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Originally snd_hrtimer_callback() used iprtd->period_time for
some jiffies based estimation to determine the right moment
to call snd_pcm_period_elapsed(). As timer drifts may well be a
problem, this was changed in commit b4e82b5b785670b6 to be based
on buffer transmission progress, using iprtd->offset and
runtime->buffer_size to calculate the amount of data since last
period had elapsed.
Unfortunately, iprtd->offset counts in bytes, while
runtime->buffer_size counts frames, so adding these to find some
delta is like comparing apples and oranges, and eventually results
in negative delta values every now and then. This is no big harm,
because it simply causes snd_pcm_period_elapsed() being called
more often than necessary, as negative delta is taken for a
large unsigned value by implicit conversion rule.
Nonetheless, the calculation is broken, so one would replace
the runtime->buffer_size by its equivalent in bytes.
But then, there are chances snd_pcm_period_elapsed() is called
late, because calculating the moment for the elapsed period
into delta is based against the iprtd->last_offset, which is not
necessarily the first byte of the period in question, but some
random byte which the FIQ handler left us with in r8/r9 by
accident. Again, negative impact is low, as there are plenty of
periods already prefilled with data, and snd_pcm_period_elapsed()
will probably be called latest when the following period is
reached. However, the calculation is conceptually broken, and we
are best off removing the clever stuff altogether.
snd_pcm_period_elapsed() is now simply called once everytime
snd_hrtimer_callback() is run, which may not be most accurate,
but at least this way we are quite sure we dont miss an end of
period. There is not much extra effort wasted by superfluous
calls to snd_pcm_period_elapsed(), as the timer frequency
closely matches the period size anyway.
Signed-off-by: Oskar Schirmer <oskar@scara.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The recent kernels got regressions on ASUS W7J with ALC660 codec where
no sound comes out. After a long debugging session, we found out that
setting the pin control on the unused NID 0x10 is mandatory for the
outputs. And, it was found out that another magic of NID 0x0f that is
required for other ASUS laptops isn't needed on this machine.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66081
Reported-and-tested-by: Andrey Lipaev <lipaev@mail.ru>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_bytes_put treats the data in the binary control as big endian
words, however snd_soc_bytes_get uses the endian of the host machine.
This causes the two functions to be inconsistant with how the mask is
applied on little endian machines.
This patch applies the big_endian format used in snd_soc_bytes_put to
snd_soc_bytes_get.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The array limits are supposed to be in units of u32 instead of in bytes.
The current code has a potential array overflow.
Fixes: c614475b0ea9 ('ALSA: dice: add a proc file to show device information')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This machine also has mono output if run through DAC node 0x03.
Cc: stable@vger.kernel.org (v3.10+)
BugLink: https://bugs.launchpad.net/bugs/1256212
Tested-by: David Chen <david.chen@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the previous commit 1f0bbf03cb82 added the pin config for the bass
speaker, this patch adds the corresponding LFE-only channel map on
ASUS ET2700.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a fixup entry for the missing bass speaker pin 0x16 on ASUS ET2700
AiO desktop. The channel map will be added in the next patch, so that
this can be backported easily to stable kernels.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This both devices need limit for internal dmic.
[cosmetic change; renamed fixup name by tiwai]
Signed-off-by: Oleksij Rempel <linux@rempel-privat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current generic parser assumes blindly that the volume and mute
amps are found in the aamix node itself. But on some codecs,
typically Analog Devices ones, the aamix amps are separately
implemented in each leaf node of the aamix node, and the current
driver can't establish the correct amp controls. This is a regression
compared with the previous static quirks.
This patch extends the search for the amps to the leaf nodes for
allowing the aamix controls again on such codecs.
In this implementation, I didn't code to loop through the whole paths,
since usually one depth should suffice, and we can't search too
deeply, as it may result in the conflicting control assignments.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65641
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the setting of the register KIRKWOOD_PLAYCTL which did
always streaming on both I2S and SPDIF, ignoring the DAI ID.
The bug was introduced by the commit 75b9b65ee5a
"ASoC: kirkwood: add S/PDIF support"
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch removes the 32 bits format which is not supported by S/PDIF
output.
