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This merges the changes for converting to new PM ops for platform
and some other drivers.
Also move some header files to local places from the public
include/sound.
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ASoC: Updates for 3.6
This has been a pretty quiet release - very little activity in framework
terms, mostly just a few new drivers and updates:
- Added the ability to add and remove DAPM paths dynamically, mostly for
reparenting on clock changes.
- New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
reference platform and ttc-dkp.
- New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
- New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI
Isabelle and Wolfson Microelectronics WM5102 and WM5110
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Generic updates for sound 3.6
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Add a DECLARE_TLV_DB_RANGE() macro so that dB range information
can be specified without having to count the items manually for
TLV_DB_RANGE_HEAD().
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add the DECLARE_TLV_CONTAINER() macro to allow having static
TLVs containing more than one item.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add helper macros with a little bit of preprocessor magic to
automatically compute the length of a TLV item. This lets us avoid
having to compute this by hand, and will allow to use items that do
not use a fixed length.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since we're now relying on DAPM for things like enabling clocks when we
reparent the clocks for widgets we need to either use conditional routes
(which are expensive) or remove routes at runtime. Add a route removal
API to support this use case.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
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They aren't modified by the core so the drivers can declare them const.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is a bit clean up of public sound header directory.
Some header files in include/sound aren't really necessary to be
located there but can be moved to their local directories gracefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Straightforward conversion to the new pm_ops from the legacy
suspend/resume ops.
Since we change vx222, vx_core and vxpocket have to be converted,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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()
...
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The code handles this fine already, we just need new macros in the header
for drivers to create the controls.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
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This patch implements the spdif IN driver for ST peripheral
Signed-off-by: Vipin Kumar <vipin.kumar@st.com>
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch add support for the SPEAr ASoC pcm layer in ASoC
framework. The pcm layer uses common snd_dmaengine framework.
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch add support for synopsys I2S controller as per the ASoC
framework.
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Currently the sound dmaengine pcm helper functions implement the pcm_pointer
callback by trying to count the number of elapsed periods. This is done by
advancing the stream position in the dmaengine callback by one period.
Unfortunately there is no guarantee that the callback will be called for each
elapsed period. It may be possible that under high system load it is only called
once for multiple elapsed periods. This patch addresses the issue by
implementing support for querying the current stream position directly from the
dmaengine driver. Since not all dmaengine drivers support reporting the stream
position yet the old period counting implementation is kept for now.
Furthermore the new mechanism allows to report the stream position with a
sub-period granularity, given that the dmaengine driver supports this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Currently the sound dmaengine pcm helper functions implement the pcm_pointer
callback by trying to count the number of elapsed periods. This is done by
advancing the stream position in the dmaengine callback by one period.
Unfortunately there is no guarantee that the callback will be called for each
elapsed period. It may be possible that under high system load it is only called
once for multiple elapsed periods. This patch renames the current implementation
and documents its shortcomings and that it should not be used anymore in new
drivers.
The next patch will introduce a new snd_dmaengine_pcm_pointer which will be
implemented based on querying the current stream position from the dma device.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by Vinod Koul <vinod.koul@linux.intel.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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There's a dependency but no #include.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is essentially the reverse of snd_pcm_rate_to_rate_bit().
This is generally useful as the Compress API uses the rate bit
directly and it helps to be able to map back to the actual sample
rate.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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Control type added for cases where a specific range of values
within a register are required for control.
Added convenience macros:
SOC_SINGLE_RANGE
SOC_SINGLE_RANGE_TLV
Added accessor implementations:
snd_soc_info_volsw_range
snd_soc_put_volsw_range
snd_soc_get_volsw_range
Signed-off-by: Michal Hajduk <Michal.Hajduk@diasemi.com>
Signed-off-by: Adam Thomson <Adam.Thomson@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Prior to this patch, the CPU side of a DAI link was specified using a
single name. Often, this was the result of calling dev_name() on the
device providing the DAI, but in the case of a CPU DAI driver that
provided multiple DAIs, it needed to mix together both the device name
and some device-relative name, in order to form a single globally unique
name.
However, the CODEC side of the DAI link was specified using separate
fields for device (name or OF node) and device-relative DAI name.
This patch allows the CPU side of a DAI link to be specified in the same
way as the CODEC side, separating concepts of device and device-relative
DAI name.
I believe this will be important in multi-codec and/or dynamic PCM
scenarios, where a single CPU driver provides multiple DAIs, while also
booting using device tree, with accompanying desire not to hard-code the
CPU side device's name into the original .cpu_dai_name field.
Ideally, both the CPU DAI and CODEC DAI loops in soc_bind_dai_link()
would now be identical. However, two things prevent that at present:
1) The need to save rtd->codec for the CODEC side, which means we have
to search for the CODEC explicitly, and not just the CODEC side DAI.
2) Since we know the CODEC side DAI is part of a codec, and not just
a standalone DAI, it's slightly more efficient to convert .codec_name/
.codec_of_node into a codec first, and then compare each DAI's .codec
field, since this avoids strcmp() on each DAI's CODEC's name within
the loop.
However, the two loops are essentially semantically equivalent.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Adds a supply-widget variant for connection to the clock-framework.
This widget-type corresponds to the variant for regulators.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Adds a function getting the stream-name as a string for
a specific stream.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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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.
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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>
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There's no space for the sign bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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
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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>
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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>
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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>
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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>
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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>
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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>
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There are no users any more and new drivers should be using supply widgets
which fully replace it anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Zeng Zhaoming <zengzm.kernel@gmail.com>
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This change adds the logic to support using the jack detect mechanism built
in to the codec to detect both when a jack was inserted and what type of
jack is present.
This change also supports the use of an external mechanism for headphone
detection. If this mechanism exists, when the max98095_jack_detect function
is called, the hp_jack is simply passed NULL.
This change supports both simple headphones, powered headphones, microphones
and headsets with both headphones and a mic.
Signed-off-by: Rhyland Klein <rklein@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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In version 3.4 the driver core acquired probe deferral which is a core way
of doing essentially the same thing as ASoC has been doing since forever
to make sure that all the devices needed to make up the card are present
without needing open coding in the subsystem.
Make basic use of this probe deferral mechanism for the cards, removing the
need to handle partially instantiated cards. We should be able to remove
even more code than this, though some of the checks we're currently doing
should stay since they're about things like suppressing unneeded DAPM runs
rather than deferring probes.
In order to avoid robustness issues with our teardown paths (which do need
quite a bit of TLC) add a check for aux_devs prior to attempting to set
things up, this means that we've got a reasonable idea that everything will
be there before we start. As with the removal of partial instantiation
support more work will be needed to make this work neatly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
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Currently operations on jack reporting take the CODEC mutex both to protect
the current jack status and also to protect the DAPM run which is triggered
on status updates. Since the addition of a DAPM-specific lock we no longer
need to worry about locking DAPM as it has its own finer grained lock so
create a per jack lock to take care of the jack status.
This is both cleaner where the jack isn't specifically associated with a
CODEC and clearer as it's much more obvious what the lock is protecting.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Takashi Iwai
Clemens Ladisch
Mark Brown
Linus Torvalds
Vipin Kumar
Rajeev Kumar
Lars-Peter Clausen
Dimitris Papastamos
Hans de Goede
Adam Thomson
Stephen Warren
Ola Lilja
Kuninori Morimoto
Brian Austin
Liam Girdwood
Kristoffer KARLSSON
Ricardo Neri
Fengguang Wu
Rhyland Klein