove() to sta32x_remove().
2. Remove a redundant regulator_bulk_free() call in sta32x_i2c_remove(),
as we will call regulator_bulk_free() in sta32x_remove().
3. Remove unneeded snd_soc_codec_set_drvdata(codec, NULL) in sta32x_i2c_remove.
The i2c core will set the clientdata to NULL.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Johannes Stezenbach <js@sig21.net>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix "error: too few arguments to function 'ams_delta_set_bias_level'"
build errors in ams-delta.c that were introduced after commit d4c6005 ("ASoC:
Add context parameter to card DAPM callbacks") by adding dapm context
to ams_delta_set_bias_level calls.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The new parser may use "PCM" volume, but it was missing the vmaster
slave list, thus "Master" volume didn't control it.
Reference: https://bugzilla.kernel.org/show_bug.cgi?id=41342
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASoC core tries to not enforce symmetric rates when
two streams open simultaneously. It does so by checking
rtd->rate being zero. This works exactly once after booting
because it is not set to zero again when the streams close.
Fix this by setting rtd->rate when no active stream is left.
[This leads to lots of warnings about not enforcing the symmetry in some
situations as there's a race in the userspace API where we know we've
got two applications but don't know what rates they want to set.
-- broonie ]
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The previous patch to fsl_dma.c ("fix initialization of DMA buffers")
left behind an unused local variable that causes a build warning.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PowerPC Freescale SSI driver is claiming the IRQ when the IRQ when
the device is opened, which means that the /proc/interrupts entry for
the SSI exists only during playback or capture. This also meant that
the user won't know that the IRQ number is wrong until he tries to use
the device. Instead, we should claim the IRQ when the device is probed.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a machine driver to support the EVAL-ADAU1373 board connected to a
Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the Analog Devices ADAU1373 audio codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA doesn't really have good mechanisms for dealing with these so we just
log them - the hardware already has automatic shutdown support.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure we have access to all the advanced DSP functinality offered by the
WM8962 by running the system clock at 512fs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Implement new fixup entries for Quanta FL1 and Fujitsu Lifebook
specific COEF and pin configurations. Removed the model entries
from alc269_quirks.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the duplicated creation of capture-mixer elements for some static
ALC268 configurations. The capture mixers must be put to cap_mixer field
instead of mixers array.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently it is only possible to route one source per switch into a mixer.
This patch modifies the code, so that it is possible to route multiple sources
into a mixer via the same switch. One use-case for this is routing a stereo
channel pair into a mono-mixer via the same switch.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Pull control is availalbe for WM8994 AIF2, generally disabled as part of
the GPIO configuration in order to save power after system startup. As on
newer devices in the series there is no GPIO functionality on these pins
this will happen less naturally so have the driver disable the pulls as the
AIF is probed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is exactly one match or no match at all during the for loop iteration,
thus we can break from the for loop once a match is found.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove unused variable 'dai' to eliminate below warning.
CC sound/soc/nuc900/nuc900-pcm.o
sound/soc/nuc900/nuc900-pcm.c: In function 'nuc900_dma_new':
sound/soc/nuc900/nuc900-pcm.c:321: warning: unused variable 'dai'
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit b8c0dab9bf
"ASoC: core - PCM mutex per rtd",
the global pcm_mutex is not being used any more.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit aea170a099
"ASoC: soc-cache: Add reg_size as a member to snd_soc_codec",
the codec_drv pointer variable is not used in snd_soc_lzo_get_blksize.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit d779fce5d7
"ASoC: soc-cache: Ensure flat compression uses a copy of the defaults cache",
the codec_drv pointer variable is not used any more.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
asoc cache layer can't support this kind of spi registers well.
remove cache support and read/write registers directly
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
[This will be used by the ad193x driver to fix the fact that the
original author of the driver put a bodge for their particular chip into
a the generic ASoC register I/O abstraction layer which looked like an
obvious bug which ended up getting fixed in 3.0. Sadly there were no
comments documenting what was going on. A minimally invasive correction
to the driver is to remove the register cache support and go direct to
the hardware all the time so we're adding a new feature -- broonie]
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
system clock is 24.576MHz instead of 12.288MHz
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
dac word len value should left shift before setting
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Acked-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
fix dac word len mask and adc tdm fmt shift value
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
request_any_context_irq() returns a negative value on failure.
On success, it returns either IRQC_IS_HARDIRQ or IRQC_IS_NESTED.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.orG
Refactoring the code using snd_pcm_hw_constraint_pow2() helper function.
Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AES32 supports the non-standard 128kHZ, and this is enabled only when
SNDRV_PCM_RATE_KNOT is set in hw.rates field.
Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Xonar Essence ST/STX, the connector J14 has been confirmed to be
a digital input, so enable it in the driver.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove a field that is not used at all. This remained from
earlier tests, but the current driver has decided not to handle
iris notifications.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC260 HP models work with the BIOS auto-parser. Let's cut them off.
Also move alc260_hp_master_*() to alc262_quirks.c as these are still
referred from there.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/826081
The original reporter needs 'Headphone Jack Sense' enabled to have
audible audio, so add his PCI SSID to the whitelist.
Reported-and-tested-by: Muhammad Khurram Khan
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Newer RME cards like RayDAT and AIO support 32 samples per period. This
value is encoded as {1,1,1} in the HDSP_LatencyMask bits in the control
register.
Since {1,1,1} is also the representation for 8192 samples/period on
older RME cards, we have to special case 32 samples and 32768 bytes
according to the actual card.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Axel Lin