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_pcm_limit_hw_rates() will initialize the minimum and maximum sample rate for
the PCM stream based on the rates specified in the rates field. Since we call
snd_pcm_limit_hw_rates() after soc_pcm_init_runtime_hw() it will essentially
overwrite the min and max rate set in soc_pcm_init_runtime_hw(). This may cause
the minimum or maximum rate to be set to a value outside the range of one of the
components if one of the components sets either SNDRV_PCM_RATE_CONTINUOUS or
SNDRV_PCM_RATE_KNOT and the other component specified a discrete rate via
SNDRV_PCM_RATE_[0-9]* that is outside of the first component's rate range. To
fix this first calculate the minimum and maximum rates using
snd_pcm_limit_hw_rates() and then on top of that apply the contraints specified
in the snd_soc_pcm_stream structs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Takashi iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In order to make sure that the sample rate is in the supported range of both
components the maximum rate of the card should be the minimum of the maximum
rate of each components. There is one special case to consider though, if
max_rate is set to 0 this means there is no maximum specified, so use
min_not_zero() macro which will give use the desired result.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Takashi iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
These are managed automatically in current revisions.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As the priv is not assigned to card->drvdata, it is NULL, so when
unload module, it will cause NULL pointer oops.
Assign priv to card->drvdata to fix this issue.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the hp mic pin has no VREF bits, the driver forgot to set PIN_IN
bit. Spotted during debugging old MacBook Airs.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a headphone jack is configurable as input, the generic parser
tries to make it retaskable as Headphone Mic. The switching can be
done smoothly if Capture Source control exists (i.e. there is another
input source). Or when user explicitly enables the creation of jack
mode controls, "Headhpone Mic Jack Mode" will be created accordingly.
However, if the headphone mic is the only input source, we have to
create "Headphone Mic Jack Mode" control because there is no capture
source selection. Otherwise, the generic parser assumes that the
input is constantly enabled, thus the headphone is permanently set
as input. This situation happens on the old MacBook Airs where no
input is supported properly, for example.
This patch fixes the problem: now "Headphone Mic Jack Mode" is created
when such an input selection isn't possible.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For Wireless USB audio devices, use multiple isoc packets per URB for
inbound endpoints with a datainterval < 5. This allows the WUSB host
controller to take advantage of bursting to service endpoints whose
logical polling interval is less than the 4ms minimum polling interval
limit in WUSB.
Signed-off-by: Thomas Pugliese <thomas.pugliese@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most Thinkpad Edge series laptops use conexant codec, so far although
the codecs have different minor Vendor Id and minor Subsystem Id,
they all belong to the cxt5066 family, this change can make the
mute/mic-mute LEDs support more generic among cxt_5066 family.
This design refers to the similar solution for the realtek codec
ALC269 family in the patch_realtek.c.
Cc: Alex Hung <alex.hung@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use bus->power_keep_link_on instead. The controller shouldn't go to
D3 when the link isn't reset, so essentially avoiding the link reset
means avoiding the runtime PM.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the missing pcbeep default amp for ALC668.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise we'll skip sync on resume.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
55e5b6fd5af04b6d8b0ac6635edf49476ff298ba
(ASoC: rsnd: use regmap instead of original register mapping method)
support regmap/regmap_field on Renesas sound driver.
It needs CONFIG_REGMAP now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A collection of small fixes in HD-audio quirks and runtime PM, ASoC
rcar, abs8500 and other codecs. Most of commits are for stable
kernels, too.
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Merge tag 'sound-fix2-3.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull second set of sound fixes from Takashi Iwai:
"A collection of small fixes in HD-audio quirks and runtime PM, ASoC
rcar, abs8500 and other codecs. Most of commits are for stable
kernels, too"
* tag 'sound-fix2-3.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Set current_headset_type to ALC_HEADSET_TYPE_ENUM (janitorial)
ALSA: hda - Provide missing pin configs for VAIO with ALC260
ALSA: hda - Add headset quirk for Dell Inspiron 3135
ALSA: hda - Fix the headphone jack detection on Sony VAIO TX
ALSA: hda - Fix missing bass speaker on ASUS N550
ALSA: hda - Fix unbalanced runtime PM notification at resume
ASoC: arizona: Set FLL to free-run before disabling
ALSA: hda - A casual Dell Headset quirk
ASoC: rcar: fixup dma_async_issue_pending() timing
ASoC: rcar: off by one in rsnd_scu_set_route()
ASoC: wm5110: Add post SYSCLK register patch for rev D chip
ASoC: ab8500: Revert to using custom I/O functions
ALSA: hda - Also enable mute/micmute LED control for "Lenovo dock" fixup
ALSA: firewire-lib: include sound/asound.h to refer to snd_pcm_format_t
ALSA: hda - Select FW_LOADER from CONFIG_SND_HDA_CODEC_CA0132_DSP
ALSA: hda - Enable mute/mic-mute LEDs for more Thinkpads with Realtek codec
ASoC: rcar: fixup mod access before checking
This commit fix out of specification about the value of FDF field in out packet
with 'no data'. This affects blocking mode.
According to IEC 61883-6, there is two way to generate AMDTP packets include no
data in blocking mode.
Way 1. an empty packet defined in IEC 61883-1
- Size of packet is 2 quadlets.
- The value of FDF is sfc.
- The packet includes only CIP headers
Way 2. a special non-empty packet defined in IEC 61883-6
- Size of packet is following to blocking mode
- The value of FDF is 0xff. This value is 'NO-DATA'. This means 'The receiver'
must ignore all the data in a CIP with this FDF code'.
- The packet includes dummy data.
But current implementation is a combination of them.
- Size of packet is 2 (way 1)
- FDF = 0xff (way 2)
This causes BeBoB chipset cannot sound.
This patch applies Way 1.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
current_headset_type should be of the HEADSET_TYPE enum, not the
HEADSET_MODE enum. Since ALC_HEADSET_TYPE_UNKNOWN and ALC_HEADSET_MODE_UNKNOWN
are both 0, this patch is just janitorial.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some models (or maybe depending on BIOS version) of Sony VAIO with
ALC260 give no proper pin configurations as default, resulting in the
non-working speaker, etc. Just provide the whole pin configurations
via a fixup.
Reported-by: Matthew Markus <mmarkus@hearit.co>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bunch of device specific fixes, nothing with a general impact here.
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Merge tag 'asoc-v3.13-5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.13
A bunch of device specific fixes, nothing with a general impact here.
The laptop has a built-in speaker on NID 0x1a. It's an LFE only on
the right channel, so we need to provide an explicit chmap, too.
There might be other surround speakers, but they can fixed in addition
at later point, so let's fix the easier bass speaker at first.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65091
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull slave-dmaengine changes from Vinod Koul:
"This brings for slave dmaengine:
- Change dma notification flag to DMA_COMPLETE from DMA_SUCCESS as
dmaengine can only transfer and not verify validaty of dma
transfers
- Bunch of fixes across drivers:
- cppi41 driver fixes from Daniel
- 8 channel freescale dma engine support and updated bindings from
Hongbo
- msx-dma fixes and cleanup by Markus
- DMAengine updates from Dan:
- Bartlomiej and Dan finalized a rework of the dma address unmap
implementation.
- In the course of testing 1/ a collection of enhancements to
dmatest fell out. Notably basic performance statistics, and
fixed / enhanced test control through new module parameters
'run', 'wait', 'noverify', and 'verbose'. Thanks to Andriy and
Linus [Walleij] for their review.
- Testing the raid related corner cases of 1/ triggered bugs in
the recently added 16-source operation support in the ioatdma
driver.
- Some minor fixes / cleanups to mv_xor and ioatdma"
* 'next' of git://git.infradead.org/users/vkoul/slave-dma: (99 commits)
dma: mv_xor: Fix mis-usage of mmio 'base' and 'high_base' registers
dma: mv_xor: Remove unneeded NULL address check
ioat: fix ioat3_irq_reinit
ioat: kill msix_single_vector support
raid6test: add new corner case for ioatdma driver
ioatdma: clean up sed pool kmem_cache
ioatdma: fix selection of 16 vs 8 source path
ioatdma: fix sed pool selection
ioatdma: Fix bug in selftest after removal of DMA_MEMSET.
dmatest: verbose mode
dmatest: convert to dmaengine_unmap_data
dmatest: add a 'wait' parameter
dmatest: add basic performance metrics
dmatest: add support for skipping verification and random data setup
dmatest: use pseudo random numbers
dmatest: support xor-only, or pq-only channels in tests
dmatest: restore ability to start test at module load and init
dmatest: cleanup redundant "dmatest: " prefixes
dmatest: replace stored results mechanism, with uniform messages
Revert "dmatest: append verify result to results"
...
When a codec is resumed, it keeps the power on while the resuming
phase via hda_keep_power_on(), then turns down via
snd_hda_power_down(). At that point, snd_hda_power_down() notifies
the power down to the controller, and this may confuse the refcount if
the codec was already powered up before the resume.
In the end result, the controller goes to runtime suspend even before
the codec is kicked off to the power save, and the communication
stalls happens.
The fix is to add the power-up notification together with
hda_keep_power_on(), and clears the flag appropriately.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The FLL must be placed into free-run mode before disabling
to allow it to entirely shut down.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
You're looking at a casual headset patch,
for a specific hardware it will match,
and suddenly, the headset jack will work,
so please apply this simple quirk!
BugLink: https://bugs.launchpad.net/bugs/1253038
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DMAEngine will stall without this patch
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
If "id == ARRAY_SIZE(routes)" then we read one space beyond the end of
the routes[] array.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Certain registers require patching after the SYSCLK has been brought up
add support for this into the CODEC driver.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
It's been reported that these break audio on Snowball so revert them
until a Snowball user has time to investigate.
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
