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The interrupt handler azx_interrupt will call azx_update_rirb,
which may call snd_hda_queue_unsol_event, snd_hda_queue_unsol_event
will dereference chip->bus pointer.
The problem is we alloc chip->bus in azx_codec_create
which will be called after we enable IRQ and enable unsolicited
event in azx_probe.
This will cause Oops due dereference NULL pointer. I meet it, good luck:)
[Rearranged the NULL check before the tracepoint and added another
NULL check of bus->workq -- tiwai]
Signed-off-by: Wang YanQing <udknight@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The older Conexant codecs have up to two EAPDs and these are supposed
to be rather statically turned on. The new generic parser code
assumes the dynamic on/off per path usage, thus it resulted in the
silent output on some machines.
This patch fixes the problem by simply assuming the static EAPD on for
such old Conexant codecs as we did until 3.8 kernel.
Reported-and-tested-by: Christopher K. <c.krooss@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The audio driver mistakenly allows 64 bit addresses to be created for
the audio driver on Nvidia GPUs. Unfortunately, the hardware normally
only supports up to 40 bits of DMA. This can cause system panics as
well as misdirected data when the address is > 40 bits as the upper
part the address is truncated.
Signed-off-by: Mike Travis <travis@sgi.com>
Reviewed-by: Mike Habeck <habeck@sgi.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some Asihpi formats are not supported or invalid, and their mapping to
ALSA format is set to -1.
Before performing the format conversion into ALSA bitwise formats,
add a consistency check for the requested format, as done in
snd_card_asihpi_playback_formats().
Compile tested only.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.
Change such conversions to use this function and silence sparse
warnings.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This enables better volume controls than the current model parser.
Also, because the original quirk for X220 was added to fix
docking station support, add the TP410 fixup instead.
Reported-by: Willian Jon McCann <william.jon.mccann@gmail.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ARM cannot handle udelay for more than 2 miliseconds, so we
should use mdelay instead for those.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It's yet another ALC269-variant.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The commit [b209c4df: ALSA: emu10k1: cache emu1010 firmware] broke the
firmware loading of the dock, just (mistakenly) ignoring a different
firmware for docks on some models. This patch revives them again.
Bugzilla: https://bugs.archlinux.org/task/34865
Reported-and-tested-by: Tobias Powalowski <tobias.powalowski@googlemail.com>
Cc: <stable@vger.kernel.org> [v3.8+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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These are being reported as being so noisy at high mic boost levels,
so they are unusable in practice.
Therefore artificially limit the boosts.
BugLink: https://bugs.launchpad.net/bugs/1089795
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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.
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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The Stumpy ChromeBox needs its pin configs fixed up.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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This reverts commit 6ab317419c62850a71e2adfd1573e5ee87d8774f.
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>
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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>
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* 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>
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Back-merge for cleaning up usb-audio code the recent commit modified,
and further UAC2 autoclock patches.
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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>
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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>
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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>
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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>
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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>
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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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