When pin default configs are overridden via patch option, these are
evaluated before fixups are applied. Since some fixups change the
whole codec trees and/or add pins dynamically, this sanity check might
not pass when pins aren't present at the time the function is called.
We may reorder the execution, but an easier fix is simply to disable
this sanity check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix to return a negative error code from the error handling
case instead of 0, as returned elsewhere in this function.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The flag bus->shutdown implies that the control elements might have
been already destroyed. When a codec is resumed at this state and
tries to call vmaster hook (e.g. in snd_hda_gen_init()), it would
refer to a non-existing object, resulting in Oops in the end.
This patch just adds a check of the flag in the caller side for
avoiding such a crash.
Though, the best would be to clear hook->sw_kctl by the destructor of
the corresponding ctl element, but vmaster uses its own private_free,
it can't be done easily. So let it be for a while.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When graphics initializes the HDMI chip, sometimes this leads to
pins going into D3 and right channel being muted. If the audio driver
finishes initialization before the graphic driver does, this situation
becomes permanent.
This is a workaround that checks for this situation and corrects it on
playback prepare. It has been verified working on at least one machine.
BugLink: https://bugs.launchpad.net/bugs/1167270
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When setting up the aamix output paths, use the primary DAC instead of
the individual DAC for each output as default. Otherwise multiple
DACs will be turned on for a single aamix widget, which results in
doubly or more volumes, because the duplicated signals will be sent
through all these DACs for a single stream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we have a loopback mixer control, this should manage the state
whether the output paths include the aamix or not. But the current
code blindly initializes the output paths with aamix = true, thus the
aamix is enabled unless the loopback mixer control is changed.
Also, update_aamix_paths() called by the loopback mixer control put
callback invokes snd_hda_activate_path() with aamix = true even for
disabling the mixing. This leaves the aamix path even though the
loopback control is turned off.
This patch fixes these issues:
- Introduced aamix_default() helper to indicate whether with_aamix is
true or false as default
- Fix the argument in update_aamix_paths() for disabling loopback
Reported-by: Lydia Wang <LydiaWang@viatech.com.cn>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For capture, the delay through the codec contributes to the time stamp
of the sample recorded at the A to D. Rename the codec time stamp
function appropriately.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
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Merge tag 'asoc-v3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.10
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
With this patch, a TRRS headset mic cannot be successfully detected
on the Asus X101CH, and we can also distinguish between headphone
and headset automatically.
Buglink: https://bugs.launchpad.net/bugs/1169138
Co-authored-by: Kailang <kailang@realtek.com>
Tested-by: Luis Henriques <luis.henriques@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some machines, there is a headset jack that can support both
headphone, headsets (of both CTIA and OMTP type) and mic-in.
On other machines, the headset jack supports headphone, headsets
(both CTIA and OMTP), but not mic-in.
This patch implements that functionality as different capture sources.
Buglink: https://bugs.launchpad.net/bugs/1169143
Tested-by: David Chen <david.chen@canonical.com>
Co-authored-by: Kailang <kailang@realtek.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we have a flag for headphone mics, we can use that flag
in the jack creation instead of creating the jack manually.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I never liked that we move our speaker and hp pins to line out
if there are not any line outs; but now that we do,
add some convenience functions to find hp and speaker pins even
if they have been moved.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows a specific mic to get the "Headphone Mic" name, in addition
to the existing "Headset Mic" name.
Also, it allows for a special mark: if the sequence number is set
to 0xc, that's an indication to prefer it for headset mic, and if it's
set to 0xd, that's an indication to prefer it for headphone mic.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For playback add the codec-side delay to the timestamp, for capture
subtract it. This brings the timestamps in line with the time that
was recently added to the delay reporting.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct pin configs for the Acer AC700. Most importantly indicate
that SPDIF is connected, it routes to HDMI out.
Similar to Aspire models, chain in the DMIC fixup and allow it to be
applied to this codec (ALC269VB) as well.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DSP in the CA0132 codec adds a variable latency to audio depending
on what processing is being done. Add a new patch op to return that
latency for capture and playback streams. The latency is determined
by which blocks are enabled and knowing how much latency is added by
each block.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new codec PCM ops, get_delay(), to obtain the codec/stream-
specific PCM delay count. When it's NULL, nothing changes.
This new feature was requested for CA0132, which has significant
delays in the path depending on the running DSP code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
changed is not initialized in path_power_down_sync, but it is expected
to be false in case no change happened in the loop. So set it to
false.
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 6ab317419c.
The commit [6ab317419c: ALSA: hda - Allow power_save_controller option
override DCAPS] changed the behavior of power_save_controller so that
it can override the driver capability. This assumed that this option
is rarely changed dynamically unlike power_save option. Too naive.
It turned out that the user-space power-management tool tries to set
power_save_controller option to 1 together with power_save option
without knowing what's actually doing. This enabled forcibly the
runtime PM of the controller, which is known to be broken om many
chips thus disabled as default.
So, the only sane fix is to revert this commit again. It was intended
to ease debugging/testing for runtime PM enablement, but obviously we
need another way for it.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=56171
Reported-and-tested-by: Nikita Tsukanov <keks9n@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename "Digitial In" to "Digital In". This function is only used for
proc output, so should not cause any problems to change.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Added the device ID to the modalias list and assinged ALC662 patches
for it
* Added 4 port support for the device ID 0671 in alc662_parse_auto_config
Signed-off-by: Rainer Koenig <Rainer.Koenig@ts.fujitsu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch let ELD debug message show 'pin_eld->monitor_present' which reflects
the real pin response to verb GET_PIN_SENSE.
'eld->monitor_present' should not be used here because 'eld' is a temp
structure now and so its "monitor_present" is not set.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0.
Otherwise it will be returned uninitialized as non-zero after ELD info is got
successfully. Thus hdmi_present_sense() will always assume ELD info is invalid
by mistake, and /proc file system cannot show the proper ELD info.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: stable@vger.kernel.org
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Turing on the headphone amp interferes with the impedance measurement
used to detect a TRRS style headset microphone. Delay the HP turn on
until 500ms after the jack is detected, allowing the mic detection
state machine to run to completion.
