| Commit message (Collapse) | Author | Age |
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Now that we have regular register mapped controls we should be splitting
the control sets for ADSP1 and ADSP2 as the register maps are not
identical. Do that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Ensure that the outputs are fully enabled before we begin passing audio
through them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/misc into asoc-arizona
ASoC/extcon: arizona: Fix interaction between HPDET and headphone outputs
This patch series covers both ASoC and extcon subsystems and fixes an
interaction between the HPDET function and the headphone outputs - we
really shouldn't run HPDET while the headphone is active. The first
patch is a refactoring to make the extcon side easier.
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Remove duplicated include.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Since ASoC does not yet really have the framework features needed to
support propagating sample rates through the device well yet implement
basic support for the ISRCs equivalent to that we currently have for the
ASRCs. The user can opt for 8kHz or 16kHz as the rate for the DSP blocks
in addition to the main audio rate, these being the primary use cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Ensure that the device state does not diverge from the state we have set
in the register map in order to make the behaviour clearer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Help with debuggability.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Some system designs have been identified which repurpose portions of the
speaker driver circuits for other functions which will require that they
not be managed using DAPM. Prepare for this by factoring out the creation
of the speaker widgets into the core driver, the widgets will be replaced
by dummy ones when the additional functions are enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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I2S requires stereo clocking even for mono data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Allow users to keep on specifying their output frequency when disabling
the reference clock.
Reported-by: Kyung Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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When live configuring a FLL configuration with no synchroniser disable the
synchroniser in case the previous configuration used one.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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For optimal performance the FLL loop gain should be adjusted depending on
the frequency of the input clock for the loop.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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If we are using a high freqency SYNCCLK then increasing the bandwidth of
the synchroniser improves performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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If there is only one clock active the FLL should use REFCLK rather than
SYNCCLK as the clock to synchronise with since REFCLK is always required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Enabling the FLL when REFCLK is being configured is not what the user
would expect and can cause issues if SYNCCLK has no specified frequency.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Since we are automatically managing the mutes we may as well also manage
the volume update bits, disabling volume updates while none of the inputs
are active. Since we are doing this we may as well allow the volumes to
ramp together so only enable volume updates once at the end of power up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch allows the REFCLK to be set directly allowing much greater
flexibility in how the FLLs are configured.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Previously updates that only changes FLL source would be missed, this
patch corrects this. We also ensures that both REFCLK and SYNCCLK
frequency changes are considered, in preparation for future updates.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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In preparation for additional features on the FLL this patch factors out
the code for enabling an FLL into a seperate function.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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In preparation for additional features on the FLL this patch factors out
the code for disabling an FLL into a seperate function.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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In preparation for additional features on the FLL this patch factors out
the code which checks if an FLL is currently enabled into a seperate
function.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch caches the current SYNCCLK settings in the arizona_fll struct
and uses these to simplify the code which determines which source should
be used for the REFCLK and SYNCCLK inputs.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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In preparation for additional features on the FLL this patch moves the
code selecting the REFCLK source based on the 32kHz clock into the FLL
initialisation function.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Adds a driver for Asahi Kasei's AK5386 Single-ended 24-Bit 192kHz
delta-sigma ADC. The device has no control port interface but an
optional RESET/PDN GPIO pin.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Move the enabling of the TX diode to hw_params() and disable it again in
hw_free(). This way, the diode is only switched on as long as it needs
to be.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Firmwares may provide some firmware wide configuration regions which can
be configured by the coefficient files using the firmware ID as the
algorithm ID, include these in the algorithm list.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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For future work to have specific handling for some firmwares.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Older .bin files report the global coefficients as absolute address writes
to zero; maintain compatibility with them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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ADAU1373_BCLKDIV_SOURCE is defined as BIT(5) which uses UL constants. On
amd64 the result of the ones complement operator is then truncated to
unsigned int according to the prototype of snd_soc_update_bits(). I think
gcc is correctly warning that the upper 32 bits are lost.
sound/soc/codecs/adau1373.c: In function 'adau1373_hw_params':
sound/soc/codecs/adau1373.c:940:3: warning: large integer implicitly truncated to unsigned type [-Woverflow]
gcc version 4.6.3
Add 2 more BCLKDIV mask macros as explained by Lars:
The BCLKDIV has three fields. The bitclock divider (bit 0-1), the samplerate
(bit 2-4) and the source select (bit 5). Here we want to update the bitclock
divider field and the samplerate field. When I wrote the code I was lazy and
used ~ADAU1373_BCLKDIV_SOURCE as the mask, which for this register is
functionally equivalent to ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK.
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Commit 88a8516a2128 (ALSA: usbaudio: implement USB autosuspend)
introduced autopm for all USB audio/MIDI devices. However, many MIDI
devices, such as synthesizers, do not merely transmit MIDI messages but
use their MIDI inputs to control other functions. With autopm, these
devices would get powered down as soon as the last MIDI port device is
closed on the host.
Even some plain MIDI interfaces could get broken: they automatically
send Active Sensing messages while powered up, but as soon as these
messages cease, the receiving device would interpret this as an
accidental disconnection.
Commit f5f165418cab (ALSA: usb-audio: Fix missing autopm for MIDI input)
introduced another regression: some devices (e.g. the Roland GAIA SH-01)
are self-powered but do a reset whenever the USB interface's power state
changes.
To work around all this, just disable autopm for all USB MIDI devices.
Reported-by: Laurens Holst
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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With this patch, a TRRS headset mic cannot be successfully detected
on the Asus X101CH, and we can also distinguish between headphone
and headset automatically.
Buglink: https://bugs.launchpad.net/bugs/1169138
Co-authored-by: Kailang <kailang@realtek.com>
Tested-by: Luis Henriques <luis.henriques@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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On some machines, there is a headset jack that can support both
headphone, headsets (of both CTIA and OMTP type) and mic-in.
On other machines, the headset jack supports headphone, headsets
(both CTIA and OMTP), but not mic-in.
This patch implements that functionality as different capture sources.
Buglink: https://bugs.launchpad.net/bugs/1169143
Tested-by: David Chen <david.chen@canonical.com>
Co-authored-by: Kailang <kailang@realtek.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When recording at 176.2KHz or 192Khz, the device adds a 32-bit length
header to the capture packets, which obviously needs to be ignored for
recording to work properly.
Userspace expected: L0 L1 L2 R0 R1 R2
...but actually got: R2 L0 L1 L2 R0 R1
Also, the last byte of the length header being interpreted as L0 of
the first sample caused spikes every 0.5ms, resulting in a loud 16KHz
tone (about the highest 'B' on a piano) being present throughout
captures.
Tested at all sample rates on an E-Mu 0404USB, and tested for
regressions on a generic USB headset.
Signed-off-by: Calvin Owens <jcalvinowens@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Now that we have a flag for headphone mics, we can use that flag
in the jack creation instead of creating the jack manually.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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I never liked that we move our speaker and hp pins to line out
if there are not any line outs; but now that we do,
add some convenience functions to find hp and speaker pins even
if they have been moved.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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