| Commit message (Collapse) | Author | Age |
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Both the mask and mode settings are wrong in current code.
According to the datasheet:
S2PCFG0 (0x0A)
BIT[3:1] DATA_FORMAT
serial interface protocol format:
000: left Justified
001: I2S (default)
010: right justified
100: PCM no delay
101: PCM delay
111: DSP
Thus fixes the defines for LEFT_J_DATA_FORMAT, I2S_DATA_FORMAT, and
RIGHT_J_DATA_FORMAT.
Also adds define for DATA_FORMAT_MSK.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3.
From Eric Nelson:
"We also found that for the microphones we have here (commodity PC boom mics) a
default value of 2 for the gain gives the best results."
So change the default microphone gain as well.
Signed-off-by: Eric Nelson <eric.nelson@boundarydevices.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch fixes some code that implements a work-around to a hardware bug in
the ac97 controller on the pxa27x. A bug in the controller's warm reset
functionality requires that the mfp used by the controller as the AC97_nRESET
line be temporarily reconfigured as a generic output gpio (AF0) and manually
held high for the duration of the warm reset cycle. This is what was done in
the original code, but it was broken long ago by commit fb1bf8cd
([ARM] pxa: introduce processor specific pxa27x_assert_ac97reset())
which changed the mfp to a GPIO input instead of a high output.
The fix requires the ac97 controller to obtain the gpio via gpio_request_one(),
with arguments that configure the gpio as an output initially driven high.
Tested on a palm treo 680 machine. Reportedly, this broken code only prevents a
warm reset on hardware that lacks a pull-up on the line, which appears to be the
case for me.
Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Signed-off-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Cold reset on the pxa27x currently fails and
pxa2xx_ac97_try_cold_reset: cold reset timeout (GSR=0x44)
appears in the kernel log. Through trial-and-error (the pxa270 developer's
manual is mostly incoherent on the topic of ac97 reset), I got cold reset to
complete by setting the WARM_RST bit in the GCR register (and later noticed that
pxa3xx does this for cold reset as well). Also, a timeout loop is needed to
wait for the reset to complete.
Tested on a palm treo 680 machine.
Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Acked-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Update lm49453_reg_defs values as per LM49453 HW revision-B
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Add adc, mic, sidetone volume ranges and appropriately added the controls.
Fix the DAC HP/EP/LS/LO/HA maximum gain values.
Signed-off-by: MR Swami Reddy <mr.swami.reddy@ti.com>
Tested-by: Vinod Koul <vinod.koul@intel.com>
--
sound/soc/codecs/lm49453.c | 43 ++++++++++++++++++++++++-------------------
1 files changed, 24 insertions(+), 19 deletions(-)
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The mode variable is either 0 or 1.
To update mode setting, the mask should be BIT(0) rather than BIT(1).
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Omair M. Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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In the case of no-match, return -EINVAL instead of 0.
Since we assign i to ret in the for loop, ret always less than
ARRAY_SIZE(clk_map_table). Thus remove the boundary checking for ret.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The driver had the property check for 'cirrus,amutec_eq_bmutec' the
wrong way around. That happens if you misspell the property in the
bindings during tests.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Make the flag in the pdata of type bool to fix a sparse warning.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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When probing aux_dev, initializing is as below:
device_initialize()
device_add()
So when remove aux_dev, we need do as below:
device_del()
device_put()
Otherwise, the rtd_release() will not be called.
So here using device_unregister() to replace device_del(),
like the action in soc_remove_link_dais().
Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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After called device_initialize(), even device_add() returns
error, we still need use the put_device() to release the reference
to call rtd_release(), which will do the free() action.
Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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commit 9bde4f0b1c (ASoC: core: Fix SOC_DOUBLE_RANGE() macros) introduced
the following build warning:
sound/soc/soc-core.c:2999:6: warning: unused variable 'ret' [-Wunused-variable]
Remove the unused 'ret' variable.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Although we've had macros defining double _RANGE controls for a while now
they've not actually been backed up properly by the implementation, it's
treated everything as mono. Fix that by implementing the handling in the
stereo controls, ensuring that the mono controls don't mistakenly get
treated as stereo.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
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When front-end PCM session is in paused state, back-end
PCM session will be put in paused state as well if given
front-end PCM session is the only client of given back-end.
Then, application closes front-end PCM session, DPCM
framework will not allow back-end enters HW_FREE state
so back-end will never get shutdown completely.
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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These are not supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The FLL source constants were numbered as a simple enumeration but were
being used in the code as direct values to be written to the registers.
Renumber the constants to reflect the usage.
Reported-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */
Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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The mute LED is in this case connected to the Mic1 VREF.
The machine also exposes the following string in BIOS:
"HP_Mute_LED_0_A", so if more machines are coming, it probably
makes sense to try to do something more generic, like for the
IDT codec.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1096789
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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vortex_wt_setdsout performs bit-negation on the bit position (wt&0x1f)
rather than on the resulting bitmask. This code is never actually
invoked (vortex_wt_setdsout is always called with en=1), so this does
not currently cause any problem, and this patch is simply cleanup.
Signed-off-by: Nickolai Zeldovich <nickolai@csail.mit.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Added a fix for hardware dependence bug where a sound card failure
should not result in reading beyond array memory index.
Signed-off-by: Asim Kadav <kadav@cs.wisc.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch is based on 3.8-rc1. It fixes two things:
1) A kernel panic caused by incorrect allocation of a u8 variable
"bootresponse".
