Age | Commit message (Collapse) | Author |
|
Add documentation for the controls present in the SPDIF input
controller driver
Signed-off-by: Damien.Horsley <Damien.Horsley@imgtec.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
It's totally outdated. We need a revised version later, maybe better
integrated into kernel doc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Added missing model entries and updated the codec names.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Modern machines tend to have only one headset jack nowadays, and they
often need these quirks. Let's allow them applicable via model
option for ease of debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Add documentation describing Jack kcontrols and how to use them
with HD-Audio and ASoC.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.1
More updates for v4.1, pretty much all drivers:
- Lots of cleanups from Lars, mainly moving things from the CODEC level
to the card level.
- Continuing improvements to rcar from Morimoto-san, pcm512x from
Howard and Peter, the Intel platforms from Vinod, Jie, Jin and Han,
and to rt5670 from Bard.
- Support for some non-DSP Qualcomm platforms, Google's Storm
platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC.
|
|
Currently 'Playback Volume' is the correct way to express an analogue
volume control. However, this control name has initialisation defaults
applied when using 'alsactl restore' and in some cases this is not
appropriate. An example would be a control that has a selection of
0db and -6dB of gain that is intended to set the fullscale ouput
voltage of a DAC. The TI pcm512x family of DAcs have such a control.
In this case the device/driver reset defaults are preferred.
Signed-off-by: Howard Mitchell <hm@hmbedded.co.uk>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
David suggested that the name "power_mgmt" is too ambiguous. Rename
the flag with a bit clearer one "power_save_node".
Also, add the corresponding description to HD-Audio.txt, too.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Audio timestamps can be extracted from sample counters, wall clocks,
PHC clocks (Ethernet AVB), on-demand synchronized snapshots. This
patch provides the ability to report timestamping capabilities, select
timestamp types and retrieve timestamp accuracy, if supported.
Details can be found in Documentations/sound/alsa/timestamping.txt
This functionality is introduced by reclaiming the reserved_aligned
field introduced by commit9c7066aef4a5eb8e4063de28f06c508bf6f2963a
in snd_pcm_status to provide userspace with selection/query capabilities.
Additional driver_tstamp and audio_tstamp_accuracy fields are also added.
snd_pcm_mmap_status remains a read-only structure with only
the audio timestamp value accessible from user space. The selection
of audio timestamp type is done through snd_pcm_status only
This commit does not impact ABI and does not impact the default
behavior. By default audio timestamp is aligned with hw_pointer and
reports the DMA position. Backwards compatibility is handled by using
the HDAudio wall clock for playback and the hw_ptr for all other
cases.
For timestamp selection a new STATUS_EXT ioctl is introduced with
read/write parameters. Alsa-lib will be modified to make use of
STATUS_EXT.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Also update the documentation to the latest state.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This patch adds a new proc entry for PCM substreams to inject an
XRUN. When a PCM substream is running and any value is written to its
xrun_injection proc file, the driver triggers XRUN. This is a useful
feature for debugging XRUN and error handling code paths.
Note that this entry is enabled only when CONFIG_SND_PCM_XRUN_DEBUG is
set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
ALSA PCM core has a mechanism tracking the PCM hwptr updates for
analyzing XRUNs. But its log is limited (up to 10) and its log output
is a kernel message, which is hard to handle.
In this patch, the hwptr logging is moved to the tracing
infrastructure instead of its own. Not only the hwptr updates but
also XRUN and hwptr errors are recorded on the trace log, so that user
can see such events at the exact timing.
The new "snd_pcm" entry will appear in the tracing events:
# ls -F /sys/kernel/debug/tracing/events/snd_pcm
enable filter hw_ptr_error/ hwptr/ xrun/
The hwptr is for the regular hwptr update events. An event trace
looks like:
aplay-26187 [004] d..3 4012.834761: hwptr: pcmC0D0p/sub0: POS: pos=488, old=0, base=0, period=1024, buf=16384
"POS" shows the hwptr update by the explicit position update call and
"IRQ" means the hwptr update by the interrupt,
i.e. snd_pcm_period_elapsed() call. The "pos" is the passed
ring-buffer offset by the caller, "old" is the previous hwptr, "base"
is the hwptr base position, "period" and "buf" are period- and
buffer-size of the target PCM substream.
(Note that the hwptr position displayed here isn't the ring-buffer
offset. It increments up to the PCM position boundary.)
The XRUN event appears similarly, but without "pos" field.
The hwptr error events appear with the PCM identifier and its reason
string, such as "Lost interrupt?".
The XRUN and hwptr error reports on kernel message are still left, can
be turned on/off via xrun_debug proc like before. But the bit 3, 4, 5
and 6 bits of xrun_debug proc are dropped by this patch. Also, along
with the change, the message strings have been reformatted to be a bit
more consistent.
Last but not least, the hwptr reporting is enabled only when
CONFIG_SND_PCM_XRUN_DEBUG is set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This document was not really up-to-date. Add recent additions to this
standard - based on what the HDA driver currently does, which is some
kind of a de facto standard.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Just add the PCI ID for the STX II. It appears to work the same as the
STX, except for the addition of the not-yet-supported daughterboard.
Tested-by: Mario <fugazzi99@gmail.com>
Tested-by: corubba <corubba@gmx.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Just add missing recent entries.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
HP Spectre 13 has the IDT 92HD95 codec, and BIOS seems to set the
default high-pass filter in some "safer" range, which results in the
very soft tone from the built-in speakers in contrast to Windows.
Also, the mute LED control is missing, since 92HD95 codec still has no
HP-specific fixups for GPIO setups.
This patch adds these missing features: the HPF is adjusted by the
vendor-specific verb, and the LED is set up from a DMI string (but
with the default polarity = 0 assumption due to the incomplete BIOS on
the given machine).
