Some of the Digital mixer kcontrol max values were off by 1 not allowing a max of 0dB.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Explained by Takashi in <s5hfwbtmz0q.wl%tiwai@suse.de>
> The reason is because get_min_max*() isn't called in the place you
> created these controls, and get_min_max() would be called only for
> integer volumes later even if uninitialized. A short cut for booleans.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the PCM read/write loop, the driver calls snd_pcm_update_hw_ptr()
at each time at the beginning of the loop. Russell King reported that
this hogs CPU significantly.
The current code assumes that the pointer callback is very fast and
cheap, also not too much fine grained. It's not true in all cases.
When the pointer advances short samples while the read/write copy has
been performed, the driver updates the hw_ptr and gets avail > 0
again. Then it tries to read/write these small chunks. This repeats
until the avail really gets to zero.
For avoiding this situation, a simple workaround is to call
snd_pcm_update_hw_ptr() only once at starting the loop, assuming that
the read/write copy is performed fast enough. If the available count
becomes short, it goes to snd_pcm_wait_avail() anyway, and this
processes right.
Tested-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When pdata->reset_pin is passed with a negative value (means gpio
is invalid), then chip->reset_pin will not be assigned to a vaule,
it will use default value 0. This will cause unexpected behavior.
So, add this patch to correct.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Slightly more than expected as rc7, but all are reasonablly small fixes.
A few additions of HD-audio fixup entries, a couple of other regression
fixes including a revert, and a few other trivial oneliners.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Slightly more than expected as rc7, but all are reasonablly small
fixes. A few additions of HD-audio fixup entries, a couple of other
regression fixes including a revert, and a few other trivial
oneliners."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: sh: fix migor.c compilation
ALSA: HDA: Lessen CPU usage when waiting for chip to respond
Revert "ALSA: hda - Set codec to D3 forcibly even if not used"
ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup
ALSA: hdsp - Provide ioctl_compat
ALSA: hda/realtek - Add missing CD-input pin for MSI-7350 mobo
ALSA: hda/realtek - Add a fixup for Acer Aspire 5739G
ALSA: echoaudio: Remove incorrect part of assertion
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>
They pollute the global namespace and cause sparse to complain.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
It can only be used with a machine driver so the idiomatic thing for
ASoC is to select this driver from the machine driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The controller is compatible with HD-audio 1.0a with some specific
restrictions.
- The BDLE entries can't be over 4k boundary
- No position-buffer and no MSI
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Makes sparse happy and avoids polluting the global namespace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
We need to read the real register values
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
An API update which wasn't sufficiently thorough in updating the tree...
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Build fix for SH in 3.4
An API update which wasn't sufficiently thorough in updating the tree...
We're trying to remove all usage of the ASoc level cache and I/O code and
for a device like this with a pretty sparse register map the rbtree cache
is a better idea anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add driver for running I2S with the MSP-block.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
[Fixed trailing whitespace -- broonie]
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix a recent compilation breakage, caused by a change in SH clock API.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
When an IRQ for some reason gets lost, we wait up to a second using
udelay, which is CPU intensive. This patch improves the situation by
waiting about 30 ms in the CPU intensive mode, then stepping down to
using msleep(2) instead. In essence, we trade some granularity in
exchange for less CPU consumption when the waiting time is a bit longer.
As a result, PulseAudio should no longer be killed by the kernel
for taking up to much RT-prio CPU time. At least not for *this* reason.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Tested-by: Arun Raghavan <arun.raghavan@collabora.co.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the fixup helper functions in patch_realtek.c to hda_auto_parser.c
so that they can be used in other codec drivers like patch_conexant.c.
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>
The upper limit of the available minors isn't necessarily 128 + unit,
but it's rather up to 256. Fixing this allows more than 8 devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are left-over codes from the ancient days with the static device
number limitation of 8. Actaully OSS can support up to 16 cards.
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>
The call for alc_auto_parse_customize_define() must be done after the
fixup pre-probe initialization. Otherwise SKU_IGNORE fixup won't work
properly (e.g. HP RP5800 with ALC662 codec).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we instantiate an aux_dev we use a fake rtd as part of the process
which doesn't have a dai_link associated with it. Fix the dpcm startup
code to cope with this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
snd_hdsp uses its own ioctls to acquire config- and status information.
Expose the corresponding ioctl handler via ioctl_compat, so that 32bit applications can use it on 64bit kernels.
Signed-off-by: Andre Schramm <andre.schramm@iosono-sound.com>
Reviewed-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
None of the machines uses the gain ramp possibility for HS/HF.
This code path is mostly unused and it does not reduces the pop
noise on the output (it alters it to sound a bit different).
The preferred method to reduce pop noise is to use ABE.
Remove the gain ramp, and related features form the driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since this is a generic API which should support any userspace interface
for reporting jacks update the documentation a little to make that a bit
clearer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allocating the SSP DMA parameters in startup, freeing it in
shutdown instead of freeing and re-allocating it in hw_params.
After doing that, the logic is clear and more safe.
Signed-off-by: guoyh <guoyh@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acer Aspire 5739G requires the same fix-up for 4930G to support the
surround / bass speakers.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43180
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This assertion seems to imply that chip->dsp_code_to_load is a pointer.
It's actually an integer handle on the actual firmware, and 0 has no
special meaning.
The assertion prevents initialisation of a Darla20 card, but would also
affect other models. It seems it was introduced in commit dd7b254d.
ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting...
ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000
ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware...
ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20
------------[ cut here ]------------
WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]()
Hardware name: Dell DM051
BUG? (!chip->dsp_code_to_load || !chip->comm_page)
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As good as nothing exciting here; just a few trivial fixes for
various ASoC stuff.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound sound fixes from Takashi Iwai:
"As good as nothing exciting here; just a few trivial fixes for various
ASoC stuff."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: omap-pcm: Free dma buffers in case of error.
ASoC: s3c2412-i2s: Fix dai registration
ASoC: wm8350: Don't use locally allocated codec struct
ASoC: tlv312aic23: unbreak resume
ASoC: bf5xx-ssm2602: Set DAI format
ASoC: core: check of_property_count_strings failure
ASoC: dt: sgtl5000.txt: Add description for 'reg' field
ASoC: wm_hubs: Make sure we don't disable differential line outputs
Add the PCI ID of the Asus Xonar DGX card; it's otherwise
identical with the DG.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nothing terribly exciting here, a bunch of small and simple fixes
scattered around the place.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for 3.4
Nothing terribly exciting here, a bunch of small and simple fixes
scattered around the place.
Signed-off-by: Oleg Matcovschi <oleg.matcovschi@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
This patch improves playback quality for few sample rates like 8000 and
11025 Hz.
This also fixes an issue observed during testing of pll slave mode. Due
to the issue, on some rare occasions there was no sound output for first
time playback after system boot, though all subsequent playbacks were
fine. It was mainly because of the sequence in which SRM bit was
enabled.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than invalidating the cached DCS value every time the headphone
gain changes store multiple values, indexed by gain. This allows the
optimisation we get from the cache to take effect more often.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This binding doesn't include the nvidia,model or nvidia,audio-routing
properties the other Tegra audio DT bindings have, because this binding
is targetted at a single machine, rather than for any machine using the
tlv320aic23 codec.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As s3c2412-i2s is using the s3c_i2sv2 it should call the more specialised
s3c_i2sv2_register_dai instead of simply calling snd_soc_register_dai.
Without this call the snd_soc_dai_ops structure isn't initialised correctly.
Signed-off-by: Heiko Stuebner <heiko@sntech.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is now very standard behaviour for CODECs so shouldn't be device
specific and we shouldn't really be trying to peer into the register
cache from atomic context anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the conversion to module_init_i2c() the original open coded module
exit function was left. Remove it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Georgi Vlaev <joe@nucleusys.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core allocates the live copies, we shouldn't try to duplicate it and
were buggy trying to do so as we were using uninitialised data for the
control data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We should check dailess before dereferencing.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* commit f9dfbf9 "ASoC: tlv320aic23: convert to soc-cache" leads to
a bug preventing resumeof the codec as regmap expects a 9 bits data
register but 0xFFFF is passed in tlv320aic23_set_bias_level and this
values gets cached preventing any write to the TLV320AIC23_PWR
register as the final value produced by regmap is (register << 9) | value
* this patch solves the problem by only working on the 9 bits the
register contains.
Signed-off-by: Eric Bénard <eric@eukrea.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
It tries to clk_get the clock. And if it failed, it assumes the clock
by default enabled.
Signed-off-by: Richard Zhao <richard.zhao@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Class W can be used for any path where only data from the DAC is routed
to the headphones. Currently we only enable it when the direct DAC to
headphone path is used but it can also be enabled for paths that go via
the output mixer providing the DAC is the only input to the output mixer.
Implement support for this, including updates to the class W status when
the output mixer configuration is changed. This also allows us to enable
the DC servo optimisations for DAC to headphone paths where the output
mixer is used.
In general the direct DAC path is still preferred as this will offer
better performance on most wm_hubs devices but these additional paths
can simplify use case management.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the analogue portions of the checks for class W are the same over
all the devices factor out these checks into wm_hubs and while we're at
it also use wm_hubs_dac_hp_direct() to enable class W optimisations on
more paths.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The optimisations which we can do with caching the headphone DCS result in
wm_hubs have only been enabled in cases where class W is enabled. However,
there are more use cases which can benefit from the cache, especially with
WM8994 series devices with their more advanced digital routing.
Rather than keying off the class W information from the CODECs have a
check in wm_hubs for a suitable path and use that to determine if we can
deploy our headphone optimisations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove writable debugFS permission, use simple_open() and
fix indentation.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes a bug discovered during testing of non pll slave mode.
Due to the bug chip was not getting correctly configured and as a result
there was no sound output while playback. After applying this patch,
both pll and non pll modes work fine.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reduce our stack consumption by moving the params off the stack, they
are reasonably large and might be an issue on platforms with small stacks.
Reported-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ackeded-by: Liam Girdwood <lrg@ti.com>
A workaround for an ASUS laptop and a few ASoC changes;
most of the commits are tagged for stable, too.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A workaround for an ASUS laptop and a few ASoC changes; most of the
commits are tagged for stable, too."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: wm8994: Improve sequencing of AIF channel enables
ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E
ASoC: fsi: update for dmaengine prep_slave_sg fallout.
ASoC: core: Fix card RTD count for deferred probe.
ASoC: cs42l73: don't use negative array index
ASoC: dapm: Ensure power gets managed for line widgets
If a driver using a custom mic detection callback has provided a table
of mic detection rates via platform data then use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use a slightly larger debounce when identifying accessory type and a
slightly smaller one when detecting buttons in response to user feedback
from large scale testing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When we're not actively doing audio we don't need the microphone biases
to be regulated, noise is not important when we are not looking at the
audio signal. Save some power by putting the MICBIAS regulators into
bypass mode when not doing audio.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide an ioctl marshaller for ASoC platform drivers.
This will use the default ALSA handler if no platform
handler exists.
This is also required for DPCM BE PCMs as snd_pcm_info()
will call the ioctl as part of stream startup.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some on SoC DSP HW is very tightly coupled with DMA and DAI drivers. It's
necessary to allow some flexability wrt to PCM operations here so that we
can define a bespoke DPCM trigger() PCM operation for such HW.
A bespoke DPCM trigger() allows exact ordering and timing of component
triggering by allowing a component driver to manage the final enable
and disable configurations without adding extra complexity to other
component drivers. e.g. The McPDM DAI and ABE are tightly coupled on
OMAP4 so we have a bespoke trigger to manage the trigger to improve
performance and reduce complexity when triggering new McPDM BEs.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some component drivers will need to be able to look up their
DAI link substream and RTD data. Provide a mechanism for this.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch allows DPCM to dynamically alter the FE to BE PCM links
at runtime based on mixer setting updates. DAPM is looked up after
every mixer update and we perform a DPCM runtime update if the
mixer has a change of value.
This patchs adds/changes the following :-
o Adds DPCM runtime update core.
o Changes soc_dapm_mixer_update_power() and soc_dapm_mux_update_power()
to return if a change has occured rather than 0. No other users check
atm.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add debugFS files for DPCM link management information.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Dynamic PCM core allows digital audio data to be dynamically
routed between different ALSA PCMs and DAI links on SoC CPUs with
on chip DSP devices. e.g. audio data could be played on pcm:0,0 and
routed to any (or all) SoC DAI links.
Dynamic PCM introduces the concept of Front End (FE) PCMs and Back
End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that
they can dynamically route digital audio data to any supported BE
PCM. A BE PCM has no ALSA device, but represents a DAI link and it's
substream and audio HW parameters.
e.g. pcm:0,0 routing digital data to 2 external codecs.
FE pcm:0,0 ----> BE (McBSP.0) ----> CODEC 0
+--> BE (McPDM.0) ----> CODEC 1
e.g. pcm:0,0 and pcm:0,1 routing digital data to 1 external codec.
FE pcm:0,0 ---
+--> BE (McBSP.0) ----> CODEC
FE pcm:0,1 ---
The digital audio routing is controlled by the usual ALSA method
of mixer kcontrols. Dynamic PCM uses a DAPM graph to work out the
routing based upon the mixer settings and configures the BE PCMs
based on routing and the FE HW params.
DPCM is designed so that most ASoC component drivers will need no
modification at all. It's intended that existing CODEC, DAI and
platform drivers can be used in DPCM based audio devices without
any changes. However, there will be some cases where minor changes
are required (e.g. for very tightly coupled HW) and there are
helpers to support this too.
Somethimes the HW params of a FE and BE do not match or are
incompatible, so in these cases the machine driver can reconfigure
any hw_params and make any DSP perform sample rate / format conversion.
This patch adds the core DPCM code and contains :-
o The FE and BE PCM operations.
o FE and BE DAI link support.
o FE and BE PCM creation.
o BE support API.
o BE and FE link management.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
commit 4183eed2 (ASoC: core: Add signed multi register control) introduced
the variable 'min',but it is not used.
Remove it to fix the following build warning:
sound/soc/soc-core.c: In function 'snd_soc_put_xr_sx':
sound/soc/soc-core.c:2990: warning: unused variable 'min'
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mostly a one to one converion. On one occasion the patch replaces a
snd_soc_read-snd_soc_write sequence with regmap_update_bits though as it helps
to keep the conversion simple.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have never really updated that version number and probably never will, so
just remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Not all advertised rates are available for all sysclk frequencies. Add
additional sysclk based rate constraints.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The sysclock is fixed, so just set it up once in the init callback instead of
setting it repeatably in the hw_params callback.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 980b0bc69 ("ASoC: blackfin: Use dai_fmt") converted the blackfin ASoC
machine drivers to use the dai_links dai_fmt field to setup their DAI format.
For the bf5xx-ssm2602 the commit removed the manual call to snd_soc_dai_set_fmt,
but missed to set the dai_links dai_fmt field.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If FLL bypass is left enabled when we disable the CODEC then the output
clock will be left running which consumes a small amount of additional
current. Only enable bypass when there is an output.
Signed-off-by: Kyung-Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/usb/endpoint.c: In function 'queue_pending_output_urbs':
sound/usb/endpoint.c:298: warning: 'packet' may be used uninitialized in this function
Cc: Daniel Mack <zonque@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds controls for the effects section on the FTU devices.
Some of these controls need volume quirks. They are added to
mixer.c.
[fixed missing break by tiwai]
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is in preparation for more FTU controls to come.
Should help keeping names a bit shorter.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds db gain information to M-Audio Fast Track Ultra (8R) devices.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename mixer_vol_tlv to snd_usb_mixer_vol_tlv and export it to make
it reuseable in mixer_quirks.c.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge snd_maudio_ftu_create_ctl() and snd_ebox44_create_ctl() into
snd_create_std_mono_ctl().
As opposed to the ftu and ebox-44 specific functions, a TLV callback
can be specified for controls created by snd_create_std_mono_ctl().
[fixed minor checkpatch.pl warnings by tiwai]
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While we need to clean up unused single ended line outputs we don't want
to do this if the outputs are in differential mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added support for a control that strobes a bit in
a register to high then back to low (or the inverse).
This is typically useful for hardware that requires
strobing a singe bit to trigger some functionality
and where exposing the bit in a normal single control
would require the user to first manually set then
again unset the bit again for the strobe to trigger.
Added convenience macro.
SOC_SINGLE_STROBE
Added accessor implementations.
snd_soc_get_strobe
snd_soc_put_strobe
Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added control type that can span multiple consecutive codec registers
forming a single signed value in a MSB/LSB manner.
The control dynamically adjusts to the register word size configured
in driver.
Added convenience macro.
SOC_SINGLE_XR_SX
Added accessor implementations.
snd_soc_info_xr_sx
snd_soc_get_xr_sx
snd_soc_put_xr_sx
Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While reading through sound/soc/codecs/wm8994.c I noticed a fair
amount of trailing whitespace. This patch gets rid of it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Slightly larger than normal - the DAPM fix is a "this should always have
worked" type of thing which is very clear and should have no impact on
systems that don't need it. The WM8994 fix is driver specific but
pretty important for that driver.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: updates for 3.4
Slightly larger than normal - the DAPM fix is a "this should always have
worked" type of thing which is very clear and should have no impact on
systems that don't need it. The WM8994 fix is driver specific but
pretty important for that driver.
This ensures a clean startup of the channels, without this change some
use cases could result in issues in a small proportion of cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Merge tag 'mfd-for-linus-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6
Pull MFD fixes from Samuel Ortiz:
"We have 3 build fixes, a OMAP USB host PHY reset fix and the twl6040
conversion to an i2c driver. The latter may not sound like a fix but
the twl6040 MFD driver won't probe without it, triggering an OMAP4
audio regression."
* tag 'mfd-for-linus-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6:
mfd: Fix modular builds of rc5t583 regulator support
mfd: Fix asic3_gpio_to_irq
ARM: OMAP3: USB: Fix the EHCI ULPI PHY reset issue
mfd: Convert twl6040 to i2c driver, and separate it from twl core
mfd : Fix dbx500 compilation error
Drop some struct members and definitions that became obsolete during
the refactorization of the driver.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes for a few regressions of HD-audio, originated partly from 3.4
and partly 3.3.
The fixes for ThinkPad docking-station are for 3.3 kernels, thus they
are based on 3.3 then merged back to 3.4, so that they can be merged
to stable tree cleanly. The non-trivial merge conflicts are because
of this action.
In addition, a copule of trivial fixes for documentation and a long-
statnding issue in the listing of built-in sound driver at boot time.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Fixes for a few regressions of HD-audio, originated partly from 3.4
and partly 3.3.
The fixes for ThinkPad docking-station are for 3.3 kernels, thus they
are based on 3.3 then merged back to 3.4, so that they can be merged
to stable tree cleanly. The non-trivial merge conflicts are because
of this action.
In addition, a couple of trivial fixes for documentation and a long-
standing issue in the listing of built-in sound driver at boot time."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/conexant - Set up the missing docking-station pins
ALSA: hda/conexant - Don't set HP pin-control bit unconditionally
ALSA: workaround: change the timing of alsa_sound_last_init()
ALSA: hda/sigmatel - Fix inverted mute LED
ALSA: hda/realtek - Fix regression on Quanta/Gericom KN1
ALSA: fix core/vmaster.c kernel-doc warning
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>
According to the reporter, external mic starts to work if the
laptop-dmic model is used. According to BIOS pin config, all
pins are consistent with the alc269vb_laptop_dmic fixup, except
for the external mic, which is not present.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/950490
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge fixes for Thinkpad docking-station regressions for 3.3 kernels
back to 3.4. These were committed in that branch to make the stable
merging easier.
Conflicts:
sound/pci/hda/patch_conexant.c
ThinkPad 410,420,510,520 and X201 with cx50585 & co chips have the
docking-station ports, but BIOS doesn't initialize for these pins.
Thus, like the former X200, we need to set up the pins manually in the
driver.
The odd part is that the same PCI SSID is used for X200 and T400, thus
we need to prepare individual fixup tables for cx5051 and others.
Bugzilla entries:
https://bugzilla.redhat.com/show_bug.cgi?id=808559https://bugzilla.redhat.com/show_bug.cgi?id=806217https://bugzilla.redhat.com/show_bug.cgi?id=810697
Reported-by: Josh Boyer <jwboyer@redhat.com>
Reported-by: Jens Taprogge <jens.taprogge@taprogge.org>
Tested-by: Jens Taprogge <jens.taprogge@taprogge.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some output pins on Conexant chips have no HP control bit, but the
auto-parser initializes these pins unconditionally with PIN_HP.
Check the pin-capability and avoid the HP bit if not supported.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Springbank module can support a range of sample rates, selected at
runtime via GPIO configuration. Allow these to be configured at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current alsa_sound_last_init() was called as __initcall().
So, on current ALSA, only devices that had been properly
registered at this point were shown.
So, it will show "No soundcards found" if driver requests
probe deferment. it's often misleading.
This patch delays the timing of alsa_sound_last_init()
as workaround.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviwed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While refactoring the mute-LED handling for HP laptops, I messed up
the polarity check in a wrong way. The red (or the mute-LED if any)
should appear in the muted state, corresponding to GPIO on.
Reported-by: Mikko Vinni <mmvinni@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Through the transition to the auto-parser, the support for
Quanta/Gericom KN1 got broken. There are two problems behind it:
- This machine doesn't like the default COEF setup for ALC260 we take
now as default
- BIOS doesn't set the pins correctly at all; especially the machine
uses only the pin 0x0f for both headphone and speaker
This patch adds the fixup as a workaround for these issues.
Reported-and-tested-by: Uros Vampl <mobile.leecher@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for ASoC DSP support.
Add a DAPM API call to determine whether a DAPM audio path is valid between
source and sink widgets. This also takes into account all kcontrol mux and mixer
settings in between the source and sink widgets to validate the audio path.
This will be used by the DSP core to determine the runtime DAI mappings
between FE and BE DAIs in order to run PCM operations.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Connect the WM1250-EV1 baseband simulator on Littlemill systems up to
the CODEC AIF2 using the new CODEC<->CODEC link support, allowing a wider
range of use cases to be represented.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Leading up to the ->device_prep_slave_sg change in
185ecb5f4f 'dmaengine: add context
parameter to prep_slave_sg and prep_dma_cyclic' a generic wrapper was
added in place to guard against the API change, though the fsi driver
wasn't updated in the process (presumably its dmaengine support hadn't
been merged yet at the time). This trivially switches over to the new
wrapper and gets it building again.
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix kernel-doc warning in sound/core/vmaster.c:
Warning(sound/core/vmaster.c:429): No description found for parameter 'private_data'
Signed-off-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is nothing audio-specific about the rcode_string() helper, so move
it from snd-firewire-lib into firewire-core to allow other code to use it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de> (fixed sound/firewire/cmp.c)
Since AIF3 shares clock signals with other audio interfaces in order to
ensure it doesn't drive undesirable clocks we need to tristate it. Rather
than forcing the machine driver to do so have the driver do this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch converts multiple if conditions in to single if with "&&"s.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently we increment the number of RTD's per card during the DAI link
bind. This can cause an incorrect RTD count when we cannot find a component
and defer the probe (and hence perform the DAI link bind for the card again).
Fix the count so that it is cleared before every card registration
and bind attempt.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current DA7210 driver does support PLL mode fully. It uses fixed
value of input master clock and PLL mode is enabled and disabled based
on the sampling frequency being used for playback or recording. It also
doesn't support Sample Rate Measurement feature of DA7210 hardware.
This patch adds full support for PLL and SRM. Basically following three
modes of operation are possible for DA7210 hardware,
(1) I2S SLAVE mode with PLL bypassed
(2) I2S SLAVE mode with PLL enabled
(3) I2S Master mode with PLL enabled
This patch adds support for all three modes. Also, in case of SLAVE mode
with PLL, it supports SRM (Sample Rate Measurement) feature of the chip.
Actually this patch was submitted earlier and received some review
comments, but after that the driver got update by other patches. Because
of that, I am considering this as new patch and not versioning it based
of previous patches. This version tries to take care of all review
comments received for earlier submissions.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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ASoC: Merge tag 'v3.4-rc3' into for-3.5
Linux 3.4-rc3 contains a bunch of Tegra changes which are conflicting
annoyingly with the new development that's going on for Tegra so merge
it up to resolve those conflicts.
Conflicts:
sound/soc/soc-core.c
sound/soc/tegra/tegra_i2s.c
sound/soc/tegra/tegra_spdif.c
Fix the following build warning:
sound/soc/soc-dapm.c: In function 'snd_soc_dai_link_event':
sound/soc/soc-dapm.c:2913: warning: format '%lx' expects type 'long unsigned int', but argument 3 has type 'u64'
'%llx' should be used with 'u64' type.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than having the user half start a stream but avoid any DMA to
trigger data flow on links which don't pass through the CPU create a
DAPM route between the two DAI widgets using a hw_params configuration
provided by the machine driver with the new 'params' member of the
dai_link struct. If no configuration is provided in the dai_link then
use the old style even for CODEC<->CODEC links to avoid breaking
systems.
This greatly simplifies the userspace usage of such links, making them
as simple as analogue connections with the stream configuration being
completely transparent to them.
This is achieved by defining a new dai_link widget type which is created
when CODECs are linked and triggering the configuration of the link via
the normal PCM operations from there. It is expected that the bias
level callbacks will be used for clock configuration.
Currently only the DAI format, rate and channel count can be configured
and currently the only DAI operations which can be called are hw_params
and digital_mute(). This corresponds well to the majority of CODEC
drivers which only use other callbacks for constraint setting but there
is obviously much room for extension here. We can't simply call
hw_params() on startup as things like the system clocking configuration
may change at runtime and in future it will be desirable to offer some
configurability of the link parameters.
At present we are also restricted to a single DAPM link for the entire
DAI. Once we have better support for channel mapping it would also be
desirable to extend this feature so that we can propagate per-channel
power state over the link.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
We should always have a CODEC already there when registering a CODEC DAI
and for CODEC<->CODEC links a dai_link will have two CODECs so it's much
simpler to do things at registration time.
This results in a slight change in the error handling for failed CODEC
DAI registrations but practically speaking these are never supposed to
fail so there shouldn't be much issue. The change is that we don't fail
the overall CODEC registration if the DAI registration fails; this seems
more robust anyway as we may not need to use a given DAI in a particular
system.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Shawn Guo
Brian Austin
Mark Hills
Takashi Iwai
Bo Shen
Linus Torvalds
Mark Brown
Ola Lilja
Guennadi Liakhovetski
David Henningsson
Andrew Lunn
Axel Lin
Andre Schramm
Peter Ujfalusi
guoyh
Clemens Ladisch
Oleg Matcovschi
Ashish Chavan
Stephen Warren
Heiko Stübner
Liam Girdwood
Eric Bénard
Richard Zhao
Richard Zhao
Fabio Estevam
Lars-Peter Clausen
Kyung-Kwee Ryu
Daniel Mack
Andrew Morton
Felix Homann
Kristoffer KARLSSON
Jesper Juhl
Kuninori Morimoto
Masanari Iida
Paul Mundt
Randy Dunlap
Fabio Estevam