Rename the driver, and header file from twl4030-codec to
twl4030-audio.
To avoid breakage change depending drivers at the same time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
CC: Misael Lopez Cruz <misael.lopez@ti.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
Since we now route the front DAC via aa-mix widget, adding the aa-mix
to surrounds will result in a mix-up of both front and surround PCM
signals. For avoiding this, the aa-mix routes have to be disabled
for surround paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the individual DAC is available for the headphone output, the driver
should create the DAC for its volume control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1718S, the multi-channel path should be like following:
DAC 0-->Mixer 9(index 5)-->Mixer 0(index 1)-->Front Pin;
DAC 1-->Mixer 1(index 0)-->Surround Pin;
DAC 2-->C/LFE Pin;
DAC 3-->Mixer 2(index 0)-->Side Pin;
But current code built Surround and Side path through index 1 of
Mixer 1 and 2. So Adjusting Surround and Side channel amplifier is
invalid. This patch fixes the issue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1718S, Mixer 9 doesn't expose the connection to DAC 0. So when
building up a 'PCM Playback' amplifier control, it will fail since
getting DAC 0 index of Mixer 9 returned -1. So I added a dac_mixer_idx
to indicated the actual index of DAC 0 to Mixer 9. Following is the
patch and next mail is another.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for ASoC dynamic PCM support (AKA ASoC DSP)
Platform will also support DAPM so separate out the probe function
to simplify the code (just like the codec probe).
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unmute DAC on front speaker path when Independent HP is enabled.
When to enable Independent HP, the front speaker won't output any sound
for VT1708, VT1708B, VT1708S and VT1702.
I find the via_independent_hp_put() routine will mute DAC 0 path in Mixer 0.
For these codecs, when using Independent HP, there could have two
independent streams, one is from DAC0-->Mixer0-->Front Pin, the other is
from DAC3-->GainSW3-->Side Pin.
So I added a check for DAC-->Mixer path in activate_output_path().
If current path is DAC-->Mixer, no need to mute DAC index in Mixer.
In fact, to change connection of Headphone pin or Mux connected with HP
is enough.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the re-implementation of Independent-HP mode in the
case where the DAC is shared between HP and side-channel streams.
Now the driver tries to parse the output-path using the pre-parsed
side-channel DAC for the independent HP output, too.
When a playback PCM stream is opened with this shared mode, the
Independent-HP mixer switch can't be changed for avoiding the conflict,
thus it returns -EBUSY error.
One remaining unintuitive issue is that the DAC volume is still
controlled as "Side" volume although it's shared by both independent-HP
and side streams.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The char can be unsigned on some architectures. Since the code checks
the negative values, they should be declared as signed char explicitly.
sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type
sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the wrongly converted short values:
sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type
sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More for neatness than any actual performance improvement.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
It was consistently wrong for everything except WM8993 so should be no
functional change.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.comm>
The Blackfin DMA controller can report one frame beyond the end of the
buffer in the wraparound case but ALSA requires that the pointer always
be in the buffer. Do the wraparound to handle this. A similar bug is
likely to apply to the other Blackfin PCM drivers but the code is less
obvious to inspection and I don't have a user to test.
Reported-by: Kieran O'Leary <Kieran.O'Leary@wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
I noticed that the last character of the ELD monitor name is lost,
this fixes the issue.
This fix should be confirming to the HDA spec, and works together with
the DRM part of the ELD patch.
The HDA spec does not mention that Monitor_Name_String is an '\0'
ending string, and it allows NML to be 1, which is only valid when MNL
does not count the possible ending '\0'.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move codec init verb which is only applicatable for VT1708.
I've found the root cause that jack plugged in can't be detected.
The verb in vt1708_init_verbs is used to power down jack detect circuit.
This verb is only applicable to VT1708. vt1708 didn't implement jack
detect function in hardware, so we should shut down this function to
avoid noise. But for other codecs, hardware implement jack detect
function. If sending this verb during initialization, jack detect will
be invalid. So I move this verb from via_parse_auto_config() to
patch_vt1708().
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create snd_hda_get_conn_index() helper function for obtaining the
connection index of the widget. Replaced the similar codes used in
several codec-drivers with this common helper.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a issue to enable unsolicited response to line-out pins.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a machine driver to support the EVAL-ADAV801 and EVAL-ADAV803 boards
connected to a Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the Analog Devices ADAV801 and ADAV803 audio codec.
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>
This patch updates the email address of the sound drivers supported by me to an
email account I will use on a more regular basis in the future.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we're not using the new auto parser as a fallback yet,
add it manually as a quirk.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conexant ID 506c was found on Acer Aspire 3830TG. As users report
no playback, sending to stable should be safe.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/783582
Reported-by: andROOM
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use static inline for dummy function to fix the warnings like below
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_init’:
sound/pci/hda/patch_sigmatel.c:4387:3: warning: statement with no effect
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_resume’:
sound/pci/hda/patch_sigmatel.c:4927:3: warning: statement with no effect
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More similar fixes like previous commits: handle the exceptional case
like ALC267 where no volume amp is found in ADC widget but in the
capsrc widget instead.