Signed-off-by: Chih-Chung Chang <chihchung@chromium.org>
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the independent HP reduced the availability of the
side surround output, because there are only 4 DACs for 7.1 and a HP
outputs. Adjust the badness tables for VIA so that 7.1 outputs are
activated for the cost of missing independent HP.
Once when we implement the dynamic DAC switching to multiple outputs,
this conflicts will be eased in future...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The lack of independent HP mode shouldn't be too bad, but currently
its badness is set a bit too high. Let's lower it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The standard badness values don't seem to fit to all preferences.
Some configuration prefer the side output over the headphone, some
want the speaker over the surround, etc.
This patch moves the badness table pointers into hda_gen_spec, so that
the codec driver can override them.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge back for-linus branch for the badness table adjustment for VIA codecs
* for-linus:
ALSA: hda - Fix DAC assignment for independent HP
ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader
ALSA: hda - Fix typo in checking IEC958 emphasis bit
ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
ALSA: snd-usb: mixer: propagate errors up the call chain
ALSA: usb: Parse UAC2 extension unit like for UAC1
ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
The generic parser should evaluate the availability of the independent
HP when specified. Otherwise a DAC without the direct connection to
the corresponding pin may be assigned for the HP, but the driver
doesn't check it at all. The problem was actually seen on some
machines with VT1708s or equivalent codec, where DAC0 is assigned to
HP although it can be connected only via aamix.
This patch adds the badness evaluation for the independent HP to make
it working properly.
Reported-by: Lydia Wang <LydiaWang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we have a "Headset Mic" name, let's use it for some devices
we know for sure has a headset mic jack.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Headset mic jacks, i e TRRS style jacks with Headphone Left,
Headphone Right, Mic and GND signals, are becoming increasingly
common and are now being shipped by several manufacturers.
Unfortunately, the HDA specification does not give us any hint
of whether a Mic pin belongs to such a jack or not, but it would
still be helpful for the user to know (especially if there is one
TRS Mic jack and one TRRS headset jack).
This new fixup causes the first (non-dock, non-internal) mic to
be a headset mic jack. The algorithm can be later refined if needed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current DSP loader code abuses snd_hda_lock_devices() for ensuring
the DSP loader not conflicting with the other normal operations. But
this trick obviously doesn't work for the PM resume since the streams
are kept opened there where snd_hda_lock_devices() returns -EBUSY.
That means we need another lock mechanism instead of abuse.
This patch provides the new lock state to azx_dev. Theoretically it's
possible that the DSP loader conflicts with the stream that has been
already assigned for another PCM. If it's running, the DSP loader
should simply fail. If not -- it's the case for PM resume --, we
should assign this stream temporarily to the DSP loader, and take it
back to the PCM after finishing DSP loading. If the PCM is operated
during the DSP loading, it should get an error, too.
Reported-and-tested-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a typo in convert_to_spdif_status() about checking the
emphasis IEC958 status bit. It should check the given value instead
of the resultant value.
Reported-by: Martin Weishart <martin.weishart@telosalliance.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I forgot to update spec->gpio_data in the automute hook, so it will be
overridden at the init sequence, thus the machine is still silent when
no headphone jack is plugged at boot time.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new HP desktop machines have Realtek codecs and their LEDs are
controlled via GPIO as for many laptop models. Add similar hooks as
well as in patch_sigmatel.c for controlling LEDs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The verb to set up the digital beep via AC_VERB_SET_DIGI_CONVERT_2
should be executed at resume as well. Use the cached write for it
being performed automatically at resume.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While playing the digital beep tone, the codec shouldn't be turned
off. This patch adds proper snd_hda_power_up()/down() calls at each
time when the beep is played or off.
Also, this fixes automatically an unnecessary codec power-up at
detaching the beep device when the beep isn't being played.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling snd_hda_attach_beep_device() and
snd_hda_detach_beep_device() in each codec driver, move them to the
generic parser. The codec driver just needs to set spec->beep_nid for
activating the digital beep.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The argument passed to snd_hda_attach_beep_device() is a widget NID
while spec->beep_amp holds the composed value for amp controls.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the transition to the generic parser, the hook to the codec
specific automute function was forgotten. This resulted in the silent
output on some MacBooks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
spec->dsp_state is initialized to DSP_DOWNLOAD_INIT, no need to reset
and check it in ca0132_download_dsp().
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of using the dspload_is_loaded() function, check the dsp_state
that is kept in the spec. The dspload_is_loaded() function returns
true if the DSP transfer was never started. This false-positive leads
to multiple second delays when ca0132_setup_efaults() times out on
each write.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If dspload_image() fails, it was ignored and dspload_wait_loaded() was
still called. dsp_loaded should never be set to true in this case,
skip it. The check in dspload_wait_loaded() return true if the DSP is
loaded or if it never started.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If there are no internal speakers, we should not turn the eapd switch
off, because it might be necessary to keep high for Headphone.
BugLink: https://bugs.launchpad.net/bugs/1155016
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the driver doesn't power down the widget at going down to D3
when the widget node has an EAPD capability and EAPD is actually set
on all codecs unless codec->power_filter is set explicitly.
This caused a problem on some Conexant codecs, leading to click
noises, and we set it as NULL there. But it is very unlikely that the
problem hits only these codecs.
Looking back at the development history, this workaround for EAPD was
introduced just for some laptops with STAC9200 codec, then we applied
it blindly for all. Now, since it's revealed to have an ill effect,
we should disable this workaround per default and apply only for the
known requiring systems.
The EAPD workaround is implemented now as snd_hda_codec_eapd_power_filter(),
and this has to be set explicitly by the codec driver when needed.
As of now, only patch_stac9200() sets this one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the dynamic array allocations for pins, converters and PCM arrays
instead of the fixed size arrays. The modern HDMI codecs get more and
more pins, and we don't know the sensitive limit.
Most of the patch are spent for the straight conversions from the
fixed array access to snd_array helpers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the connection list expansion in hda_codec.c and hda_proc.c, the
value returned from snd_hda_get_num_raw_conns() is used as the array
size to store the connection list. However, the function returns
simply a raw value of the AC_PAR_CONNLIST_LEN parameter, and the
widget list with ranges isn't considered there. Thus it may return a
smaller size than the actual list, which results in -ENOSPC in
snd_hda_get_raw_conections().
