For processing the firmware handling properly for built-in kernels,
implement an asynchronous firmware loading with
request_firmware_nowait(). This means that the codec probing is
deferred when the patch option is specified.
Tested-by: Thierry Reding <thierry.reding@avionic-design.de>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This merges the changes for converting to new PM ops for platform
and some other drivers.
Also move some header files to local places from the public
include/sound.
Move snd_hda_mixer_amp_switch_put_beep() to hda_beep.c as a clean up
to remove one more ifdef.
Also add the corresponding get callback to return consistently the
digital beep state independently from the mixer amp value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Straightforward conversion to the new pm_ops from the legacy
suspend/resume ops.
Since we change vx222, vx_core and vxpocket have to be converted,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After cancel_delayed_work_sync returns, the power down work either never
started (power_on == 1) or finished (power_on == 0). In the former case
there is no need to power up again.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This addresses an issue encountered when a pcm is opened while
transitioning to low power state (codec->power_on == 1 &&
codec->power_transition == -1). Add snd_pcm_power_up_d3wait to
hda_codec. This function is used to power up from azx_open as opposed
to snd_hda_power_up used from codec_exec_verb. When powering up from
azx_open, wait for pending power downs to complete, avoiding the power
up continuing in parallel with the power down on the work queue.
The specific issue seen was with the CS4210 codec, it powers off the ADC
and DAC nid in its suspend handler. If it is re-opened before the
~100ms power down process completes, the ADC and DAC nid are initialized
while powered down and audio is lost until another suspend/resume cycle.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
codec may reject power state transition requests(reporting PS-ERROR set),
in that case we re-issue a power state setting and check error bit again.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
if EPSS supported, transition from D3 state to D0 state in less
than 10ms
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add function to check whether power states supported by specific
codec node.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'hda-switcheroo' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull VGA-switcheroo audio client support for HD-audio from Takashi Iwai.
This depended on the recent drm pull.
* tag 'hda-switcheroo' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - unlock on error in azx_interrupt()
ALSA: hda - Support VGA-switcheroo
ALSA: hda - Export snd_hda_lock_devices()
ALSA: hda - Check the dead HDMI audio controller by vga-switcheroo
Like the previous fixes for cache hash accesses, a protection over
accesses to the widget connection list array must be provided.
Together with this action, remove snd_hda_get_conn_list() which can be
always race, and replace it with either snd_hda_get_num_conns() or
snd_hda_get_connections() calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's a preliminary work for the vga-switcher support.
Export the function to do pseudo-lock for the sound card to be used
in other places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that the FLOAT format on CS4206 results in simple
noises, which implies that this is no right format as is.
Since CS4206 is the only codec supporting the float, let's disable it
until we find the correct format.
Reported-and-tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amp and caps hashes aren't protected properly for concurrent
accesses. Protect them via a new mutex now.
But it can't be so simple as originally thought: since the update of a
hash table entry itself might trigger the power-up sequence which
again accesses the hash table, we can't cover the whole function
simply via mutex. Thus the update part has to be split from the mutex
and revalidated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the power_transition up/down state instead of boolean bit, so
that the power-up sequence can cancel the pending power-down work
properly. Also, by moving cancel_delayed_work_sync() before the
actual power-up sequence, make sure that the delayed power-down is
completed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the recent commit, the resume procedure is always performed at
the resume time. This makes the pre_resume hack for VREF mute LED on
some HP laptops superfluous. As this is the only user of pre_resume
(and there is no user of post_suspend) ops, let's kill them again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some power-saving states have been left unchanged in
snd_hda_codec_reset(), and this is a potential danger because the
function may be called in various situations including the continuous
operation after that call.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a fix for the problem in commit 785f857d1c, the pop noise
issue on some machines with ALC269. The problem was the uninitialized
state after the resume due to the delayed resume of the codec chips.
In that commit, we tried to fix by forcibly putting the codec to D3 at
suspend. But, this still also leaves the uninitialized state after
resume, and it _might_ be still problematic with some BIOS. Since the
commit turned out to regress another issues, we reverted it in the
end.
Now, in this fix, try to fix by turning on the codec immediately at
the resume path. We need to take care of the power-saving in this
case. When the device is woken up at the power-saved state, it should
go power-saving again after the resume.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 785f857d1c.
The commit causes a problem with the wrong D3 state after suspend
because the call of hda_set_power_state() involves with the power-up
sequence, which changes the power_count, and this confuses the resume
sequence that checks the power_count as well.
Originally, this go-to-D3 sequence should be a simple task without the
power-up sequence. But, it'd need some proper sanity checks in the
case of power-saved state, so it's not too easy to write now in the
3.4-rc cycle.