Also minor checks for avoiding possible erros: no connection-select
when the pin has a single selection, and add beep verbs only when the
0x1d is used for beep.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC259 seems to provide an invalid widget capability for the input-src
selector widget. The widget shows the input-amp while it's a selector,
and this confuses the current ALC882 initialization code that is used
for ALC259, too. For fixing this, check the amp capability and handle
the connection selection individually.
Also, ALC259 has no mute bit in DAC volume, so we need to initialize
it as ZERO instead of MUTE.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269 and compatible codecs have the output volume in DACs, thus we
can't use the ALC880's code as is. Fixed by checking the amp caps and
picking up the right widget for initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Look through the known NIDs that may have EAPD capabilities and turn
on/off them appropriately instead of checking the individual vendor ids.
This will also avoid the forgotten entries of newly added codec ids
in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the regression introduced by the commit
1f0f4b8036
ALSA: hda - Reduce static init verbs for Realtek auto-parsers
The input amps of mixer widgets should be unmuted as default (as
usually they have no assigned mixer switches).
More fixes in this commit are, however, for ALC260: ALC260 codec can
have multiple output mixers connnected to a single DAC althouh the
driver didn't pick up them properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Realtek codecs have the output pins hardwired with certain DACs.
These DACs have to be assigned at first and assign the rest for
multi-DAC pins so that all DACs can be assigned properly.
Without such an optimization, speaker outputs may be assigned to the
same DAC as the headphone or others.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In alc_auto_add_multi_channel_mode(), when the primary HP workaround
is enabled, it re-initializes the DAC list but calls alc662's function
in a fixed way. This isn't pretty suitable for other codecs, of course.
Now we call it with fill_dac function pointer so that the proper
function can be called at that point.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of using fixed init verbs, initialize DACs, ADCs and mixers
more dynamically for Realtek auto-parsers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add idma related register definitions to support idma.
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jaswinder.singh@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I2S registers can be used for control idma.
Previously, register is defined in i2s.c.
For sharing the registers, It is moved to i2s-regs.h
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jaswinder.singh@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add WM8994 PCM machine driver to support PCM audio
on SMDKV310, SMDKC210 boards.
Playback and Capture supports 8kHz sampling rates.
and It is tested on SMDKV310, SMDKC210.
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jaswinder.singh@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a new helper function snd_hda_get_conn_list().
Unlike snd_hda_get_connections(), this function doesn't copy the
connection-list but gives the raw pointer for the cached list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
each device has a unique mac address, which can be used to distinguish
multiple devices in the same machine. we therefore include the full mac
address in the device shortname and the last 6 bytes in the device id.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the sound/ directory there are two files (flagged by 'make
versioncheck'); sound/pci/asihpi/asihpi.c and
sound/soc/codecs/wm8991.c that include linux/version.h although they
don't need it. This patch removes the unneeded includes.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change alc_get_line_out_pfx() in patch_realtek.c to provide the channel
specific name and assign the index so that each caller doesn't have to
set the channel name by itself.
Also, check the multi-io case with the primary hp-out; for the multi-io
channels, assign the channel name instead of "Headphone" with indices.
This makes the mixer names more intuitive and reduces confusion.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some BIOS set up the pin config wrongly as line-out although it's
supposed to be a speaker out. In most cases, though, we can judge
the validity by checking the connection type -- when it's FIXED,
mostly it's an invalid line-out but a speaker.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even if the machine has no line-out but only HP-out, try to detect the
multi-io. It'll allow more possibilities for 5.1 outputs on laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't rely on the codec's channels_min information to decide wheter or
not allocate a substream's DMA buffer. Rather check if the substream
itself was allocated previously.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
SND_MXC_SOC_SSI looks to be unused, so kill it.
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8918 is register compatible with the WM8904 with a subset of the
functionality. Add the device ID, a subsequent patch will ensure that only
the relevant functionality is exported to userspace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Some devices do not support manual updates of the DC servo.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
BIOS lists the internal speaker as an internal line-out. Change to
internal speaker + model=auto for better auto-mute capabilities.
BugLink: http://bugs.launchpad.net/bugs/754964
Reported-by: Marc Legris <marc.legris@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some VIA codecs like VT1702 provide the input-route only to specific
ADCs such as digital-mic inputs. These routes aren't covered by the
normal primary ADC, and for now, user had to open the capture stream
assigned to that special ADC manually for using such inputs.
This patch implements a way to switch the current ADC dynamically per
the input-source selection in such a case. When this workaround is
activated, the driver provides only one capture stream and one input-
source control but with the full possible inputs. The driver switches
the ADC to be used (or being used) according to the input-source on the
fly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The base hardware revision of the Maxim 98095 part is 0x40; the code
which outputs the revision of the hardware has been updated to
properly use uppercase alphabetic values for the revision numbers.
Also, the use of a constant for the length 'max98095_dai' has been
replaced with ARRAY_SIZE().
Signed-off-by: Taylor Hutt <thutt@chromium.org>
Acked-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When smart51 mode is enabled, auto-mute these surround outputs
as well as the primary line-out. Also this patch includes minor
clean-ups.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unify the VT1709 10ch and 6ch parsers, as well as VT1708B 8ch and 4ch
parsers. They have no difference now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codecs like VT1708 needs more complicated routing using the mixer
widget rather than the simple selector widgets.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The surround/CLFE/side DACs on VT1708B and co have no amp but the
connected selector widgets have the amp instead. Fix the parser to
check these selector widgets for the possible mixer controls as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the check of the multiple loopback-mixer, which gave sometimes
a wrong index assigned to an element even for different names, e.g.
Mic and Front Mic. Now check the label properly for avoid duplication.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input jacks assigned as the smart51 outputs must be in the same
stack, either rear, front or other. Also, prefer line-in as the surround
to mic-in.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a issue to create playback volume control if pin has amplifier capability
but not DAC.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the order of the output-path list in a way from the DAC to the
target pin. Also now the list include the target pin, too.
Together with this format change, simplify the arguments of
parse_output_path() function, and fix the initialization in
via_auto_init_output().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop "Capture" prefix from the mic-boost names.
Otherwise some control names can overflow the max name length.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create patch_ca0132.c, to add support for devices featuring the
Creative CA0132 HD-audio codec.
This driver implements :-
* 1 playback subdevice to headphone and speaker
* 2 capture subdevices:
i - Mic-in
ii- Line-in
* mixer device
Advanced DSP features are not yet included.
Developed and maintained by Creative Labs, Inc.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create a master volume and mute control of playback for VT1718S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When switch HP independent mode, mute/unmute connctions of mixer which is
connected to headphone for VT1718S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some invalid initial verbs and correct some wrong initial verbs
for VT1718S codec.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER.
This keeps headphone automute and microphone input from operating
on at least one laptop from Opti Systems.
Without the override, the BIOS parser does a fine job setting the
card up and everything works.
Tested-By: Peter Schneider <e.at.chi.kaen@googlemail.com>
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"ret" is supposed to be signed here. The current code will only
return -EIO on error, instead of a more appropriate error code such
as -EAGAIN etc.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The reporter, who is running kernel 2.6.38, reports that
he needs to set model=auto for the headphone output to work
correctly.
BugLink: http://bugs.launchpad.net/bugs/761022
Cc: stable@kernel.org (v2.6.38+)
Reported-by: Jo
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the existing aa-loop list for simplifying the check for analog
low-current mode. Also fix the stream count test for playback streams.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Issue the init verbs of unsolicited events dynamically from the parsed
results for VIA codecs. Also, consolidate the unsol handlers for HP
and line-out mutes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly like the previous commit, initialize the input-paths dynamically
from the parsed results instead of the fixed array for VIA codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of fixed array for each codec type, initialize the output path
dynamically from the parsed results.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix races in handling of HP DAC and independent streams for VIA codecs.
Also, allow the HP output path without front-DAC, and removed
unnecessary activation of HP mixer elements.
This also removes the handling of shared side/HP stream; it's anyway
implemented in a broken way, so we need to re-implement the feature
later...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of ignoring the invalid pin configuration, return the error.
This will avoid unexpected crash, anyway.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create capture-related mixer elements dynamically from the parsed
ADCs and input-pins instead of fixed values for each codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of using the secondary substream, create an individual PCM
stream for HP-independent PCM. Otherwise it's difficult to handle
different channel numbers with multi-channel stream in the sam PCM
stream structure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VIA codecs, we shouldn't create a substream for independent HP mode,
when no individual HP DAC is found.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Parse the output-paths more dynamically, i.e. traverse the paths
from each output pin instead of fixed assignment for each codec.
Now all codecs are using the same output parser code.
The smart51 setup doesn't work with this change, and will be fixed
in the next commits.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mute the outputs via pin-controls instead of amps for the auto-mute
handling. This makes our life easier as it avoids conflict of the states
between the mixer elements and the auto-mute toggles.
With this change, we can use vmaster for the master control easily now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The jack-detect control should be created at the time of build_controls
callback instead of calling snd_hda_add_ctls() at the tree-parsing time.
For that, copy the control to the temporary array like other cases.
Also, fixed typos of vt1708_jack_detect in all places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of giving the fixed ADC list, parse the widgets and fill in
ADCs dynamically.
Also, probe the stereo-mixer input more dynamically, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently VIA driver controls the power-state of each pin per jack
detection. But, it means that the power-state mismatch may occur when
the machine doesn't give the proper jack-detection.
For avoiding this problem, a new control element "Dynamic Power-Control"
is provided so that user can turn on/off the pin-power control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previously we were using the DAPM context rather than a widget as the
argument for update_bits() so we didn't need to care that our list walk
of widgets left us one beyond the end of the list. Now we're using them
for the register update we need to make sure we're pointing at an actual
widget not the list_head.
