ASoC: Add GPIO support for jack reporting interface
Add GPIO support to jack reporting framework in ASoC using gpiolib calls.
The gpio support exports two new functions: snd_soc_jack_add_gpios and
snd_soc_jack_free_gpios.
Client drivers using gpio feature must pass an array of jack_gpio pins
belonging to a specific jack to the snd_soc_jack_add_gpios function. The
framework will request the gpios, set the data direction and request irq.
The framework will update power status of related jack_pins when an event on
the gpio pins comes according to the reporting bits defined for each gpio.
All gpio resources allocated when adding jack_gpio pins can be released
using snd_soc_jack_free_gpios function.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Philipp Zabel [Tue, 3 Mar 2009 15:10:51 +0000 (16:10 +0100)]
ASoC: Use network mode with 2 slots for 16-bit stereo in pxa-ssp/Zylonite
For consistency with 24-bit and 32-bit modes, don't send 16-bit stereo
in one 32-bit transfer. Use 2 slots instead on Zylonite. It should result
in exactly the same behaviour.
Now it is possible to use 16-bit single slot transfers in pxa-ssp, which
are needed for Magician to get two frame clock pulses per sample
(one for each channel).
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Tested-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the UDA1380's interpolator or decimator are set to be clocked from
the WSPLL (which syncs to the WSI signal), the DAI link must be running
to change the interpolator/decimator registers (which include volume
controls and digital mute setting).
* Queue work in the alsa PCM_START .trigger to flush registers
as soon as the link is running. This replaces the .prepare
and .digital_mute callbacks.
* Use the SILENCE override instead of MTM for muting and remove
its alsa control to avoid confusion.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Daniel Mack [Sat, 28 Feb 2009 12:21:03 +0000 (13:21 +0100)]
ASoC: fix typo and removed unneeded switch case for cs4270
This removes a misspelled comment and got rid of superfluous switch
case.
Signed-off-by: Daniel Mack <daniel@caiaq.de> Acked-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Sat, 28 Feb 2009 21:14:20 +0000 (21:14 +0000)]
ASoC: Add SND_SOC_DAPM_PIN_SWITCH controls for exposing DAPM pins
On some systems it is desirable for control for DAPM pins to be provided
to user space. This is the case with things like GSM modems which are
controlled primarily from user space, for example. Provide a helper which
exposes the state of a DAPM pin to user space for use in cases like this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Sun, 22 Feb 2009 20:04:41 +0000 (20:04 +0000)]
ASoC: Only write back non-default registers when resuming WM8753
This will reduce the number of writes done on resume, allowing that to
complete faster (especially on systems with very slow I2C like the
current Samsung driver).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Wed, 18 Feb 2009 12:39:05 +0000 (14:39 +0200)]
ASoC: TWL4030: Add digital loopback support
This patch adds the digital loopback/bypass support for twl4030 codec.
The digital loopback will let the digimic0 (routed in the TX1 capture path
inside of TWL4030) data to be routed back to the RX2 playback path
(I2S stereo). It can also route the analog capture date routed through the
TX1 back to RX2.
Effectively the digital loopback is routing the audio from the TX1 capture path
to the RX2 playback path.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Mon, 16 Feb 2009 20:49:16 +0000 (20:49 +0000)]
ASoC: Refactor WM8731 device registration
Move the WM8731 driver to use a more standard device registration
scheme where the device can be registered independantly of the ASoC
probe.
As a transition measure push the current manual code for registering
the WM8731 into the individual machine driver probes. This allows
separate patches to update the relevant architecture files with less
risk of merge issues.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Mon, 16 Feb 2009 17:51:54 +0000 (17:51 +0000)]
ASoC: Actively manage MCLK for AT91SAM9G20-EK
We have software control of the MCLK for the WM8731 so save a bit of
power by actively managing it within the machine driver, enabling it
only while the codec is active.
Once ASoC supports multiple boards and doesn't require the soc-audio
device the initial clock setup should be pushed down into the arch/arm
code but for now this reduces merge issues.
Tested-by: Sedji Gaouaou <sedji.gaouaou@atmel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Kevin Hilman [Fri, 13 Feb 2009 19:36:37 +0000 (11:36 -0800)]
ASoC: Fix DaVinci module unload error
Fix for the error when the audio module is unloaded. On unregistering
the platform_device, platform_device_release will free the platform
data.If platform data is static the kernel panics when it is freed.
Instead use the platform device helper function to add data.
This change has been tested on DM644x EVM, DM644x SFFSDR and DM355 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com> Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Thu, 12 Feb 2009 19:33:19 +0000 (19:33 +0000)]
ASoC: Only register AC97 bus if it's not done already
ASoC supports both explicit codec drivers for AC97 devices and a simple
driver which uses the standard ALSA AC97 framework for codec support.