This patch fixes the bug by parsing the connection list correctly also
for snd_hda_get_num_raw_conns().
Reported-and-tested-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dereference should be moved below the NULL test.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
script/kernel-doc reports the following type of warnings (when run in verbose
mode):
Warning(sound/core/init.c:152): No description found for return value of
'snd_card_create'
To fix that:
- add missing descriptions of function return values
- use "Return:" sections to describe those return values
Along the way:
- complete some descriptions
- fix some typos
Signed-off-by: Yacine Belkadi <yacine.belkadi.1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Expose the newly added TCO LTC and sync check functions to userspace.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds new ALSA controls to query the LTC state from userspace.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch prepares snd_hdspm_get_sync_check() to also check the TCO
sync state. The added feature will be exposed to the user in a later
commit.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Considerably shorten the code by using a macro. Though this won't lower
the binary size, it makes the source more readable.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For 96kHz, MADI allows to multiplex the samples (SMUX) or to use a
dedicated 96K mode. The RME cards default to 96K mode, but since not all
external MADI equipment supports this, provide a switch to users that
changes the on-wire protocol to SMUX.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When using the additional Time Code Option module in slave mode or the
SYNC-In wordclock connector, the sample rate needs to be returned by
hdspm_external_sample_rate().
Since this sample rate may contain any value with 1Hz granularity, we
need to round it to a common rate as done by the OSX driver.
[Fixed missing function declarations by tiwai]
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit introduces hdspm_get_pll_freq() to avoid code duplication.
Reading the sample rate from the DDS register will be required by
upcoming code.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a headphone pin is set up as a shared hp/mic pin, we rather want
to keep it as a headphone primarily as default, but the driver
overrides it always as a mic pin, just because the input controls are
created after outputs. Add a check of pin NID and skip the
re-initialization of pinctl for such a shared hp/mic pin.
Reported-by: Jonathan Woithe <jwoithe@just42.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the headphone mic jack enum control is created (via explicitly
specification by user), it doesn't make much sense to change the I/O
direction dynamically per capture source change, since the I/O
direction is rather controlled over the enum ctl.
This also reduces the implicit dependency between the capture source
and the hp mic jack enum ctls, which might confuse a program accessing
the whole control elements at once like alsactl.
In addition, this patch introduces update_hp_automute_hook() function
to call the proper hook function. It's just to remove the open codes
in multiple places in hda_generic.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no big merit to distinguish these two hints. Instead, just
have a single flag, add_jack_modes, for creating the jack mode enum
ctls for both I/O directions.
The hint string parser code is left and translated as add_jack_modes
just for keeping compatibility.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commits added the capability to change the pin control of
hp/mic shared jack, but it actually didn't work as expected when the
value is changed from the output to the input, since I forgot to reset
the pin I/O bit in that case. This patch fixes the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to let user test the known workaround more easily, give a few
known fixups for ALC260 to the model strings so that it can be passed
via the module option.
Also, move the unusual setups found in FSC S7020 fixup into a special
model, fujitsu-jwse, Jonathan Woithe Special Edition.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a headphone jack is configured as a shared hp/mic jack, the jack
mode enum needs to handle both input and output directions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch improves the generic parser code to allow to set up the
headphone jack as a mic input. User can enable this feature by giving
hp_mic hint string.
The former shared hp/mic feature for the single built-in mic is still
retained. This detection can be disabled now via hp_mic_detect hint
string, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set card->private_data in snd_ice1712_create for fixing NULL
dereference in snd_ice1712_remove().
Signed-off-by: Sean Connor <sconnor004@allyinics.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the zero check `hda_frame_size_words == 0' before the modulus
`buffer_size_words % hda_frame_size_words'.
Also remove the redundant null check `buffer_addx == NULL'.
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the new control cannot be created, this function will return to avoid
snd_hda_ctl_add dereferencing a NULL control pointer.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the SPDIF control array cannot be reallocated, the function
will return to avoid dereferencing a NULL pointer.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 57e5c63007 "emu10k1: allow to
disable the SRC" force hardware use only one rate (48000 hz).
EMU 0404/1010/1616 have support two hardware sampling rates (44100 and
48000 hz). This patch add check if we have EMU 0404/1010/1616 and
choose correct sample rate to restrict.
Signed-off-by: Mihail Zenkov <mihail.zenkov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This expands the regression fix from
d28215996b.
The firmware also needs to be loaded when it was already cached.
Signed-off-by: Florian Zeitz <florob@babelmonkeys.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes initialization of the AK4114 chip so spdif capture is working properly.
Worked out together with Pavel Hofman.
Signed-off-by: Jonas Petersen <jnsptrsn1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Along with a clean up commit [e9f66d9b9: ALSA: pci: clean up using
module_pci_driver()], bt87x driver lost the functionality of load_all
parameter. This patch does a partial revert of the commit only for
bt87x.c to recover it.
Reported-by: Clemens Ladisch <cladisch@googlemail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ELD validity can change during the lifetime of a presence detect,
so we need to be able to listen for changes on the ELD control.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because the eld buffer can be simultaneously accessed from both
workqueue context (updating) and process context (kcontrol read),
we need to protect it with a mutex to guarantee consistency.
To avoid holding the mutex while reading the ELD info from the
codec, we introduce a temporary eld buffer.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For better readability, the information that is parsed out of the
ELD data is now put into a separate parsed_hdmi_eld struct.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previously, it was possible to read the eld data of the previous
monitor connected. This should not be allowed.
Also refactor the function slightly.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, eld_valid is never set to false, except at kernel module
load time. This patch makes sure that eld is no longer valid when
the cable is (hot-)unplugged.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the recent update, Fujitsu S7020 laptop with ALC260 codec lost the
speaker output, no matter how the amps and the pins are set. After a
long debugging session, we found out that the default codec init code
is harmful for this machine, and we have to reset it to
ALC_INIT_NONE.
Reported-and-tested-by: Jonathan Woithe <jwoithe@just42.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These two machines have no mute LED string in BIOS.