Fix originally suggested by Liam on IM.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The Sigma code is in drivers/firmware which is only included on a very
small subset of architectures and so ends up breaking the build on
others. There's a pending patch to make the directory build as standard
but it's not merged yet.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we have changed the position_fix default for ATI and AMD
to be LPIB (see commit 50e3bbf989), we can remove the quirks that
were added for ATI chipsets.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The via driver spews warnigs like
hda-codec: no NID for mapping control Independent HP:0:0
with some codecs because snd_hda_add_nid() is called with nid=0.
This patch fixes it by skipping the call when no corresponding widget
is found.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This board has hardware switches for selecting SPI or I2C, so don't
require I2C for this driver.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit 13882a82ee (optimize iso queueing by setting
wake only after the last packet), drivers are required to call
fw_iso_context_queue_flush() after queueing a batch of packets.
The missing call would have an effect only if the controller
queue underruns, but then the DMA would stop completely.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit f2b3614cef (Don't check DMA time-out too shortly),
drivers need no longer restrict their PCM period length to be shorter
than 10 seconds.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed remaining issues of the signedness bug discovered by Dan Carpenter.
A check was remaining that tests if unsigned rt->rate is >= 0.
Changed that so that rt->rate now consistently uses ARRAY_SIZE(rates)
as invalid rate value and not -1.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use less specific names for suspend/resume to match the probe/remove funcs
where these are now used.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Scott Jiang <scott.jiang@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The codec name should not have a "-codec" suffix since this is not part of
a MFD. This was incorrectly changed during the multi-component updated.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The only thing the init func does is register a spi driver, so if that
fails, we return the value back up to the caller who will display an
error message for us. So drop the redundant checking/message.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a machine driver to support the ADAU1701 SigmaDSP processors on
Analog Devices BF5XX evaluation boards.
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>
This patch adds support for the Analog Devices ADAU1701 SigmaDSP.
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>
I no longer work at Bluewater Systems. Update my email address accordingly. I
have deleted my email address from C files rather than change it. This
was suggested by several people, since the commit from my new email
address will cause scripts/get_maintainer.pl to function properly. I
have not added the .mailmap entry as suggested by Joe because I think
it is no longer necessary if I touch all the files which had my name
in them.
Signed-off-by: Ryan Mallon <rmallon@gmail.com>
Cc: Andre Renaud <andre@bluewatersys.com>
Cc: H Hartley Sweeten <hsweeten@visionengravers.com>
Cc: Russell King <linux@arm.linux.org.uk>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Andrew Victor <avictor.za@gmail.com>
Cc: David Woodhouse <dwmw2@infradead.org>
Cc: Anton Vorontsov <cbou@mail.ru>
Cc: Paul Mundt <lethal@linux-sh.org>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Cc: Joe Perches <joe@perches.com>
Cc: Jesper Juhl <jj@chaosbits.net>
Cc: Andrew Morton <akpm@linux-foundation.org>
Cc: trivial@kernel.org
Cc: linux-kernel@vger.kernel.org
Reviewed-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
This is a reworked patch from Creative to change the PLL code to address
unreliable 44100Hz initialization.
Signed-off-by: Harry Butterworth <heb1001@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This time with soc_widget_update_bits reflecting recent soc_update_bits changes.
Currently widget IO is tightly coupled to the CODEC drivers. Future platform DSP
devices have mixer components that can alter power usage and hence require full
DAPM support.
This provides a generic widget IO operation wrapper in preparation for
future patches that implement platform driver DAPM.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Kill tasklet usage in rawmidi core code. Use workq for the event callback
instead of tasklet (which is used only in core/seq/seq_midi.c).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a partial revert of 28f65c11f2 ("treewide: Convert uses of
struct resource to resource_size(ptr)") as the code is rewritten
in the sound tree and thus the change is obsolete.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Initialise model-specific DAC and ADC parts.
Add controls for output and mic source selection.
Rename some mixer controls according to ControlNames.txt.
Remove Playback switches for Line-in and IEC958-in - these
were controlling the input mute/unmute which affected
capture too. Use the capture switches to control the
input mute/unmute instead - it's less confusing.
Initialise the WM8775 to invert the left-right clock
to swap the left and right channels of the mic and aux
input.
Signed-off-by: Harry Butterworth <heb1001@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have a double-free bug in
sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload().
We already call release_firmware(fw) on line 258, so when we then do it
again after usb6fire_fw_ezusb_write() returns <0, we have a double-free.
Easily fixed by just removing the last call to release_firmware().
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will be removed in -next so let's drop it from mainline as soon as
we can in order to minimise surprises.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to facilitate merging with the register map I/O replace the use
of control_data for the bulk writes with direct lookup of the client data
from the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Normally DAPM will power up any connected audio path. This is not ideal
for sidetone paths as with sidetone paths the audio path is not wanted in
itself, it is only desired if the two paths it provides a sidetone between
are both active. If the sidetone path causes a power up then it can be
hard to minimise pops as we first power up either the sidetone or the main
output path and then power the other, with the second power up potentially
introducing a DC offset.
Address this by introducing the concept of a weak path. If a path is marked
as weak then DAPM will ignore that path when walking the graph, though all
the relevant controls are still available to the application layer to allow
these paths to be configured.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
For clarity and to help ongoing refactoring in this area create a new file
to contain the physical I/O functions, separating them out from the cache
operations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
If given a -1 cmd parameter then make_exec_verb() returns -1 without
setting the res output value.
Prior to this change snd_hda_codec_read() assumed that make_exec_verb()
unconditionally set res regardless of the cmd value.
This change explicitly checks the make_exec_verb() return value before
consuming the potentially unset res value.
Signed-off-by: Greg Thelen <gthelen@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a whole bunch of functions which just call straight through to
do_hw_read(). Simplify this situation by removing them and using hw_read()
directly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Using static inline functions can reduce compilation messages
and macro misuse.
sound/pci/hda/patch_conexant.c: In function ‘patch_cxt5045’:
sound/pci/hda/patch_conexant.c:1232:3: warning: statement with no effect
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-mute setup for Acer Aspire-one with ALC268 was set wrongly
during the clean-up of auto-mute function. Fixed now.
Tested-by: Borislav Petkov <bp@alien8.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the necessary details to support the PCIe version of
E-MU's 0404 card.
From comparing the PCBs it seems the PCIe version just added a PCIe
chipset and left all other components pretty much in place.
For anyone intrigued to take a look at the PCB there are pictures I took
at <http://babelmonkeys.de/~florob/E-MU%200404/>.
Signed-off-by: Florian Zeitz <florob@babelmonkeys.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI version of the RME HDSP MADI card uses 0xcf as revision ID. Just
add this to the list of supported cards.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When using Word Clock on RME MADI cards, AutoSync mode was alternating
betweeen MADI and WC due to a typo: AutoSync is indicated in the second
status register (status2), not the first one (status).
While the proc output was always correct, the reported WC frequency to
ALSA was unstable as mentioned in
http://mailman.alsa-project.org/pipermail/alsa-devel/2008-March/006723.html
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the MIDI part, we need to acquire (and release) the hmidi->lock,
access to the global hdspm structure is serialized through
hmidi->hdspm->lock instead.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The name argument of request_irq() appears in /proc/interrupts, and
it's quite ugly when the name entry contains a space or special letters.
In general, it's simpler and more readable when the module name appears
there, so let's replace all entries with KBUILD_MODNAME.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/761171
The original reporter needs the model=auto quirk for his internal
speakers to be audible in the latest daily snapshot, so add an entry in
the quirk table for his PCI SSID.
A trivially different version of this patch using the model=asus quirk
should be applied to the 2.6.38 and 2.6.39 stable kernels. We don't use
the asus quirk in 3.0-rc2, because 3.0-rc2's autoparser is much
improved.
Reported-and-tested-by: tomdeering7
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The convention for pci_driver.name entry in kernel drivers seem to be
the module name or equivalent ones. But, so far, almost all PCI sound
drivers use more verbose name like "ABC Xyz (12)", and these are fairly
confusing when appearing as a file name.
This patch converts the all pci_driver.name entries in sound/pci/* to
use KBUILD_MODNAME for more unified appearance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up snd_printk() helper using the %pV prefix for recursive printks.
This also automagically fixes an Oops with RO/NX-enabled modules.
Tested-by: Maarten Lankhorst <m.b.lankhorst@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Reatlek model quirks use master_mute bool switch for controlling
the master-mute of outputs. For these cases, the initialization of HP
pins/amps were forgotten during the transition to the common automute
helper function in 3.0 development time, and resulted in the muted HP
output as default.
This patch fixes the issue by adjusting the HP output explicitly with
master_mute switch.
Tested-by: Michal Hocko <mhocko@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The tag number was forgotten to be fixed after cleaning up the model
quirks for ALC262 fujitsu and lenovo-3000 models.
Tested-by: Michal Hocko <mhocko@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SSYNC register was once defined as 0x34-37 in the old Intel datasheet,
but corrected later to 0x38-3b. For fixing the register usage, a new
bit-flag is introduced for indicating the old ICH SSYNC register, and
ICH* PCI entries are added explicitly to enable this quirk.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Several fixes as well where the +1 was missing.
Done via coccinelle scripts like:
@@
struct resource *ptr;
@@
- ptr->end - ptr->start + 1
+ resource_size(ptr)
and some grep and typing.
Mostly uncompiled, no cross-compilers.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Some HP laptops with AD1981 have SPDIF connections, but currently the
driver disables it statically. Better to check the pin default config
to judge whether to enable or disable the SPDIF.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If DMA active status should be checked, I2SCON register should be referenced.
In this patch, Fix the incorrect referencing of I2SCON register.
Reported-by : Lakkyung Jung <lakkyung.jung@samsung.com>
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Make sure we follow naming convention for all PCM ops.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for the new ASoC Dynamic PCM support (AKA DSP support).
The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed
at runtime between the PCM device end (or Frontend - FE) and the physical DAI
(Backend - BE) using regular kcontrols (just like a hardware CODEC routes
audio in the analog domain). The Dynamic PCM core therefore must be
able to call PCM operations for both the Frontend and Backend(s) DAIs at
the same time.
Currently we have a global pcm_mutex that is used to serialise
the ASoC PCM operations. This patch removes the global mutex
and adds a mutex per RTD allowing the PCM operations to be reentrant and
allow control of more than one DAI at at time. e.g. a frontend PCM hw_params()
could configure multiple backend DAI hw_params() with similar or different
hw parameters at the same time.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for Dynamic PCM support (AKA DSP support).
There will be future patches that add support to allow PCMs to be dynamically
routed to multiple DAIs at startup and also during stream runtime. This patch
moves the ASoC core PCM operaitions into a new file called soc-pcm.c. This will
in simplify the ASoC core features into distinct files.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently it is possible that snd_soc_new_{mixer,mux,pga} is called with a
DAPM context not matching the widgets context. This can lead to a wrong
prefix_len calculation, which will result in undefined behaviour. To avoid
this always use the DAPM context from the widget itself.
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
Device tree integer properties are encoded in big-endian format, but some of
the Freescale ASoC drivers were assuming that the host is in big-endian format
as well. Although this is true, it's better to use endian-safe accessors.
Also add a check for a failed ioremap() call in the SSI driver.
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 DMA (PCM) driver used by some Freescale PowerPC supports separate DAIs
for playback and capture, so DMA buffers should be allocated only for the
initialized streams. Instead of checking for the number of active channels,
which apparently is not reliable, check to see if the actual stream object
exists.
Also provide a better name for the DMA interrupt.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we have the EP93xx DMA engine driver in place, we convert the ASoC
drivers (I2S, AC97 and PCM) to take advantage of this new API. There are no
functional changes.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Acked-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Don't require an audio rate SYSCLK in hw_params() in order to better
support microphone detection use cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We really should be getting the interrupt - if we don't get one it's very
likely that the configuration is incorrect and audio will fail. Also
increase the timeout substantially in this case for safety.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The chip can actually support SPI so we shouldn't assume we've got an I2C
device even though that's the most common configuration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Currently the rbtree code will write out the entire register map when
doing a cache sync which is wasteful and will slow things down. Check
to see if the value we're about to write is the default and don't bother
restoring it if it is, either the value will have been retained or the
device will have been reset and holds the value already.
We should really store the defaults in the nodes but this resolves the
immediate issue.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Some ASoC components depend on other ASoC components to provide clocks and
power resources in order to probe() and vice versa for remove().
Allow components to be ordered so that components can be probed() and removed()
in sequences that conform to their dependencies.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently pcm_new() passes in 3 arguments :- card, pcm and DAI.
Refactor this to only pass in 1 argument (i.e. the rtd) since struct rtd contains
card, pcm and DAI along with other members too that are useful too.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of checking the azx_dev index with a fixed number (4), check
the stream direction of the assigned substream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When reading from the position-buffer results in -1, handle as it's
invalid and falls back to LPIB mode as well as 0.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
Commit f97d0c6d5f ("ASoC: AD1836: Add input gain control for ADC2") contained
a typo in the register name, causing a build error. This patch fixes 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>
removing unnecessary if(ret) checks
This updated patch corrects a minor spelling problem in the commit message
and resolves two other (similar) issues found in wm8940.c by Jonathan Cameron.
Signed-off-by: Greg Dietsche <Gregory.Dietsche@cuw.edu>
Acked-by: Jonathan Cameron <jic23@cam.ac.uk>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: https://launchpad.net/bugs/792712
The original reporter states that sound from the internal speakers is
inaudible until using the model=auto quirk. This symptom is due to an
existing quirk mask for 0x102802b* that uses the model=dell quirk. To
limit the possible regressions, leave the existing quirk mask but add
a higher priority specific mask for the reporter's PCI SSID.
Reported-and-tested-by: rodni hipp
Cc: <stable@kernel.org> [2.6.38+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AD1836 has a PGA for its second ADC. This patch adds a control for
adjusting the the gain of the PGA.
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>
The control_type field is never used, so it can be removed. 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: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The AD183X codec devices are mostly register compatible and can easily be
supported by the same driver. The main difference between those devices
is the number of DACs and ADCs.
This patch adjusts the driver to allocate the controls, DAPM widgets and
routes for the DACs and ADCs dynamically based on the chip type.
The AD1836 is a bit special in that it supports different modes for its second
ADC, so it needs some special handling. Right now the driver hardcodes the mode
to the differential PGA mode.
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>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write.
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>
The different ADC and DAC controls follow the same scheme, so add some helper
macros for declaring them.
This should make the code a bit more readable and also decreases the code size
a bit.
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>
Now that the CODEC driver supports it defer configuration of the system
clock until bias management which is a much more idiomatic place to do
system power control and makes things a lot more happy when we're using
both interfaces.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This allows the card driver to use the bias level variable more easily in
multi component systems.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The card callback will get called for each DAPM context in the card so it
can be useful for it to know which device is currently undergoing a
transition.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
It's redundant now thanks to the use of the generic trace infrastructure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
If the only widgets active within a CODEC are supplies and micbiases we
are not passing audio, we are probably just doing microphone detection.
This will not generally require either fully accurate reference voltages
or much power so
If this turns out to be unsuitable for some systems we can provide a
facility to override this decision.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Rather than a simple flag to say if we want the DAPM context to be at full
power specify the target bias state. This should have no current effect
but is a bit more direct and so makes it easier to change our decisions
about the which bias state to go into in future.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow more dynamic management of the device clocking by allowing BCLK to
be calculated when we set SYSCLK. This means that if the system is idle
when hw_params() runs then we don't try to use the SYSCLK used in that case
to set up the BCLK dividers, we can instead wait until a later point such
as bias level configuration. This makes it easier to manage low power modes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Avoids issues if someone does a read followed by restore and doesn't mask
out only the bits being updated.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When the FLL locks on the WM8915 an interrupt is generated. For safety
error out if we don't get that interrupt when the IRQ output of the
WM8915 is hooked up. Since we *really* expect an interrupt but the
threaded IRQ handler may take a bit longer than expected to get
scheduled also dramatically increase the delay in this case.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The general concept of this change is to create a PCM device for each
pin widget instead of each converter widget. Whenever a PCM is opened,
a converter is dynamically selected to drive that pin based on those
available for muxing into the pin.
The one thing this model doesn't support is a single PCM/converter
sending audio to multiple pin widgets at once.
Note that this means that a struct hda_pcm_stream's nid variable is
set to 0 except between a stream's open and cleanup calls. The dynamic
de-assignment of converters to PCMs occurs within cleanup, not close,
in order for it to co-incide with when controller stream IDs are
cleaned up from converters.
While the PCM for a pin is not open, the pin is disabled (its widget
control's PIN_OUT bit is cleared) so that if the currently routed
converter is used to drive a different PCM/pin, that audio does not
leak out over a disabled pin.
We use the recently added SPDIF virtualization feature in order to
create SPDIF controls for each pin widget instead of each converter
widget, so that state is specific to a PCM.
In order to support this, a number of more mechanical changes are made:
* s/nid/pin_nid/ or s/nid/cvt_nid/ in many places in order to make it
clear exactly what the code is dealing with.
* We now have per_pin and per_cvt arrays in hdmi_spec to store relevant
data. In particular, we store a converter's capabilities in the per_cvt
entry, rather than relying on a combination of codec_pcm_pars and
the struct hda_pcm_stream.
* ELD-related workarounds were removed from hdmi_channel_allocation
into hdmi_instrinsic in order to simplifiy infoframe calculations and
remove HW dependencies.
* Various functions only apply to a single pin, since there is now
only 1 pin per PCM. For example, hdmi_setup_infoframe,
hdmi_setup_stream.
* hdmi_add_pin and hdmi_add_cvt are more oriented at pure codec parsing
and data retrieval, rather than determining which pins/converters
are to be used for creating PCMs.
This is quite a large change; it may be appropriate to simply read the
result of the patch rather than the diffs. Some small parts of the change
might be separable into different patches, but I think the bulk of the
change will probably always be one large patch. Hopefully the change
isn't too opaque!
This has been tested on:
* NVIDIA GeForce 400 series discrete graphics card. This model has the
classical 1:1:1 codec:converter:pcm widget model. Tested stereo PCM
audio to a PC monitor that supports audio.
* NVIDIA GeForce 520 discrete graphics card. This model is the new
1 codec n converters m pins m>n model. Tested stereo PCM audio to a
PC monitor that supports audio.
* NVIDIA GeForce 400 series laptop graphics chip. This model has the
classical 1:1:1 codec:converter:pcm widget model. Tested stereo PCM,
multi-channel PCM, and AC3 pass-through to an AV receiver.
* Intel Ibex Peak laptop. This model is the new 1 codec n converters m
pins m>n model. Tested stereo PCM, multi-channel PCM, and AC3 pass-
through to an AV receiver.
Note that I'm not familiar at all with AC3 pass-through. Hence, I may
not have covered all possible mechanisms that are applicable here. I do
know that my receiver definitely received AC3, not decoded PCM. I tested
with mplayer's "-afm hwac3" and/or "-af lavcac3enc" options, and alsa a
WAV file that I believe has AC3 content rather than PCM.
I also tested:
* Play a stream
* Mute while playing
* Stop stream
* Play some other streams to re-assign the converter to a different
pin, PCM, set of SPDIF controls, ... hence hopefully triggering
cleanup for the original PCM.
* Unmute original stream while not playing
* Play a stream on the original pin/PCM.
This was to test SPDIF control virtualization.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A future change won't store an entire hda_pcm_stream just to represent
the capabilities of a codec; a custom data-structure will be used. To
ease that transition, modify hdmi_eld_update_pcm_info to expect the
hda_pcm_stream to be pre-initialized with the codec's capabilities, and
to update those capabilities in-place based on the ELD.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A future change will significantly rework the generic implementation
in order to support codecs with a different number of pins and
converters. Isolate the more custom codec variants from this change by
duplicating the small portions of generic code they share. This
simplifies the later rework of that previously shared code, since we
don't have to consider the more custom codecs, and also prevents
support for those codecs from regressing.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SPDIF output controls apply to converter widgets. A future change
will create a PCM device per pin widget, and hence a set of SPDIF output
controls per pin widget, for certain HDMI codecs. To support this, we
need the ability to virtualize the SPDIF output controls. Specifically:
* Controls can be "unassigned" from real hardware when a converter is
not used for the PCM the control was created for.
* Control puts only write to hardware when they are assigned.
* Controls can be "assigned" to real hardware when a converter is picked
to support output for a particular PCM.
* When a converter is assigned, the hardware is updated to the cached
configuration.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, the data that backs the kcontrols created by
snd_hda_create_spdif_out_ctls is stored directly in struct hda_codec. When
multiple sets of these controls are stored, they will all manipulate the
same data, causing confusion. Instead, store an array of this data, one
copy per converter, to isolate the controls.
This patch would cause a behavioural change in the case where
snd_hda_create_spdif_out_ctls was called multiple times for a single codec.
As best I can tell, this is never the case for any codec.
This will be relevant at least for some HDMI audio codecs, such as the
NVIDIA GeForce 520 and Intel Ibex Peak. A future change will modify the
driver's handling of those codecs to create multiple PCMs per codec. Note
that this issue isn't affected by whether one creates a PCM-per-converter
or PCM-per-pin; there are multiple of both within a single codec in both
of those codecs.
Note that those codecs don't currently create multiple PCMs for the codec
due to the default HW mux state of all pins being to point at the same
converter, hence there is only a single converter routed to any pin, and
hence only a single PCM.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's perfectly valid for an ELD to contain no SADs. This simply means that
only basic audio is supoprted.
In this case, we still want to limit a PCM's capabilities based on the ELD.
History:
* Originally, ELD application was limited solely by sad_count>0, which
was used to check that an ELD had been read.
* Later, eld_valid was added to the conditions to satisfy.
This change removes the original sad_count>0 check, which when squashed
with the above two changes ends up replacing if (sad_count) with
if (eld_valid).
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: usb - turn off de-emphasis in s/pdif for cm6206
ALSA: asihpi: Use angle brackets for system includes
ALSA: fm801: add error handling if auto-detect fails
ALSA: hda - Check pin support EAPD in ad198x_power_eapd_write
ALSA: hda - Fix HP and Front pins of ad1988/ad1989 in ad198x_power_eapd()
ALSA: 6fire: Don't leak firmware in error path
ASoC: Fix wm_hubs input PGA ZC bits
ASoC: Fix dapm_is_shared_kcontrol so everything isn't shared
This reverts commit ed0bd2333c.
Since we reverted the TTY API change, we should revert the ASoC update
to it too.
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
CM6206: Turn off de-emphasis channel status bit in S/PDIF output.
Signed-off-by: Eric Lammerts <eric@lammerts.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update Makefile and Kconfig to build HDMI audio support for
OMAP4 SDP and Panda boards.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Add machine driver for HDMI audio on OMAP4 boards. This driver is
in charge of putting together the HDMI audio codec and the CPU DAI
and register the HDMI sound card with ALSA.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Addition of the HDMI CPU DAI driver for OMAP4. This driver is in
charge of configuring DMA settings for HDMI. Also, it finds
the HDMI video device and determines if audio playback can proceed.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
We should only call ssc_free() when ssc_request() succeeds or bad
things will happen.
Signed-off-by: Joachim Eastwood <joachim.eastwood@jotron.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the original code if auto detect failed and tea575x_tuner == 4
then we copy bogus information to chip->tea.card. I've changed the
autodetect code to cleanup and return -ENODEV on error instead.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In ad198x_power_eapd(), wrong pin NIDs are used for controlling EAPD for
HP and Front outputs of AD1988/AD1989. These are actually same with the
ones for AD1984 & co, port-A is 0x11 and port-D 0x12.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We only need to increase the detection rate to maximum if we're monitoring
for button presses as the response times needed for user interaction there
are much lower.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
One of the error paths in
sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload() neglects to free
the memory allocated for the firmware before returning, thus leaking the
memory.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Peter Ujfalusi