2) A noisy dmesg (urb status -32) caused by broken pipe to an
invalid midi endpoint.
It is also a little cleaner because there is no need for a new
QUIRK_MIDI type as suggested by kernel developers, since the device
follows exactly the MIDIMAN protocol.
Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Support the Creative BT-D1 Bluetooth USB audio device. Before this
patch, Linux had trouble finding the correct USB descriptors and bailed
out with these messages:
no or invalid class specific endpoint descriptor
Now it still prints these messages on hotplug:
snd-usb-audio: probe of ...:1.0 failed with error -5
snd-usb-audio: probe of ...:1.2 failed with error -5
snd-usb-audio: probe of ...:1.3 failed with error -5
But the device works correctly, including the HID support.
The patch is diff'ed against 3.8-rc1 but should apply to older kernels
as well.
Signed-off-by: Alexander Schremmer <alex@alexanderweb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The vmaster hook sends 1 for enabled/unmuted and 0 for disabled/muted,
but "Mute-LED Mode" being "On" refers to the LED being on, not the
volume being on.
Therefore "On" and "Off" should be switched.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This update contains overall only driver-specific fixes. Slightly
large LOC are seen in usb-audio driver for a couple of new device
quirks and cs42l71 ASoC driver for enhanced features. The others are
a few small (regression) fixes HD-audio, and yet other small / trival
ASoC fixes."
* tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:
ALSA: HDA: Fix sound resume hang
ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins
ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixup
ALSA: hda - Set codec->single_adc_amp flag for Realtek codecs
ASoC: atmel-ssc: change disable to disable in dts node
ASoC: Prevent pop_wait overwrite
ALSA: usb-audio: ignore-quirk for HP Wireless Audio
ALSA: hda - Always turn on pins for HDMI/DP
ALSA: hda - Fix pin configuration of HP Pavilion dv7
ASoC: core: Fix splitting of log messages
ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT
ASoC: cs42l73: Add DAPM events for power down.
ASoC: cs42l73: Add DMIC's as DAPM inputs.
ASoC: sigmadsp: Fix endianness conversion issue
ASoC: tpa6130a2: Use devm_* APIs
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This patch is the result of a lot of trial and error, since there are no specs
available for the device.
Full duplex support is provided, i.e. playback and recording in stereo.
The format is hardcoded at 48000Hz @ 24 bit, which is the maximum that the
device supports. Also, MIDI in and MIDI out both work.
Users will notice that the S/PDIF light also flashes when playback or recording
is active. I believe this means that S/PDIF input/output is simultaneously
activated with the analogue i/o during use.
But this particular functionality remains untested.
Note that this particular version of the patch is so far untested on the
physical hardware because I have not compiled a full kernel with the changes.
However, extensive testing has been done by many users of the hardware
who believe other versions of my patch have worked since circa 2009.
[Modified to make a function static by tiwai]
Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Resuming a switcheroo'd HDA controller hangs since the completion
is one-shot (thus works the first time). Fix by using completions
that explictly need rearming, so remain fired before.
Signed-off-by: Daniel J Blueman <daniel@quora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Haswell HDMI codec pins may report invalid connection list entries, which
will cause failure to play audio via HDMI or Display Port.
So this patch adds fixup for Haswell to workaround this hardware issue:
enable DP1.2 mode and override the pins' connection list entries with proper
value.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Xingchao Wang <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The workaround to force VREF50 for dallas/hp model with ALC861VD
was introduced in commit 8fdcb6fe4204bdb4c6991652717ab5063751414e,
but it contained wrong pincap override bits.
This patch fixes to exclude VREF80 pincap bit correctly.
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It turned out that Realtek codecs (ALC260, etc) with input amps in
audio-input widgets don't handle the multiple individual input amps.
Thus we need to set codec->single_adc_amp flag for them in general.
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: More updates for v3.8
Nothing terribly exciting here, just small localised changes.
As well as fixes there are a couple of Cirrus changes and one devm_
change which were in prior to the merge window but got missed from the
original pull to Takashi.
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Converted to use devm_gpio_request and devm_regulator_get APIs.
These are device managed and make error handling and cleanup
a bit simpler.
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Don't wrap log messages over multiple lines, it makes them hard to grep
for.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Since VSP only has one power bit, only provide one DAPM widget.
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Add power down delays between setting PDN and MCLKDIS for spk amp, spklo amp, and ear amp.
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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pop_wait is used to determine if a deferred playback close
needs to be cancelled when the a PCM is open or if after
the power-down delay expires it needs to run. pop_wait is
associated with the CODEC DAI, so the CODEC DAI must be
unique. This holds true for most CODECs, except for the
dummy CODEC and its DAI.
In DAI links with non-unique dummy CODECs (e.g. front-ends),
pop_wait can be overwritten by another DAI link using also a
dummy CODEC. Failure to cancel a deferred close can cause
mute due to the DAPM STOP event sent in the deferred work.
One scenario where pop_wait is overwritten and causing mute
is below (where hw:0,0 and hw:0,1 are two front-ends with
default pmdown_time = 5 secs):
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1
sleep 1
aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 &
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE
Since CODECs may not be unique, pop_wait is moved to the PCM
runtime structure. Creating separate dummy CODECs for each
DAI link can also solve the problem, but at this point it's
only pop_wait variable in the CODEC DAI that has negative
effects by not being unique.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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