BugLink: https://bugs.launchpad.net/bugs/1128934
Tested-by: Tammy Yang <tammy.yang@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This chip needs the speaker pin to go to D3 to avoid clicks,
so default_power_filter does not work here.
This was found on Thinkpad R61i/T61i but I guess it applies to
the entire chip. If not, quirks should be set for at least
PCI SSID 17aa:20ac.
Thanks to c4pp4 for testing.
BugLink: https://bugs.launchpad.net/bugs/886975
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a regression report wrt the IRQ issue related with the
power-save on a Dell machine, and disabling runtime PM works around.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=53441
Cc: <stable@vger.kernel.org> [v3.7+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current badness value used for the missing multi-io seems too
weak, and the multi-io tends to be skipped for desktop configurations
when no enough DACs are available. It's because the total badness of
the multi-io becomes often larger than the badness with assigning an
individual DAC to a headphone jack. This is good for one side, but it
seems that the surround outputs are more demanded by that.
This patch increases the badness value for the missing multi-io
slightly so that the multi-io would be preferred than the individual
headphone DAC if they conflict. Through the tests with hda-emu,
mostly only desktop configurations with ALC662/663 and CMI codecs are
affected by this change, and all look reasonable.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [dcda58061: ALSA: hda - Add workaround for conflicting
IEC958 controls] introduced a workaround for cards that have both
SPDIF and HDMI devices for giving device=1 to SPDIF control elements.
It turned out, however, that this workaround doesn't work well -
- The workaround checks only conflicts in a single codec, but SPDIF
and HDMI are provided by multiple codecs in many cases, and
- ALSA mixer abstraction doesn't care about the device number in ctl
elements, thus you'll get errors from amixer such as
% amixer scontrols -c 0
ALSA lib simple_none.c:1551:(simple_add1) helem (MIXER,'IEC958
Playback Switch',0,1,0) appears twice or more
amixer: Mixer hw:0 load error: Invalid argument
This patch fixes the previous broken workaround. Instead of changing
the device number of SPDIF ctl elements, shift the element indices of
such controls up to 16. Also, the conflict check is performed over
all codecs found on the bus.
HDMI devices will be put to dev=0,index=0 as before. Only the
conflicting SPDIF device is moved to a different place. The new place
of SPDIF device is supposed by the updated alsa-lib HDA-Intel.conf,
respectively.
Reported-by: Stephan Raue <stephan@openelec.tv>
Reported-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org> [v3.8]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The chip address is 32bit long but INVALID_CHIP_ADDRESS is defined as
an unsigned long. This makes dsp_chip_to_dsp_addx() misbehaving on
64bit architectures. Fix the INVALID_CHIP_ADDRESS definition to be
32bit.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The problem addressed by this fixup is not specific to Vaio Z, affecting
some Vaio all-in-one desktop PCs too. Update the code comments accordingly.
Signed-off-by: Fernando Luis Vazquez Cao <fernando@oss.ntt.co.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Vaio all-in-one desktop PCs (for example VGC-LN51JGB) are affected by
the same issue that caused Vaio Z laptops to become silent: the speaker pin
must be connected to the first DAC even though the codec itself advertises
flexible routing through any of the DACs.
Use the no-primary-hp fixup for choosing the speaker pin as the primary so
that the right DAC is assigned on this device.
Cc: stable@vger.kernel.org
Signed-off-by: Fernando Luis Vazquez Cao <fernando@oss.ntt.co.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_ali_pointer function is called with local
interrupts disabled. However it seems very strange to
reenable them in such way.
Found by Linux Driver Verification project (linuxtesting.org).
Signed-off-by: Denis Efremov <yefremov.denis@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the other code in this driver and similar
code in rme96 it seems, that spin_lock_irq in
snd_rme32_capture_close function should be paired
with spin_unlock_irq.
Found by Linux Driver Verification project (linuxtesting.org).
Signed-off-by: Denis Efremov <yefremov.denis@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes a few obvious bugs in DSP loader stuff:
- Fix possible memory leaks in the error path
- Avoid double-free calls in dma_reset()
- Properly set/unset WC bits for DMA buffers
- Add missing error status checks
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Base the DSP firmware transfer and communication timeouts on jiffy values.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the HD Audio Device IDs for the Intel Wellsburg PCH
Signed-off-by: James Ralston <james.d.ralston@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A Haswell test machine showed that the invalid connection list, but
this time it has only a single pin on the codec, thus the former fixup
code doesn't work as it assumes the three pins blindly.
This patch splits the former fixup code to two parts:
- Enable eDP 1.2 for Haswell codec
- Fix the connection list of pins on Haswell codec;
the converter list is recorded dynamically in hdmi_add_cvt(), and
applied in hdmi_add_pin()
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Haswell machines support more than one display outputs (HDMI or DP),
but its BIOS may not enable the codec's 2nd and 3rd pin and output cvt widgets.
This patch implements a board-specific fixup for Intel Haswell Machines:
If the hidden pins are not enabled by BIOS, the driver will enable them
and call common code to update the codec tree.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A codec may allow software to hide some unused pin/cvt widgets.
Sometimes BIOS does not enable the hidden widgets properly although they are
needed for the board. Thus the driver need to enable them as a board-specific
fixup and the whole tree will change.
This patch implements a common code for rereading codec widgets. So the fixup
code can call it after enabling the hidden widgets.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we set the max number of connections to be 32, but there
seems codec that gives longer connection lists like AD1988, and we see
errors in proc output and else. (Though, in the case of AD1988, it's
a list of all codecs connected to a single vendor widget, so this must
be something fishy, but it's still valid from the h/w design POV.)
This patch tries to remove this restriction. For efficiency, we still
use the fixed size array in the parser, but takes a dynamic array when
the size is reported to be greater than that.