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=74841
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Fixed multiple spelling errors.
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Carlos E. Garcia <carlos@cgarcia.org>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
|
|
Update the ASoC overview to bring it up to date with the current code base
and include multi-component.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
|
Correct spelling typo in documentation/alsa
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|
|
Using the headset mic model will cause the headset mic to be labeled
"headset mic" instead of just "mic".
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Add documentation describing DPCM with examples of a DSP based
smart phone.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
|
Update the machine driver documentation and bring it up to date
with the current code base. This includes multi component.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
|
Update the DAPM documentation and bring it up to date with the current
code base. This includes API changes and new widgets.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
|
Update the platform class driver documentation and bring it up to date
with the current code base. This includes multi component and DSP.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
|
Update the codec class driver documentation and bring it up to date
with the current code base. This includes API changes, regmap and
multi component.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial
Pull trivial tree from Jiri Kosina:
"The usual trivial updates all over the tree -- mostly typo fixes and
documentation updates"
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (52 commits)
doc: Documentation/cputopology.txt fix typo
treewide: Convert retrun typos to return
Fix comment typo for init_cma_reserved_pageblock
Documentation/trace: Correcting and extending tracepoint documentation
mm/hotplug: fix a typo in Documentation/memory-hotplug.txt
power: Documentation: Update s2ram link
doc: fix a typo in Documentation/00-INDEX
Documentation/printk-formats.txt: No casts needed for u64/s64
doc: Fix typo "is is" in Documentations
treewide: Fix printks with 0x%#
zram: doc fixes
Documentation/kmemcheck: update kmemcheck documentation
doc: documentation/hwspinlock.txt fix typo
PM / Hibernate: add section for resume options
doc: filesystems : Fix typo in Documentations/filesystems
scsi/megaraid fixed several typos in comments
ppc: init_32: Fix error typo "CONFIG_START_KERNEL"
treewide: Add __GFP_NOWARN to k.alloc calls with v.alloc fallbacks
page_isolation: Fix a comment typo in test_pages_isolated()
doc: fix a typo about irq affinity
...
|
|
Fix double words "is is" in Documentations.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Acked-by: Rob Landley <rob@landley.net>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
|
|
Corrected the word avarage to average in the file compress_offload.txt.
Signed-off-by: Stefan Huber <steffhip@gmail.com>
Signed-off-by: Matthias Schid <aircrach115@gmail.com>
Signed-off-by: Simon Puels <simon.puels@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
|
|
VAIO-Z laptops need to use the specific DAC for the speaker output
by some unknown reason although the codec itself supports the flexible
connection. So we implemented a workaround by a new flag,
no_primary_hp, for assigning the speaker pin first.
This worked until 3.8 kernel, but it got broken because the driver
learned for a better multi-io pin mapping, and not it can assign two
mic pins for multi-io. Since the multi-io requires to be the primary
output, the hp and two mic pins are assigned in prior to the speaker
in the end.
Although the machine has two mic pins, one of them is used as a noise-
canceling headphone, thus it's no real retaskable mic jack. Thus, at
best, we can disable the multi-io assignment and make the parser
behavior back to the state before the multi-io.
This patch adds again a new flag, no_multi_io, to indicate that the
device has no multi-io capability, and set it in the fixup for
VAIO-Z. The no_multi_io flag itself can be used generically, added
via a helper line, too.
Reported-by: Tormen <my.nl.abos@gmail.com>
Reported-by: Adam Williamson <awilliam@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The current fixup for dell-bios model with STAC9228 codec contains the
override of pin 0x0c for analog mic. But this is actually just adding
a bogus pin and confuses the parser. Better to remove it for the
auto-mic switching.
Meanwhile, for a possible regression, keep the old configuration as
model=dell-bios-amic, so that people can test it again quickly.
Tested on Dell 1420n laptop.
Reported-and-tested-by: Eric Shattow <lucent@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Correct typo (double words) in documentations.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
|
|
Add new known codecs, and fix up tabs.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
These headset jacks keep coming in on more and more platforms, and
it's possible I don't catch them all. Make it easier to test and
verify by making models.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
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.
|
|
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>
|
|
Back-merged for refactoring beep stuff.
|
|
Correct spelling typos in Documentation/sound/alsa
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
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>
|
|
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>
|
|
this add new API for sound compress to support gapless playback.
As noted in Documentation change, we add API to send metadata of encoder and
padding delay to DSP. Also add API for indicating EOF and switching to
subsequent track
Also bump the compress API version
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.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>
|
|
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Change the power_save_controller option to bint from bool so that user
can override the runtime PM capability bit and force to enable or
disable the runtime PM.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The async unlink behavior has been working over years. The option was
provided only as a workaround for 2.4.x kernel. Let's get rid of it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This is a merge of a topic branch containing the support for the new
channel map API using control elements.
|
|
Set the default value of position_fix -1, and allow user passing
position_fix=0 explicitly to set the "auto" position-fix mode.
Otherwise the auto mode may be switched to others like COMBO of
VIACOMBO when the controller prefers it, thus user can't set the auto
mode any longer.
Also updated the documentation appropriately, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
For following the standard, define more channel map positions and
shuffle the items a bit:
- As both PulseAudio and gstreamer define MONO channel position
explicitly, we should follow that, too. The mono streams point to
this channel position unless they are explicitly assigned to certain
channel positions.
- Top-front-* and Top-rear-* positions are added, carried from
PulseAudio's definitions.
- Move NA and MONO definitions at the top of table right after
UNKNOWN, since these are more abstract in comparison with other
practical positions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
There is already a set of channel position definitions in alsa-lib
mixer.h, and it'd be more practical to keep the same order for the
PCM channel map, too. The value is shifted with 1 to keep zero for
UNKNOWN.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|