Pull media fixes from Mauro Carvalho Chehab.
Trivial conflict due to new USB HID ID's being added next to each other
(Baanto vs Axentia).
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (44 commits)
[media] smia: Fix compile failures
[media] Fix VIDIOC_DQEVENT docbook entry
[media] s5p-fimc: Fix control creation function
[media] s5p-mfc: Fix checkpatch error in s5p_mfc_shm.h file
[media] s5p-mfc: Fix setting controls
[media] v4l/s5p-mfc: added image size align in VIDIOC_TRY_FMT
[media] v4l/s5p-mfc: corrected encoder v4l control definitions
[media] v4l: mem2mem_testdev: Fix race conditions in driver
[media] s5p-mfc: Bug fix of timestamp/timecode copy mechanism
[media] cxd2820r: Fix an incorrect modulation type bitmask
[media] em28xx: Show a warning if the board does not support remote controls
[media] em28xx: Add remote control support for Terratec's Cinergy HTC Stick HD
[media] USB: Staging: media: lirc: initialize spinlocks before usage
[media] Revert "[media] media: mx2_camera: Fix mbus format handling"
[media] bw-qcam: driver and pixfmt documentation fixes
[media] cx88: fix firmware load on big-endian systems
[media] cx18: support big-endian systems
[media] ivtv: fix support for big-endian systems
[media] tuner-core: return the frequency range of the correct tuner
[media] v4l2-dev.c: fix g_parm regression in determine_valid_ioctls()
...
Before this patch the owner field of the /dev/radio# device fops was set to
the snd-tea575x-tuner module itself. Meaning that the module which was using
it could be rmmod-ed while the device is open, and then BAD things happen.
I know, as I found out the hard way :)
Note that there is no need to also somehow increase the refcount of the
snd-tea575x-tuner module itself, since any drivers using it will have
symbolic references to it.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
ASoC: Last minute updates
These are all new code, they've been in -next already so should be OK
for merge this time round. I'd been planning to send a pull request
today after they'd had a bit of exposure there to make sure breakage
didn't propagate into your tree.
SupherH FSI2 can use special data transfer,
but it depends on CPU-FSI2 connection style.
We can use 16bit data stream mode if it was valid connection,
and it is required for 16bit data DMA transfer / SPDIF sound output.
We can use 24bit data transfer if it was invalid connection.
We can select connection type if CPU is SH7372,
and it is always valid connection if latest SuperH.
This patch adds new bus_option and fsi_bus_setup()
for supporting these feature.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Devices with many DAIs are becoming more and more common, and generally
the more modern devices have consistent register layouts between DAIs.
Rather than have drivers open code lookups based on the DAI ID or cause
uglification in UI by having register addresses for IDs provide a base
address field they can use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.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>
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>
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>
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>
Along with the IEC-60958 channel status word, CEA-861 Audio InfoFrames
are used in HDMI and DisplayPort to describe the parameters of the audio
stream. Hence, drivers for such devices may use these definitions to, for
instance, fill a CEA-861 data structure and pass it to a display driver
to configure an IP.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
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>
This patch uses simple-card driver instead of fsi-ak4642 on each board.
To select AK4642 driver, each boards select it on Kconfig.
This patch removes fsi-ak4642 driver which is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current ASoC requires card.c file to each platforms in order to
specifies its CPU and Codecs pair.
But the differences between these were only value/strings of setting.
In order to reduce duplicate driver, this patch adds generic/simple-card.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also remove two warnings when CONFIG_SND_DEBUG is not set:
sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’:
sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable]
sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable]
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Supports larger register maps, not using unsigned ints for the full 32
bit as we rely on checking for negative registers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle.
Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros.
Add single register macro : SOC_SINGLE_SX_TLV.
Use snd_soc_info_volsw for .info
Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double.
kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros.
The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