Now the fixed array size is found only in patch_hdmi.c, but it should
be fine, as the codec itself can't support so many pins.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix these two compile errors on s390 which does not have HAS_IOPORT
nor GENERIC_HARDIRQS:
sound/pci/lx6464es/lx6464es.c: In function ‘snd_lx6464es_free’:
sound/pci/lx6464es/lx6464es.c:565:2: error: implicit declaration of function ‘ioport_unmap’
sound/soc/codecs/wm8903.c: In function ‘wm8903_set_pdata_irq_trigger’:
sound/soc/codecs/wm8903.c:1954:9: error: implicit declaration of function ‘irq_get_irq_data’
Signed-off-by: Heiko Carstens <heiko.carstens@de.ibm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c: In function ‘ca0132_is_vnode_effective’:
sound/pci/hda/patch_ca0132.c:3331:15: warning: ‘nid’ may be used uninitialized in this function [-Wmaybe-uninitialized]
sound/pci/hda/patch_ca0132.c:4345:13: warning: ‘ca0132_download_dsp’ defined but not used [-Wunused-function]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loopback list is referred by the VIA codec driver no matter
whether CONFIG_PM is set or not, thus we need to enable it always.
Otherwise it gets compile errors.
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The wm->regs[] array has WM8766_REG_COUNT (16) elements not
WM8766_REG_RESET (31).
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's mostly harmless to apply it for new models even if they have no
mic mute LED (just toggling an unused GPIO pin).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These days, GUIs such as Gnome sound settings want to be able to
show the correct jack status even when no streams are currently
running. I doubt this gives any measurable difference in power,
but if it does, the "Jack Detect" control can still be used to
turn polling off.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The VT1708 has no unsol event capability, and polling is set using
the "Jack Detect" alsamixer control. In order not to create
phantom Jack controls, temporary enable jackpoll during build_controls.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... to be less confusing for the update path.
This new kconfig will choose CONFIG_SND_HDA_DSP_LOADER, which is
basically a device-independent feature in hda_intel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit d45e6889ee ("ALSA: hda - Provide
the proper channel mapping for generic HDMI driver") added support for
custom channel maps in the HDA HDMI driver. Due to a mistake in an
'if' condition the custom map is always used even when no such map has
been set. This causes incorrect channel mapping for multichannel audio
by default.
Pass per_pin->chmap_set to hdmi_setup_channel_mapping() as a parameter
so that it can use it for detecting if a custom map has been set instead
of checking if map is NULL (which is never the case).
Reported-by: Staffan Lindberg <pike@xbmc.org>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the driver detects and invalid ELD, it gives an open error.
But it forgot to release the assigned pin, converter and spdif ctls
before returning.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add new PCI ID 0x0a0c for Haswell ULT platform.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For non-snoop mode, we fiddle with the page attributes of CORB/RIRB
and the position buffer, but also the ring buffers. The problem is
that the current code blindly assumes that the buffer is contiguous.
However, the ring buffers may be SG-buffers, thus a wrong vmapped
address is passed there, leading to Oops.
This patch fixes the handling for SG-buffers.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=800701
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we use LPIB forcibly for both playback and capture for
Poulsbo and Oaktrail devices, and this seems rather problematic.
The recent fix for LPIB delay count seems working well with these
devices, so let's enable it instead.
Reported-by: Martin Weishart <martin.weishart@telosalliance.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes a regression of the external mic not working on
HP Probook 4520s.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because currently snd_printd() and snd_printdd() macros are expanded
to empty when CONFIG_SND_DEBUG=n, a compile warning like below
appears sometimes, and we had to covert it by ugly ifdefs:
sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’:
sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable]
For "fixing" these issues better, this patch replaces snd_printd() and
snd_printdd() definitions with empty inline functions instead of
macros. This should have the same effect but shut up warnings like
above.
But since we had already put ifdefs, changing to inline functions
would trigger compile errors. So, such ifdefs is removed in this
patch.
In addition, snd_pci_quirk name field is defined only when
CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in
snd_printdd() argument triggers the build errors, too. For avoiding
these errors, introduce a new macro snd_pci_quirk_name() that is
defined no matter how the debug option is set.
Reported-by: Stratos Karafotis <stratosk@semaphore.gr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a better power filter hook for powering down unused
widgets in the generic parser.
The feature is enabled by setting hda_gen_spec.power_down_unused
flag.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put the power state synchronization at the end of the parsing of
codec. This is necessary when the power filter is changed during the
codec probe. Since the first power-up sequence is performed without
the special filter, all widgets are supposed to be ON at this point.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a hook to struct hda_codec for filtering the target power state of
each widget when powering up/down. The current hackish EAPD check is
implemented as the default hook pointer, too.
This allows codec drivers to implement own power filter. In the
upcoming changes, the generic parser will have the better power filter
based on the active paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AC_VERB_GET_POWER_STATE returns the combined bits of the actual state
and the target state. Thus, comparing the obtained value directly
with the target value can't work. The value has to be shifted and
masked properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The arguments to call is_active_nid() in activate_amp() were swapped,
and this resulted in the muted amp on some SPDIF output pins.
Also, the index to be passed to is_active_nid() must be idx_to_check.
Otherwise it checks the wrong connection in the case of implicit aamix
connection paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using the new chained_before flag, we can correct the headphone jack
detection capability easily over the existing ALC880 6stack model
(which disables the jack detection intentionally for compatibility
reason).
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=901846
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A Packard-Bell desktop machine gives no proper pin configuration from
BIOS. It's almost equivalent with the 6stack+fp standard config, just
take the existing fixup.
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=901846
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes we want to call a fixup after applying other existing
fixups, but currently the fixup chain mechanism allows only the call
the others after the target fixup. This patch adds a new flag,
chained_before, to struct hda_fixup, for allowing the chained call
before the current execution.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD1988 family and AD1882 codecs have another mixer widget (0x21)
between the analog-loopback mixer widget (0x20) and the actual
outputs. Due to this hole, the analog-loopbacks aren't sent properly
to the output pins.
As a band-aid fix, introduce another fields holding the aamix merge
path, and activate it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [26a6cb6c: ALSA: hda - Implement a poll loop for jacks as a
module parameter] introduced the polling jack detection code, but it
also moved the call of snd_hda_jack_set_dirty_all() in the resume path
after resume/init ops call. This caused a regression when the jack
state has been changed during power-down (e.g. in the power save
mode). Since the driver doesn't probe the new jack state but keeps
using the cached value due to no dirty flag, the pin state remains
also as if the jack is still plugged.
The fix is simply moving snd_hda_jack_set_dirty_all() to the original
position.
Reported-by: Manolo Díaz <diaz.manolo@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fixup function is called multiple times before parsing the pins,
so snd_BUG_ON() hits when loaded. Move it to the proper place in the
if block.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a merge of really big changes: the generic parser is heavily
enhanced for handling all cases, based on the former Realtek codec
driver code. And all codec drivers except for a few ones (CA0132,
HDMI and modem) have been converted to use the new generic driver.
Conflicts:
sound/pci/hda/patch_realtek.c
Now all AD codecs have the proper BIOS auto-parser, and we can make
it for default, finally. (AD1988 already did it because it had the
auto-parser.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As done for patch_conexant.c, put ifdef ENABLE_AD_STATIC_QUIRKS for
preparing t odrop the static quirk codes in patch_analog.c.
The whole static quirk code can be omitted by commenting out
ENABLE_AD_STATIC_QUIRKS define now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD codecs have strange implementations for choosing the SPDIF-output
mux source: the digital audio out widget may take the sources from
multiple connections, where 0x01 indicates it's a PCM while others
point ADCs. It's obviously invalid in the HD-audio spec POV, but it's
somehow convincing, too. And, to make things more complex, AD1988A
and AD1882 have deeper connection routes that aren't expressed
correctly.
In this patch, the SPDIF mux control is implemented in two ways:
- For easier one like AD1981, AD1983, AD1884 and AD1984, where the
SPDIF audio out widget takes just two or three sources, we can
simply implement via the normal input_mux and connection verb
calls.
- For the complex routes like AD1988A (but not AD1988B) or AD1882, we
prepare "faked" paths represented statically, and switch the paths
using these static ones, instead of parsing the routes from the
widget tree.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since both snd_hda_codec_flush_amp_cache() and
snd_hda_codec_flush_cmd_cache() are called usually at the same time,
we can simply combine them to a single function,
snd_hda_codec_flush_cache().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture volume put callback may call the node selection change,
and its actual call won't be triggered unless flushed. In general,
we always need to call both snd_hda_codec_flush_amp_cache() and
snd_hda_codec_flush_cmd_cache() at the same place...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both the HP auto-mute and the independent HP mode conflict with each
other. Make HP auto-mute disabled (only for the affected HP jack)
during the driver is in HP independent mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It'd be better to give another name to the secondary (alt) analog PCM
stream, which is dedicated for the independent HP out and extra
inputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The aamix NIDs are also missing for AD codecs. All AD codecs seem to
have a (more or less) working aamix widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
IDT codecs have analog-loopback mixer widgets, but we haven't cared
about it, so far. Let's set them. This will avoid also possible
wrong routes for the input paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch eventually fixes two issues:
- Handle the case where the primary output is a headphone and can have
independent HP mode;
so far we checked only the case where the headphone is the secondary
output.
- Fix the conflict of HP independent mode and aamix mode;
when switched to aamix mode, the DAC might be also switched to
another widget shared with other outputs. Then even if we disable
the DAC for the original output, it doesn't change -- because the
active route is from another (shared) DAC to HP pin through aamix.
So, in such a case, we have to prohibit the switch to aamix for HP
routes.
This fixes issues appearing on VT codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many codecs provide routes to multiple output pins through an aamix
widget, but most of them do it only from a single DAC. However, the
current generic parser checks only the aamix paths from the original
(directly bound) DACs through aamix NID, and miss the path:
primary DAC -> aamix -> target out pin
This patch adds a more check for the routes like the above.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a patch couldn't be resolved in try_assign_dacs() although the
target DAC is expected, we forgot to add a proper badness value but
continued.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since fill_and_eval_dacs() may be called repeatedly with different
configurations, setting pinctls at each time there isn't optimal.
We can set it better only once after deciding the output configuration
in parse_output_paths().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Print the information of outputs in a bit more details and concisely
in a single place instead of printing the path at each time when
detected.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conexant CX20551 codec has a mixer in NID 0x19 and a few outputs have
to take the input through this widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Looking through the whole definitions, some fields have inappropriate
array sizes, especially about the capture. The array assigned to each
input (pin) should have HDA_MAX_NUM_INPUTS entries while the array
assigned to each ADC should have AUTO_CFG_MAX_INS entries.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch "ALSA: hda - fix wrong adc_idx in generic parser" fixed the
adc_idx for the capture volume and capture switch controls. But also
modified the adc_idx retrieval for the capture source controls
wrongly. As multiple capture source controls are created in a single
shot with counts > 1, the id.index doesn't contain the real value.
The real index has to be taken via snd_ctl_get_ioffidx() as in the
original code.
This patch reverts the fixes partially to recover from the
regression.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If there's one each of HDMI and SPDIF, we should not add an index
on the one that comes second.
[slight code refactoring by tiwai]
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just stumbled over this one while reading the code.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I found a codec configuration which had six inputs, so the max of
five was not appropriate.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some BIOS version of FSC Lifebook S7110 laptop seems to give a wrong
default pin config for NID 0x15, which confuses the parser. Give a
fixup to correct the value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although I commented that boost volumes would be added only for
line-in and mic pins in the source code, the actual code excludes but
for mic-in. Fix it to accept the line-ins, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Two hooks in hda_gen_spec, cap_sync_hook and capture_switch_hook, play
very similar roles. The only differences are that the former is
called more often (e.g. at init or switching capsrc) while the latter
can take an on/off argument.
As a more generic implementation, consolidate these two hooks, and
pass snd_ctl_elem_value pointer as the second argument. If the
secondary argument is non-NULL, it can take the on/off value, so the
caller handles it like the former capture_switch_hook. If it's NULL,
it's called in the init or capsrc switch case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a standard capture switch without multiple binding is used, the
call for capture_switch_hook isn't called properly. Replace the put
ops to add the hook call in that case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This machine also has the "HP_Mute_LED_0_A" string in DMI information.
Cc: <stable@vger.kernel.org>
BugLink: https://bugs.launchpad.net/bugs/1096789
Tested-by: Tammy Yang <tammy.yang@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current generic parser code, we look for the (mic) boost
controls only on input pins. But many codecs assign the boost volume
to a widget connected to each input pin instead of the input amp of
the pin itself.
In this patch, the parser tries to look through more widgets connected
to the pin and find a boost amp.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an amp in the activation path is associated with mixer controls,
activate_amp() tries to skip the initialization. It's good, but only
if the mixer really initializes both mute and volume. Otherwise,
either the mute of the volume is left uninitialized.
This patch adds this missing check and properly initialize the
partially controlled amps in an activation path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few places creating the labels and indices of kctls for
each input pin in the current generic parser code. This is redundant
and makes harder to maintain. Let's create the labels and indices at
once and keep them in hda_gen_spec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the imux table entries can be a subset of autocfg.input table,
the indices of these aren't always same. For passing the proper index
value of autocfg.input at creating input ctl labels (via
snd_hda_autocfg_input_label()), keep the corresponding autocfg.input
idx value in the index field of each imux item, which isn't used in
the generic driver.
Also, this makes easier to check the invalid imux pin for stereo mix.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Finally we reached here. All codecs driver (except for CA0132, which
has really device-specific requirements) have been converted to use
the generic parser.
This patch appears bigger than others since it also involves with the
code shuffling, but mostly the cut-off of parser codes and adapt to
the generic parser flags. Most of fixup codecs haven't been changed
but just removed a few unnecessary codes.
The only missing stuff is the SPDIF mux control. It'll be added again
later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* test/hda-gen-parser:
ALSA: hda - Improve naming rule for primary output
ALSA: hda - Add PCM capture hook to hda_gen_spec
ALSA: hda - Record all detected ADCs in hda_gen_spec
ALSA: hda - Move vmaster TLV parsing to snd_hda_gen_parse_auto_config()
ALSA: hda - Add input jack mode enum controls to generic parser
ALSA: hda - Give more comments to hda_gen_spec flags
ALSA: hda - Add suppress_auto_mute flag to hda_gen_spec
ALSA: hda - Record the current speaker / LO mute status in hda_gen_spec
ALSA: hda - Properly call automute/switch hooks at init
When the volume or mute control of the primary output is shared with
other (headphone or speaker) outputs, we shouldn't name it as a
specific output type but rather name it with the channel name or a
generic name like "PCM".
Also, this check should be performed individually for the volume and
the mute controls because some codecs may have shared volumes but
separate mute controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the generic parser reduces the ADC list, copy the list of the
all detected ADCs and keep it.
This list can be later referred by the codec driver for finer power
controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add vmaster_tlv[] to hda_gen_spec and store the suggested TLV data
in snd_hda_gen_parse_auto_config(). This allows the codec driver to
correct the TLV data (e.g. mute capability) before actually creating
vmaster instance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just like the jack mode enum ctls for output jacks, add the support
for similar enum ctls for input pins to control the bias Vref.
The new controls will be added when spec->add_in_jack_modes is set
either by the codec driver or by a hint string.
Note that ground and 100% vrefs are excluded from the list for
simplicity, currently. We may add a new flag to allow them, too.
But I guess it's easier to put a value override in the pinfix in such
a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- spec->hp_detect has to be overridden in HDA_FIXUP_ACT_PARSE, not in
PRE_PARSE.
- Remove err == 0 check but return directly -EINVAL from
stac92xx_parse_auto_config()
- Set spec->default_polarity for 92HD71bxx
- Some code shuffles
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A new flag to skip the auto-mute handling in the generic parser, just
like suppress_auto_mic flag. It has to be set before calling
snd_hda_gen_parse_auto_config().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* test/hda-gen-parser:
ALSA: hda - Make sure fill_all_dac_nids is called for digital only codecs
ALSA: hda - force different capture controls if amp caps differ
ALSA: hda - do not add non-existing Mic boost controls
ALSA: hda - initialize channel counts correctly
ALSA: hda - fix wrong adc_idx in generic parser
ALSA: hda - Check array bounds in get_input_path
ALSA: hda - Add prefer_hp_amp flag to hda_gen_spec
ALSA: hda - fix OOPS in hda_mark_cmd_cache_dirty
ALSA: hda - Check pincap while parsing the configuration
Otherwise no PCM will be built for codecs without analog I/O.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise setting the capture volume for amps will be weird and
inconsistent (it will try to set values outside the range of the
second amp based on capabilities of the first amp).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the input node does not have any volume capable input amp,
don't add such a control.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even a single DAC can output two channels, so the channel count
is twice the number of DACs.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We use knew->index for adc_idx when we create "Capture Volume" and
"Capture Switch", so use the same to retrieve adc_idx.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c:387:19: sparse: symbol 'ca0132_voicefx' was not declared. Should it be static?
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With HDSP_TOGGLE_SETTING in place, these functions are no longer
required. Removing them makes the code DRY and considerably shorter.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDSP_TOGGLE_SETTING and its corresponding functions allow to change
settings in the control register. Instead of using the specialised
functions, use the generic code to make the code DRY.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver contains multiple similar functions that change only a single
bit in the control register, only the bit position varies.
This patch implements a generic function to toggle a certain bit
position that will be used to replace the old code.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current iobox detection code reportedly fails for various users, so
simply do what the Win32 driver does instead.
Patch originally by Karl Grill <kgrill@chello.at> and then modified to
comply with kernel coding guidelines + current HEAD.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new flag to indicate whether HP amp is turned on as default for
speaker or line-outs, and enable this for ALC260 codec, as many
machines with this codec require the HP amp even for speakers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c: In function ‘ca0132_effects_set’:
sound/pci/hda/patch_ca0132.c:3391:2: warning: too many arguments for
format [-Wformat-extra-args]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Spotted by smatch,
sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: potential
null dereference 'dma_engine'. (kzalloc returns null)
sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: we
previously assumed 'dma_engine' could be null (see line 1857)
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c:1781 dspxfr_one_seg() info: why not
propagate 'status' from dsp_dma_stop() instead of (-5)?
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent update of ca0132 driver replaced the pinctl setup to the
direct write via snd_hda_codec_write() again. This should be covered
by snd_hda_set_pin_ctl() to be safer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit c3b4eea262.