When used with the generic AC97 codec support that will provide the
ad hoc AC97 device for drivers like touchscreens to attach to so the
core shouldn't do so.
Reported-by: Manuel Lauss <mano@roarinelk.homelinux.net> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Timur Tabi [Tue, 3 Feb 2009 17:09:32 +0000 (11:09 -0600)]
ASoC: add additional controls to the CS4270 codec driver
Update the CS4270 codec driver to allow applications to use the mixer to
control Digital Loopback, Soft Ramp, Zero Cross, Popguard, and Auto-Mute.
Soft Ramp, Zero Cross, and Auto-Mute are disabled by the driver when it first
initializes the hardware, but these features either don't work or interfere
with normal ALSA behavior. However, they can now be re-enabled by an
application if desired.
Remove CONFIG_SND_SOC_CS4270_HWMUTE and always allow ASoC to control the mute
bits. The driver previously and erroneously assumed that these bits
control only external muting circuitry, but they also control internal
muting circuitry, so they should always be used.
Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: Update SDP3430 machine driver for snd_soc_card
This patch replaces "snd_soc_machine" structure by "snd_soc_card" in
SP3430 driver. This change is needed in SDP3430 driver to reflect
changes introduced by "ASoC: Rename snd_soc_card to snd_soc_machine" patch
(875065491fba8eb13219f16c36e79a6fb4e15c68).
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Mon, 9 Feb 2009 12:27:06 +0000 (14:27 +0200)]
ASoC: TLV320AIC3X: Fix volume ranges
This is a minor fix but helps to define dB ranges for volume controls.
Only DAC digital volume has full register value range from 0 to 127 but
ADC PGA gain and output stage volume controls don't.
For ADC PGA, maximum value is 119 and then it saturates to the same
gain value of 59.5 dB. For output stages, value 117 corresponds to -78.3 dB
and is muted for values 118 and above.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Philipp Zabel [Thu, 5 Feb 2009 16:48:21 +0000 (17:48 +0100)]
ASoC: pxa2xx-i2s: remove I2S pin setup
This removes the calls to pxa_gpio_mode from the pxa2xx-i2s driver.
Pin setup should be done during board init via pxa2xx_mfp_config
instead.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Acked-by: Eric Miao <eric.miao@marvell.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Philipp Zabel [Thu, 5 Feb 2009 16:48:20 +0000 (17:48 +0100)]
pxa/spitz: Setup I2S pins for pxa2xx-i2s
The spitz has a WM8750 codec connected as I2S slave but doesn't use the
PXA I2S system clock.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Acked-by: Eric Miao <eric.miao@marvell.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Philipp Zabel [Thu, 5 Feb 2009 16:48:19 +0000 (17:48 +0100)]
pxa/h5000: Setup I2S pins for pxa2xx-i2s
The iPAQ h5000 has an AK4535 codec connected as I2S slave,
PXA I2S providing SYSCLK.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Acked-by: Eric Miao <eric.miao@marvell.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Robert Jarzmik [Sat, 7 Feb 2009 13:01:58 +0000 (14:01 +0100)]
ASoC: Add initial support of Mitac mioa701 device SoC.
This machine driver enables sound functions on Mitac mio
a701 smartphone. Build upon ASoC v1, it handles :
- rear speaker
- front speaker
- microphone
- GSM
A global "Mio Mode" switch is not yet provided to cope with
audio path setup. As balance on audio chip line is no more
assured, an incorrect setup can produce a lot of heat and
even fry the battery behind the wm9713 and the speaker
amplifier.
It doesn't cope with :
- headset jack
- mio master mode
- master volume control
This driver is backported from ASoc v2, and amputated from
scenario setups and master volume control.
[Minor mods for terminology in comments -- broonie]
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Timur Tabi [Thu, 5 Feb 2009 23:56:02 +0000 (17:56 -0600)]
ASoC: optimize init sequence of Freescale MPC8610 sound drivers
In the Freescale MPC8610 sound drivers, relocate all code from the _prepare
functions into the corresponding _hw_params functions. These drivers assumed
that the sample size is known in the _prepare function and not in the
_hw_params function, but this is not true.
Move the code in fsl_dma_prepare() into fsl_dma_hw_param(). Create
fsl_ssi_hw_params() and move the code from fsl_ssi_prepare() into it.
Turn off snooping for DMA operations to/from I/O registers, since that's not
necessary.
Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mike Frysinger [Fri, 6 Feb 2009 10:12:35 +0000 (18:12 +0800)]
ASoC: Blackfin: drop unnecessary dma casts
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com> Signed-off-by: Bryan Wu <cooloney@kernel.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mike Frysinger [Fri, 6 Feb 2009 10:12:34 +0000 (18:12 +0800)]
ASoC: Blackfin: cleanup sport handling in ASoC Blackfin AC97 code
- make sport number handling more dynamic as not all
Blackfins have a linear sport map starting at 0
- indexes can be macroed away too
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com> Signed-off-by: Cliff Cai <cliff.cai@analog.com> Signed-off-by: Bryan Wu <cooloney@kernel.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Fri, 6 Feb 2009 10:01:05 +0000 (12:01 +0200)]
ASoC: WM8990: Fix kcontrol's private value use in put callback
Function wm899x_outpga_put_volsw_vu misuses the kcontrol's private value
by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it
as a pointer into struct soc_mixer_control after the commit 4eaa9819dc08d7bfd1065ce530e31b18a119dcaf.
This is very similar fix than fix to TLV320AIC3X codec made by
Eero Nurkkala <ext-eero.nurkkala@nokia.com>. This fix is compile tested
only.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com> Cc: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eero Nurkkala [Fri, 6 Feb 2009 10:01:04 +0000 (12:01 +0200)]
ASoC: TLV320AIC3X: Fix kcontrol's private value use in put callback
Function snd_soc_dapm_put_volsw_aic3x misuses the kcontrol's private value
by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it
as a pointer into struct soc_mixer_control after the commit 4eaa9819dc08d7bfd1065ce530e31b18a119dcaf.
This was causing arbitrary register writes when touching the controls
defined with SOC_DAPM_SINGLE_AIC3X.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Philipp Zabel [Tue, 3 Feb 2009 20:18:26 +0000 (21:18 +0100)]
ASoC: pxa-ssp: fix SSP port request
PXA2xx/3xx SSP ports start from 1, not 0. Thus, the probe function
requested the wrong SSP port. Correcting this unveiled another bug
where ssp_init tries to request the already-requested SSP port again.
So this patch replaces the ssp_init/exit calls with their internals
from mach-pxa/ssp.c, leaving out the redundant ssp_request and the
unneeded IRQ request. Effectively, that leaves us with not much more
than enabling/disabling the SSP clock.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Philipp Zabel [Tue, 3 Feb 2009 16:19:40 +0000 (17:19 +0100)]
ASoC: uda1380: split set_dai_fmt into _both, _playback and _capture variants
This patch splits set_dai_fmt into three variants (single interface,
dual interface playback only, dual interface capture only) so that
data input and output formats can be configured separately for dual
interface setups.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Tested-by: Vasily Khoruzhick <anarsoul@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: Don't unconditionally use the PLL in UDA1380
Without this fix driver switches to WSPLL in uda1380_pcm_prepare
even if SYSCLK was chosen (uda1380_pcm_prepare modifies UDA1380_CLK
register to disable R00_DAC_CLK before flushing reg cache)
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eero Nurkkala [Mon, 2 Feb 2009 12:20:46 +0000 (14:20 +0200)]
OMAP: ASoC: Fix spinlock misuse in omap-pcm.c
omap_pcm_trigger is called also in interrupt context so CPU flags must
be restored when returning.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update pandora board file for recent TWL4030 codec changes.
Also move output related snd_soc_dapm_nc_pin() calls to
omap3pandora_out_init(), where they belong.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Timur Tabi [Thu, 29 Jan 2009 20:28:37 +0000 (14:28 -0600)]
ASoC: fix initialization order of the CS4270 codec driver
ASoC codec drivers typically serve two masters: the I2C bus and ASoC itself.
When a codec driver registers with ASoC, a probe function is called. Most
codec drivers call ASoC first, and then register with the I2C bus in the ASoC
probe function.
However, in order to support multiple codecs on one board, it's easier if the
codec driver is probed via the I2C bus first. This is because the call to
i2c_add_driver() can result in the I2C probe function being called multiple
times - once for each codec. In the current design, the driver registers
once with ASoC, and in the ASoC probe function, it calls i2c_add_driver().
The results in the I2C probe function being called multiple times before the
driver can register with ASoC again.
The new design has the driver call i2c_add_driver() first. In the I2C probe
function, the driver registers with ASoC. This allows the ASoC probe function
to be called once per I2C device.
Also add code to check if the I2C probe function is called more than once,
since that is not supported with the current ASoC design.
Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 29 Jan 2009 12:57:50 +0000 (14:57 +0200)]
ASoC: TWL4030: Add analog loopback support
This patch adds the analog loopback/bypass support for twl4030 codec.
Details for the implementation:
It seams that the analog loopback needs the DAC powered on on the channel,
where the loopback is selected. The switch for the DACs has been moved from
the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be
powered while the audio playback is used or/and the loopback is enabled for
the channel.
The twl4030 codec powering has been reworked. Now the codec will be powered as
long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are
when the given change in the registers needs the codec power down/up cycle in
order to take effect. Otherwise the codec is on.
When the codec enters to STANDBY state and none of the loopback paths are
enabled, than the amplifiers, which are no in the DAPM path are forced to turn
off and the PLL is disabled. When playback/capture starts the disabled gains
are restored and the PLL is enabled.
When one of the loopback enabled in STANDBY mode, the disabled gains are
restored and the PLL is enabled also.
In short: the codec always goes to the lowest power state based on the
bias_level and the bypass_state.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: OMAP: Initialize XCCR and RCCR registers in McBSP DAI driver
This patch explicitly initializes McBSP Transmit Configuration
Control Register (XCCR) and Receive Configuration Control
Register (RCCR) to their reset values. Reset values are 26 ns
of DX delay and Transmit DMA disabled for XCCR register;
receive full cycle mode enabled and Receive DMA disabled for
RCCR register.
This patch requires a counterpart in OMAP McBSP driver before
to apply it. The required changes in McBSP were sent and approved
in linux-omap mailing list and patch is going upstream
(commit 3127f8f8595a064b3f1a1837fea2177902589ac3 from linux-omap-2.6
tree).
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
[ jarkko.nikula@nokia.com: Commit id for counterpart patch corrected ] Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Thu, 29 Jan 2009 13:08:20 +0000 (13:08 +0000)]
ASoC: Fix null string usage with WM8753 DAIs
The WM8753 driver multiplexes the DAI structures it exposes to the
outside world, leaving them uninitialised until the codec probes. Since
the DAI name is used during the registration and setup process provide a
dummy name.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Tue, 27 Jan 2009 09:29:42 +0000 (11:29 +0200)]
ASoC: TWL4030: Physical ADC and amplifier power switch change
Change the power switches for the physical ADC and for the
amplifiers for the analog capture path to map more closely
the actual path inside of the codec.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 23 Jan 2009 14:53:58 +0000 (14:53 +0000)]
ASoC: Fix spurious codec driver dependencies
Kbuild ignores dependency from things that are themselves selected so
ASoC machine drivers need to ensure that the control bus is being built.
This also avoids issues where multiple buses are supported by a given
codec.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 23 Jan 2009 11:40:26 +0000 (11:40 +0000)]
ASoC: Remove unneeded e7x0 inclusion of pxa-regs.h and hardware.h
pxa-regs.h and hardware.h are not intended for use directly in driver
code and references to them have been removed in other code - remove
them from the newly added e740 and e750 machine drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Timur Tabi [Fri, 23 Jan 2009 00:23:39 +0000 (18:23 -0600)]
ASoC: remove stand-alone mode support from CS4270 codec driver
The CS4270 supports stand-alone mode, where the codec is not connect to the
I2C or SPI buses. Instead, input voltages configure the codec at power-on.
The CS4270 ASoC device driver has partial support for this mode, but the
code was never tested, and partial support doesn't help anyone. It also made
the rest of the code more complicated than necessary.
[Removed redundant CS4270 dependency on I2C -- broonie]
Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Steve Sakoman [Wed, 21 Jan 2009 07:05:27 +0000 (23:05 -0800)]
ASoC: Complete Beagleboard support
Commit dc06102a0c8b5aa0dd7f9a40ce241e793c252a87 in the asoc tree
did not include the necessary Kconfig and Makefile changes. This patch
completes the support for Beagleboard
Signed-off-by: Steve Sakoman <steve@sakoman.com> Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Timur Tabi [Mon, 19 Jan 2009 17:57:46 +0000 (11:57 -0600)]
ASoC: Allow Freescale MPC8610 audio drivers to be compiled as modules
Change the Kconfig and Makefile options for Freescale MPC8610 audio drivers
so that they can be compiled as modules, and simplify the Kconfig choices
so that only the platform is selected.
Also fix the naming of the driver files to conform to ALSA standards.
Timur Tabi [Mon, 19 Jan 2009 23:14:24 +0000 (17:14 -0600)]
ASoC: fix registration of the SoC card in the Freescale MPC8610 drivers
The Freescale MPC8610 driver was defining two SOC card (snd_soc_card)
structures, partially initializing each one, but registering only one of
them with ASoC.
Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ian Molton [Sat, 17 Jan 2009 15:11:06 +0000 (15:11 +0000)]
ASoC: machine support for Toshiba e740 PDA
This patch provides suupport for the wm9705 AC97 codec on the Toshiba e740.
Note:
The e740 has a hard headphone switch that turns the speaker off and is not
software detectable or controlable. Also both headphone and speaker amps
share a common output enable.
Signed-off-by: Ian Molton <ian@mnementh.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
projects / linux-beck.git / log