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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>
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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
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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Back-merged for refactoring beep stuff.
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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>
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This leaks the beep input device after module unload, which leads to
Oops.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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It's compatible with ALC282.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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Reported-by: Harald Becker <ralda@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Just as for analog codecs, a jack that isn't suitable for detection
(in this case, NO_PRESENCE was set) should be a phantom Jack
instead of a normal one.
Thanks to Raymond Yau for spotting.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/961286
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=903869
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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