Since the recent firmware loader code supports caching at S3/S4 by
itself, we don't have to handle f/w caching in the driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Handle a potential dma_engine alloc error and fix the possible use of an
uninitialized status variable in dspxfr_one_seg(). Also correct the initial
sampling rate for Mic 1.
Update the module description.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the controls used for tuning the DSP effects.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the unsolicited response handler for incoming DSP responses and
jack detection reporting, and routines for reading the incoming DSP response.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the playback PCM open callback.
PCM stream setup and cleanup functions are added for use by PCM callbacks.
Delay stream cleanup if effects are on, to allow time for any effects tail to
finish.
Add the analog capture PCM callbacks.
Change the max channels of analog playback to 6.
Add two new PCMs: AMic2 and What-U-Hear.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the kcontrols for the DSP effects, playback and recording
source selection.
ca0132_is_vnode_effective() checks whether virtual node settings have
taken effect.
The control change helpers ca0132_pe_switch_set(), ca0132_voicefx_set()
and ca0132_cvoice_switch_set() are added to toggle playback / capture
DSP effects, ca0132_voicefx_info(), _get() and _put() are added for
input path DSP effect value access. The volume helpers are updated to
volume_info(), _get() and _set() to use the virtual nodes.
The redundant headphone and speaker switches and ct_extension function
are removed.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the framework to set effect parameters: ca0132_effects_set()
and ca0132_setup_defaults() are general functions for parameter setting and
initializing to default values. dspio_set_param() and dspio_set_uint_param()
are lower-level fns to simplify setting individual DSP parameters via an
SCP buffer transfer to the firmware.
The CA0132 chip parameter init code is added in ca0132_init_params().
In chipio_[write,read]_data(), the current chip address is auto-incremented
if no error has occurred.
ca0132_select_out() selects the current output. If autodetect is enabled,
use headphones (if jack detected) or speakers (if no jack).
ca0132_select_mic() selects the current mic in. If autodetect is enabled,
use exterior mic (if jack detected) or built-in mic (if no jack).
Init digital mic and switch between dmic and amic with ca0132_init_dmic(),
ca0132_set_dmic(). amic2 is initialized in ca0132_init_analog_mic2().
Finally, add verb tables for configuring DSP firmware.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds definitions and structs used for configuring DSP effects,
virtual nodes, effect tuning controls, and mixer control helpers.
The effect structs are also initialized.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When "alsactl restore" is performed on HDMI codecs, it tries to
restore the channel map value since the channel map controls are
writable. But hdmi_chmap_ctl_put() returns -EBADFD when no PCM stream
is assigned yet, and this results in an error message from alsactl.
Although the error is harmless, it's certainly ugly and can be
regarded as a regression.
As a workaround, this patch changes the return code in such a case to
be zero for making others happy. (A slight excuse is: when the chmap
is changed through the proper alsa-lib API, the PCM status is checked
there anyway, so we don't have to be too strict in the kernel side.)
Cc: <stable@vger.kernel.org> [v3.7+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of checking the codec SSID in find_mute_led_cfg() for HP Mini
110, set the proper spec->default_polairty in the fixup table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI vendor ID check in find_mute_led_cfg() is now superfluous
because the function is called in the fixup table entries of HP
machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Finally all codecs in patch_sigmatel.c have been converted to use the
standard fixup helpers. This change also includes trivial cleanups
like the call of common setup for GPIO LED or the removal of unused
function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This one is rather a simple conversion. The fixups for Dell machines
are implemented by fixup functions in the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This time, the only intrusive changes are for HP machines.
As the mute LED fixup and the bass speaker switch are required only
for HP machines, we can move these checks into the fixup entries; the
former is applied generically to all HP machines while the latter for
only certain models.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes (or rather often) BIOS sets the pin default configurations
obviously wrongly. Looking through these failures, one common pattern
is to enable some dead pins that are usually marked as speaker pins.
In such a case, we can skip them if the pins don't have the output
capability.
In this patch, add a check for the valid pin cap bit for each parsed
pin, and filter out when it's invalid.
The fix was originally suggested by Raymond Yau.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This conversion is a bit tricky. Since STAC927x may take two
different volume-knob initialization values depending on the model, a
new flag, spec->volknob_init, is introduced to indicate whether it's
the standard volume-knob initialization or not.
Also, Dell BIOS model is now directly mapped onto the fixup table
instead of parsing in the function. This resulted in a new model ref,
STAC_927X_DELL_BIOS_SPDIF, which is a chained entry.
Also, for reducing the fixups, virtual entries like
STAC_927X_DELL_DMIC and STAC_D965_VERBS are introduced.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rather straightforward conversion, except for ones for Intel Mac.
As Intel Mac have only unique codec SSIDs, we need to remap the fixup
again for the codec SSID and call the new fixup there.
Also, we can reduce model enums like STAC_MACMINI, which are model
aliases for backward compatibility, since they can be pointed directly
via hda_model_fixup table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Yet another step forward. As all features for VIA codecs have been
implemented in the generic driver, we can move on to migrate the VIA
codec parser, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the support for the generic auto-parser to AD codec
driver. For AD1988, the old code is replaced simply with the new
generic parser. For other codecs, new model "auto" is added and
directed to use the generic parser.
No fixup codes have been implemented yet as of now. Eventually we'd
replace each static quirk with the generic parser + fixup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just shuffle the codes and add ifdefs for testing to drop the static
quirk codes from patch_conexant.c.
By commenting out ENABLE_CXT_STATIC_QUIRKS define at the beginning of
the file, you can disable the whole static codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This time, the target is Cirrus codec. Its parser is a subset of
generic parser, so we can migrate fully with it now.
The only tricky part is the handling of SPDIF automute.
Cirrus driver sets the SPDIF out plug over the headphone. As a
workaround, set spec->gen.master_mute for toggling the headphone (and
other) mute.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CA0110 codec is a fairly straightforward hardware implementation,
and we can use the generic parser almost as is.
Just set spec->multi_cap_vol flag to follow the current behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the old parser code for C-Media auto-parser with the latest
generic parser. For compatibility reason, the static bindings are
still left, but they could be cleaned up in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai