Age | Commit message (Collapse) | Author |
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aux_dev is mainly used by the machine driver to specify analog devices,
which are registered as codecs. Making it more like a generic component
can help the machine driver to use it to specify any component with
topology info by name.
Details:
- Remove the stub 'rtd_aux' array from the soc card.
- Add a list 'aux_comp_list' to store the components of aux_devs.
And add a list head 'list_aux' to struct snd_soc_component, for adding
such components to the above list.
- Add a 'init' ops to a component for machine specific init.
soc_bind_aux_dev() will set it to be aux_dev's init. And it will be
called when probing the component.
- soc_bind_aux_dev() will also search components by name of an aux_dev,
since it may not be a codec.
- Move probing of aux_devs before checking new DAI links brought by
topology.
- Move removal of aux_devs later than removal of links. Because topology
of aux components may register DAIs and the DAI drivers will go with
removal of the aux components, we want soc_remove_link_dais() to remove
the DAIs at first.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Define API snd_soc_register_dai() to add a DAI dynamically and
create the DAI widgets. Topology can use this API to register DAIs
when probing a component with topology info. These DAIs's playback
& capture widgets will be freed when the sound card is unregistered
and the DAIs will be freed when cleaning up the component.
And a dobj is embedded into the struct snd_soc_dai_driver. Topology
can use the dobj to find the DAI drivers created by it and free them
when the topology component is removed.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Define soc_add_dai() as a wrapper to add a single DAI to a component.
It can be reused to register a DAI dynamically by topology.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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and 'asoc/topic/doc' into asoc-next
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Probing components can bring new DAI or DAI links based on the topology
info. This patch finds the unbound DAI links and bind them.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This function will return success immediately for a bound DAI link.
No need to look for the cpu/codec DAIs again.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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A machine driver can register the two ops.
When a DAI link is added or removed by a component's topology, the
ASoC core can call the ops to notify the machine driver for extra
intialization or destruction.
E.g. topology can create FE DAI links from a cpu DAI component, and
the machine driver may define an add_dai_link ops to set machine-specific
.init ops for the DAI link.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Implement a dai link list for the soc card.
Add APIs to add/remove a DAI links dynamically, e.g. by topology.
And a dobj is embedded into the struct snd_soc_dai_link. Topology can
use the dobj to find the links created by it and remove them when the
topology component is unloaded.
The predefined DAI links are reserved to keep backward compatibility.
And they will also be added to the list.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Just code refactoring, to reuse it if new DAI Links are added later
based on topology in component probing phase.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Define soc_init_dai_link() to wrap link initialization, to reuse it later
by snd_soc_instantiate_card() when adding new DAI links from topology in
component probing phase.
Move static func snd_soc_init_multicodec(), so that it can be reused by
soc_init_dai_link(). This saves adding a function declaration for
snd_soc_init_multicodec().
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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For DAPM resume, we should first change the power state of the
card and then recheck the endpoints. This ensures the dapm is
resumed first and then userspace can resume the streams.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Currently the number of DAI links is statically defined by the machine
driver at build time using an array. This makes it difficult to shrink/
grow the number of DAI links at runtime in order to reflect any changes
in topology.
We can change the DAI link array in the core to a list so that PCMs and
FE DAI links can be added and deleted at runtime to reflect changes in
use case and DSP topology. The machine driver can still register DAI links
as an array.
As the 1st step, this patch change the PCM runtime array to a list. A new
PCM runtime is added to the list when a DAI link is bound successfully.
Later patches will further implement the DAI link list.
More:
- define snd_soc_new/free_pcm_runtime() to create/free a runtime.
- define soc_add_pcm_runtime() to add a runtime to the rtd list.
- define soc_remove_pcm_runtimes() to clean up the runtime list.
- traverse the rtd list to probe the link components and dais.
- Add a field "num" to PCM runtime struct, used to specify the device
number when creating the pcm device, and for a soc card to access
its dai_props array.
- The following 3rd party machine/platform drivers iterate the rtd list
to check the runtimes:
sound/soc/intel/atom/sst-mfld-platform-pcm.c
sound/soc/intel/boards/cht_bsw_rt5645.c
sound/soc/intel/boards/cht_bsw_rt5672.c
sound/soc/intel/boards/cht_bsw_max98090_ti.c
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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In soc_link_dai_widgets() we refer to local widget variables as
playback/capture_widget, but they are really sink/source widgets,
so change the names accordingly
Suggested-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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when needed
We don't always need soc-compress in soc, here add a config item
SND_SOC_COMPRESS, when nobody select it, the soc-compress will
not be compiled.
Here also change Kconfig to 'select SND_SOC_COMPRESS' for drivers
that needed soc-compress.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Store return value of of_get_property() to a pointer of __be32 type as
device tree has big endian type. This fixes a sparse warning couple of
lines later when be32_to_cpup() is used to convert from big endian to
cpu endian.
The whole conversion is not really necessary, as we are only checking
if the value is zero or not, but I wanted to add it to remind in the
future that the data has to be converted before use. Compiler should
optimize the unnecessary operations away.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Adds DT binding for explicitly choosing a tdm mask for DAI and uses it
in simple-card. The API for snd_soc_of_parse_tdm_slot() has also been
changed.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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'asoc/topic/nuc900', 'asoc/topic/of-name' and 'asoc/topic/omap' into asoc-next
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'asoc/topic/davinci-vcif', 'asoc/topic/doc' and 'asoc/topic/dpcm' into asoc-next
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A number of functions and structures in the sound subsystem had incomplete
and/or obsolete DocBook comments, leading to warnings when the docs were
built. Correct those comments so that we can enjoy our audio in the
absence of warning noise.
Signed-off-by: Jonathan Corbet <corbet@lwn.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Current ASoC can add name_prefix for DAPM, and it is necessary for
route settings. This patch adds snd_soc_of_parse_audio_prefix() for
this purpose. It will be used with snd_soc_of_parse_audio_routing().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch fix following warnings.
Warning(.//sound/soc/soc-core.c:2855): No description found
for parameter 'platform_drv'
Warning(.//sound/soc/soc-core.c:2855): Excess function parameter
'platform_driver' description in 'snd_soc_add_platform'
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch fix following warning while make xmldocs.
Warning(.//sound/soc/soc-core.c:2148): No description found
for parameter 'ratio'
Add missing ":"
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Make sure the to free the card DAPM context if snd_soc_instantiate_card()
fails, otherwise the memory allocated for the DAPM widgets is leaked.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Use the card field of a component to indicate whether it is bound or not.
This makes a certain sense given that the field contains the card the
component is bound to and a component can only be bound to one card at a
time. And it also requires to unset the card field when the component is
unbound from the card.
This makes the probded flag redundant and it can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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A component can only be bound to a single card at a time. Binding it to
card while it is already bound to another will result in undefined
behavior.
As the undefined behavior might only manifest itself later on it is not
necessarily always straight forward to find the cause. To prevent this add
a check that refuses to bind a component to multiple cards as well as
prints a error describing the problem.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Dummy dai can be used by multiple sound card. But it only belong to one
card's dapm list. If another card use it, there will be dapm_assert_locked
warning.
[ 20.015782] WARNING: CPU: 1 PID: 661 at sound/soc/soc-dapm.c:124 dapm_assert_locked.isra.36+0x4c/0x58()
[ 20.025249] Modules linked in:
[ 20.028349] CPU: 1 PID: 661 Comm: aplay Not tainted 4.1.0-rc6-next-20150605-00004-gaee05d8-dirty #92
[ 20.037528] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree)
[ 20.044110] Backtrace:
[ 20.046614] [<80012e00>] (dump_backtrace) from [<80012fa0>] (show_stack+0x18/0x1c)
[ 20.054229] r6:809e8060 r5:00000000 r4:00000000 r3:00000000
[ 20.060002] [<80012f88>] (show_stack) from [<807a0f74>] (dump_stack+0x80/0x9c)
[ 20.067293] [<807a0ef4>] (dump_stack) from [<8002b144>] (warn_slowpath_common+0x7c/0xb4)
[ 20.075427] r5:0000007c r4:00000000
[ 20.079065] [<8002b0c8>] (warn_slowpath_common) from [<8002b1a0>] (warn_slowpath_null+0x24/0x2c)
[ 20.087898] r8:00000001 r7:88007c28 r6:ed94a680 r5:809e83e4 r4:ed83d6c0
[ 20.094747] [<8002b17c>] (warn_slowpath_null) from [<8058403c>] (dapm_assert_locked.isra.36+0x4c/0x58)
[ 20.104101] [<80583ff0>] (dapm_assert_locked.isra.36) from [<805842ec>] (dapm_mark_dirty+0x64/0xa4)
[ 20.113165] [<80584288>] (dapm_mark_dirty) from [<805853a8>] (soc_dapm_dai_stream_event.isra.42+0x30/0xc8)
[ 20.122863] r8:ed9b5dbc r7:00000000 r6:00000001 r5:00000001 r4:ed83d6c0
[ 20.129706] [<80585378>] (soc_dapm_dai_stream_event.isra.42) from [<80587e28>] (snd_soc_dapm_stream_event+0x78/0xa0)
[ 20.140264] r5:ee2ee62c r4:00000001
[ 20.143918] [<80587db0>] (snd_soc_dapm_stream_event) from [<8058957c>] (soc_pcm_prepare+0x138/0x21c)
[ 20.153058] r8:ed8d9480 r7:00000000 r6:ed9b0e00 r5:00000001 r4:ee2ee62c r3:00000000
...
This patch is to not probe the dummy component in soc_probe_component. Then
there is no widget created for dummy DAI, and also don't need to check the
dummy dai in dapm_connect_dai_link_widgets().
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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There are no more direct users of the snd_soc_codec DAPM field left. So we
can finally remove it and switch over to directly using the component DAPM
context and remove the dapm_ptr indirection.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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'asoc/topic/wm5100', 'asoc/topic/wm8741' and 'asoc/topic/wm8960' into asoc-next
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The topology core parses the FW topology file for known block types and
instanciates any common ALSA/ASoC objects that it discovers. The core
also passes any block that is does not understand to client component
drivers for enumeration.
The core exports some APIs to client drivers in order to load and unload
firmware topology data as use case require.
Currently the core deals with the following object types :-
o kcontrols. This includes TLV, enumerated and bytes controls.
o DAPM widgets. All types with any associated kcontrol.
o DAPM graph.
o FE PCM. FE PCM capabilities and configuration can be defined.
o BE DAI Link. BE DAI link capabilities and configuration can be defined.
o Codec <-> codec style links capabilities and configuration.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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If an ASoC component device does not have a device tree node, use its
parent's node instead, when looking for a matching DAI based on a
device tree reference.
This allows video device drivers to register a separate child device
for their ASoC side audio functionality. [And MFDs in general --
broonie]
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Instead of allocating two string buffers on stack and copying them
back, manipulate directly the target string buffer. This simplifies
the code well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Original issue is that the id field in the dai is not same as the id
in dai_driver when dai driver count == 1. This is due to the legacy
dai naming check, which could possibly cause issues if the audio drivers
written in assumption that dai->id would be always equal to dai_driver->id.
This assumption is true only if the dai driver count is greater than 1,
and false if dai driver count is 1. On Qcom Lpass driver we hit such
issue while adding support to apq8016.
The code path which falls back to legacy naming for cases where num_dai
== 1 does not check if there is any valid information in the dai_driver.
This patch fixes that by checking if the dai_driver has valid id and
name before falling back to legacy dai naming
Although the drivers can work around this issue by only using
dai->driver->id, but this patch attempts to fix the actual issue.
Suggested-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Kenneth Westfield <kwestfie@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The suspended flag will only be set if the CODEC bias level was either
STANDBY or OFF. This means we don't need to check for that on resume since
the condition will always be true.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There have been major modernization with the standard bus: in ALSA
sequencer core and HD-audio. Also, HD-audio receives the regmap
support replacing the in-house cache register cache code. These
changes shouldn't impact the existing behavior, but rather
refactoring.
In addition, HD-audio got the code split to a core library part and
the "legacy" driver parts. This is a preliminary work for adapting
the upcoming ASoC HD-audio driver, and the whole transition is still
work in progress, likely finished in 4.1.
Along with them, there are many updates in ASoC area as usual, too:
lots of cleanups, Intel code shuffling, etc.
Here are some highlights:
ALSA core:
- PCM: the audio timestamp / wallclock enhancement
- PCM: fixes in DPCM management
- Fixes / cleanups of user-space control element management
- Sequencer: modernization using the standard bus
HD-audio:
- Modernization using the standard bus
- Regmap support
- Use standard runtime PM for codec power saving
- Widget-path based power-saving for IDT, VIA and Realtek codecs
- Reorganized sysfs entries for each codec object
- More Dell headset support
ASoC:
- Move of jack registration to the card level
- Lots of ASoC cleanups, mainly moving things from the CODEC level to
the card level
- Support for DAPM routes specified by both the machine driver and DT
- Continuing improvements to rcar
- pcm512x enhacements
- Intel platforms updates
- rt5670 updates / fixes
- New platforms / devices: some non-DSP Qualcomm platforms, Google's
Storm platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC
Misc:
- ice1724: Improved ESI W192M support
- emu10k1: Emu 1010 fixes/enhancement"
* tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (411 commits)
ALSA: hda - set GET bit when adding a vendor verb to the codec regmap
ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T450
ALSA: hda - Fix another race in runtime PM refcounting
ALSA: hda - Expose codec type sysfs
ALSA: ctl: fix to handle several elements added by one operation for userspace element
ASoC: Intel: fix array_size.cocci warnings
ASoC: n810: Automatically disconnect non-connected pins
ASoC: n810: Consistently pass the card DAPM context to n810_ext_control()
ASoC: davinci-evm: Use card DAPM context to access widgets
ASoC: mop500_ab8500: Use card DAPM context to access widgets
ASoC: wm1133-ev1: Use card DAPM context to access widgets
ASoC: atmel: Improve machine driver compile test coverage
ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary
ALSA: control: Fix a typo of SNDRV_CTL_ELEM_ACCESS_TLV_* with SNDRV_CTL_TLV_OP_*
ALSA: usb-audio: Don't attempt to get Microsoft Lifecam Cinema sample rate
ASoC: rnsd: fix build regression without CONFIG_OF
ALSA: emu10k1: add toggles for E-mu 1010 optical ports
ALSA: ctl: fill identical information to return value when adding userspace elements
ALSA: ctl: fix a bug to return no identical information in info operation for userspace controls
ALSA: ctl: confirm to return all identical information in 'activate' event
...
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git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial
Pull trivial tree from Jiri Kosina:
"Usual trivial tree updates. Nothing outstanding -- mostly printk()
and comment fixes and unused identifier removals"
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial:
goldfish: goldfish_tty_probe() is not using 'i' any more
powerpc: Fix comment in smu.h
qla2xxx: Fix printks in ql_log message
lib: correct link to the original source for div64_u64
si2168, tda10071, m88ds3103: Fix firmware wording
usb: storage: Fix printk in isd200_log_config()
qla2xxx: Fix printk in qla25xx_setup_mode
init/main: fix reset_device comment
ipwireless: missing assignment
goldfish: remove unreachable line of code
coredump: Fix do_coredump() comment
stacktrace.h: remove duplicate declaration task_struct
smpboot.h: Remove unused function prototype
treewide: Fix typo in printk messages
treewide: Fix typo in printk messages
mod_devicetable: fix comment for match_flags
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'asoc/topic/max98090', 'asoc/topic/max98925' and 'asoc/topic/nuc900' into asoc-next
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Current snd_soc_runtime_set_dai_fmt() is called after
soc_probe_link_dais(). this means snd_soc_dai_set_fmt() will be
called after soc_new_pcm().
Before appling 1efb53a220b78fdfdbb97b726a2156713e75bdab
(ASoC: simple-card: Remove support for setting differing DAI formats)
simple-card user had (1) snd_soc_dai_set_fmt() -> soc_new_pcm(),
but, after that it is (2) soc_new_pcm() -> snd_soc_dai_set_fmt().
At least rsnd driver is assuming (1) pattern.
This patch move snd_soc_dai_set_fmt() into soc_probe_link_dais()
after the dai_link->init section to solve this issue.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-core
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-core
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If the registration of a debugfs directory fails this is treated as a
non-fatal error in ASoC and operation continues as normal. This means we
need to be careful and check if the parent debugfs directory exists if we
try to register a debugfs file or sub-directory. Otherwise we might end up
passing NULL for the parent and the file or directory will be registered in
the top-level debugfs directory.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Failing to register the debugfs entries is not fatal and will not affect
normal operation of the sound card. Don't abort the card registration if
soc_dpcm_debugfs_add() fails.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Create the card debugfs directory at the begining of the initilization
rather then the end as various steps in the initilization sequence will try
to register files and sub-directories in the card directory.
Fixes: 4e2576bd36a1 ("ASoC: soc-core: initialize debugfs in snd_soc_instantiate_card()")
Reported-by: Fabio Estevam <festevam@gmail.com>
Reported-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Linux 4.0-rc7
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The only two users of the suspend_bias_level field were two rather old
drivers which weren't exactly doing things by the book. Those drivers have
been updated and field is now unused and can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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