In short, the safest option now is to revert this affecting commit.
Of course, we need to clean up and robustify the power-saving code
better for 3.5 kernel.
Reported-by: Konstantin Khlebnikov <khlebnikov@openvz.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some old codecs like ALC880 seem to give a bogus pin capability value 0
occasionally. This breaks the new sanity check in snd_hda_set_pin_ctl().
Skip the sanity checks in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new helper function to guess the default VREF pin control bits
for mic in. This can be used to set the pin control value safely
matching with the actual pin capabilities.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For setting the pin-control values more safely to match with the
actual pin capability bits, a copule of new helper functions,
snd_hda_set_pin_ctl() and snd_hda_set_pin_ctl_cache(), are
introduced. These are simple replacement of the codec verb write with
AC_VERB_SET_PIN_WIDGET but do more sanity checks and filter out
superfluous pin-control bits if they don't fit with the corresponding
pin capabilities.
Some codecs are screwed up or ignore the command when such a wrong bit
is set. These helpers will avoid such secret errors.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since it's not always safe to assume that the vmaster hook is purely the
mute-LED control, add the flag indicating whether to expose the mute-LED
enum control or not. Currently, conexant codec sets this off for non-HP
laptops where EAPD may be used really as EAPD.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create snd_hda_add_vmaster_hook() and snd_hda_sync_vmaster_hook()
helper functions to handle the mute-LED in vmaster hook more
commonly. In the former function, a new enum control "Mute-LED Mode"
is added. This provides user to choose whether the mute-LED should be
turned on/off explicitly or to follow the master-mute status.
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the driver is changed to use vmaster or a new slave element is
added by the improvement of the parser code, user may face often the
silent output because of the muted slave mixer although Master volume
is properly set. And they complain. And I get upset.
Although such a mixer element should be initialized via "alsactl init",
it'd be more user-friendly if the known output slaves are unmuted and
set to 0dB so that user can control the volume only with Master as
default. Since Master is still set muted as default even with this
change, no risk of the speaker blow up, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've seen a problem with a pop-noise at suspend/resume on a HP
machine with ALC269, and it turned out to be an issue that the
controller going to D3 while the codec is unused.
When the device is once suspended and resumed and kept unused, the
driver doesn't initialize the codecs. Instead, the codec chips are
set up dynamically at the first usage. Now, suppose the device going
to suspend again before the codec is set up. The controller is turned
off to D3 while the codec chips are untouched. This caused a pop
noise because the codec chip might have been turned on implicitly by
the hardware.
As a workaround, the codec chip needs to be set to D3 when going to
suspend no matter whether it was used or not. Also, for making it
happening, the controller has to be always set up in the resume path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some codecs don't supply the mute amp-capabilities although the lowest
volume gives the mute. It'd be handy if the parser provides the mute
mixers in such a case.
This patch adds an extension amp-cap bit (which is used only in the
driver) to represent the min volume = mute state. Also modified the
amp cache code to support the fake mute feature when this bit is set
but the real mute bit is unset.
In addition, conexant cx5051 parser uses this new feature to implement
the missing mute controls.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42825
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In most cases, the slave strings for vmaster are identical between
volumes and switches except for "xxx Volume" and "xxx Switch" suffix.
Now snd_hda_add_vmaster() takes the optional suffix argument so that
each string can be composed with the given suffix, and we can share the
slave name strings in both volume and switch calls nicely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This typo caused the wrong codec's nid to be checked for wcaps type.
As a result, sometimes speakers would duplicate the output sent to
HDMI output.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/924320
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the driver supports up to 10 streams. This is a restriction in
hda_intel.c and hda_codec.c: in the former, the fixed array size limits
the amount, and in the latter, the fixed device-number assignment table
(in get_empty_pcm_device()) limits the possibility.
This patch reduces the restriction by
- using linked list for managing PCM instances in hda_intel.c, and
- assigning non-fixed device numbers for the extra devices
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In snd_hda_check_board_codec_sid_config(), not only comparing with the
exact value but allow the bit-mask comparison for vendor-only, etc.
Tested-by: Linus Torvalds <torvalds@linux-foundation.org>
Tested-by: Dirk Hohndel <hohndel@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The multiple headphone pins are usually handled as copied from the same
source, not as individual channels like front and surround. Thus it'd
be more correct to avoid the channel suffix for "Headphone" pin labels
in snd_hda_get_pin_label() but give an index number instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function now gives more unique names for the output pins by adding
some prefix and suffix for the location and the channels. Otherwise, it
can pass the index number.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai