A bunch of driver fixes plus one core fix which fixes problems with
misreporting values from _SX controls following a recent refactoring.
This had gone unnoticed as such controls are quite rare.
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Merge tag 'asoc-fix-v4.3-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.3
A bunch of driver fixes plus one core fix which fixes problems with
misreporting values from _SX controls following a recent refactoring.
This had gone unnoticed as such controls are quite rare.
The rate constraints in this driver are shared between all device
instances. It should not be (and is not) modified at runtime, so make them
const.
While we are at it also change the type of the rates array from u32 to
unsigned int. While both resolve to the same type, u32 is usually used to
empathize that the value is for a 32-bit hardware register or similar which
makes it slightly confusing in this context.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The rate constraints in this board driver are not modified at runtime, so
make them const.
While we are at it also remove the unnecessary 0 initializer for the mask
field.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The rate constraints in this board driver are not modified at runtime, so
make them const.
While we are at it also remove the unnecessary 0 initializer for the mask
field.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
share lrck_tx to lrck_rx when symmetric_rates enabled.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
support max 8 channels capture, please add property
'rockchip,capture-channels' in dts to enable this,
if not, support 2 channels capture default.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add driver for the digital imput to PWM output stereo
class D amplifier. It comes with filter, digitally
controlled gain, an equalizer and a dmphase filter.
Signed-off-by: Songjun Wu <songjun.wu@atmel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add AC'97 support to fsl-asoc-card using generic
ASoC AC'97 CODEC.
The SSI controller will silently enable any TX
AC'97 slots that have their bits set in SLOTREQ
received from CODEC and then will redirect some
of playback samples there.
That's why it is important to make sure that
any of CODEC playback slots that can pull samples
are set to slots 3/4 (standard PCM playback slots).
Currently, this applies to S/PDIF slots as they
were seen to pull samples sometimes even with
S/PDIF output being disabled.
Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_codec_driver.set_pll is implemented to configure the FLL.
The codec internal SYSCLK can be from either the MCLK pin directly,
or the FLL. This is configured by snd_soc_codec_driver.set_pll.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
While there is nothing wrong with the transfer_ack_begin and
transfer_ack_end callbacks per-se, the last documented user was part of the
alsa-driver 0.5.12a package, which was released 14 years ago and even
predates the upstream integration of the ALSA core and has subsequently
been superseded by newer alsa-driver releases.
This seems to indicate that there is no need for having these callbacks and
they are just cruft that can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Given a kcontrol, we may want to access the parent widget
and it's associated data. So export function to return it.
Signed-off-by: Mythri P K <mythri.p.k@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For DAI link events, DSPs would like to get notified for startup
and shutdown event as well apart for existing hw_params. This
helps managing DSP resource allocation and freeup on these events
So add support for startup and shutdown for
snd_soc_dai_link_event()
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For DSPs we can define processing blocks as DAPM PGA widgets.
Some of these proceesing blocks can be configured by usermode
like EQ etc. So we need to add support of kcontrol for PGA
widgets.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mythri P K <mythri.p.k@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for deprecating ioremap_cache() convert its usage in
skl-nhlt to memremap.
Signed-off-by: Dan Williams <dan.j.williams@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currentlly, we use Synopsys DesignWare DMA Controller for
baytrail/haswell/broadwell ADSP firmware loading, but for
skylake, we don't use it, compiling sst-firmware.c may
introduce error when CONFIG_DW_DMAC_CORE is not enabled:
sound/built-in.o: In function `sst_dma_new':
(.text+0xd7b38): undefined reference to `dw_dma_probe'
sound/built-in.o: In function `sst_dma_free':
(.text+0xd7c0a): undefined reference to `dw_dma_remove'
Here we only compile sst-firmware when CONFIG_DW_DMAC_CORE
is selected, to fix the linking error issue.
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We want these to be zero or one, but by mistake we also accept negative
values. It's harmless but we should still clean it up.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Valid values for "deemph" are zero and one but we accidentally allow
negative values as well. It's harmless but it causes static checker
warnings and we may as well clean it up.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Valid values for "deemph" are zero and one, but we mistakenly allow
negative values as well. It's harmless but we may as well clean it up.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Valid values for "deemph" are zero and one, but we accidentally allow
negative values as well. It's harmless, but static checkers complain
and we may as well clean it up.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should only accept "deemph" values of zero and one, but by mistake we
accept negatives as well. It's harmless but let's clean it up anyway.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The code tries to verify that "deemph" is zero or one, but it fails to
account for that it can be negative. It's harmless because negatives
are treated the same as one, but we should fix it just to silence the
static checker warning.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We try to check that "deemph" is zero or one, but because of an
underflow bug, it can also be negative. It's fine, negative values are
handled as non-zero. But it's messy and static checkers complain so
let's fix it.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
snd_soc_limit_volume() operates on a card and the CODEC that is passed in
is only used to look up the card. Let it directly take the card instead.
This makes it possible to use it when no snd_soc_codec is available.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
By doing software reset of wm8962 in pm_resume, all registers which
have already been set will be reset to default value without regmap
interface be involved, thus driver need to mark cache_dirty flag,
to let regcache can be updated by regcache_sync().
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_ssi_use_busif() and rsnd_ssi_is_pin_sharing() are the function
which returns current SSI status. But these requests duplicated parameter.
This patch removes duplicated parameter.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_dma_to_ssi() is no longer used, let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_mod_hw_start/stop were unclear naming.
It became rsnd_mod_power_on/off by this patch
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd_dai_call returns immediately if rsnd_mod_call return fail.
Thus, each callback-count can be unbalanced for example .init was OK,
start was OK, but, .stop was not OK. This case .quit should be called
but isn't called. And, rsnd_dai_stream_quit() also not be called.
rsnd_dai_call() should call all .stop/.quit eventhough it returns error.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd_dai_call didn't count callback-count if callback
wasn't implemented. Thus, it counts can be unbalance.
ex) .start : implemented
.stop : not implemented
This patch solve this issue
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Indicating each module method as debug message before executing is
readable/understandable.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
bfc0cfe("ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx()")
tidyuped devm_request_irq() option from ssi to mod, but devm_free_irq()
on rsnd_ssi_dma_remove() didn't modified. This patch fixups this issue.
Otherwise kernel will output WARNING message.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
struct rsnd_gen :: res array size should be RSND_BASE_MAX,
not RSND_REG_MAX. This patch fixup it, and indicates whether
each data array size is based on what
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
debug meesage for rsnd_mod_read() should be prints after read
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Use the new snd_pcm_hw_constraint_single() helper function rather than
installing a list constraint with a single value. Since
snd_pcm_hw_constraint_single() sets a static constraint while
snd_pcm_hw_constraint_list() sets a dynamic constraint the former is
slightly more efficient and it also needs less code.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function rather than
installing a list constraint with a single value. Since
snd_pcm_hw_constraint_single() sets a static constraint while
snd_pcm_hw_constraint_list() sets a dynamic constraint the former is
slightly more efficient and it also needs less code.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function rather than
installing a list constraint with a single value. Since
snd_pcm_hw_constraint_single() sets a static constraint while
snd_pcm_hw_constraint_list() sets a dynamic constraint the former is
slightly more efficient and it also needs less code.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function rather than
installing a list constraint with a single value. Since
snd_pcm_hw_constraint_single() sets a static constraint while
snd_pcm_hw_constraint_list() sets a dynamic constraint the former is
slightly more efficient and it also needs less code.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer and is slightly shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer and is slightly shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer and is slightly shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer and is slightly shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer and is slightly shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
While we are at it also fix some code style issues in the affected lines.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The register value to enable gpio2 was incorrect. So fix it.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The implementation of ssm2518_register_volatile always returns false,
this behavior is the same as no .volatile_reg callback implementation
when cache_type != REGCACHE_NONE.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
adau193x_reg_volatile() always return false.
This seems pointless because current code uses REGCACHE_NONE cache_type
which is supposed to be volatile.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
When MIDI buffer stores two or more MIDI messages, TASCAM driver
transfers asynchronous transactions including one MIDI message and
extra bytes from second MIDI message.
This commit fixes this bug by clearing needless bytes in the buffer. The
consumed bytes are already calculated correctly, thus the sequence of
transactions is already correct.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In transactions for MIDI messages, the first byte is used for label and
the rest is for MIDI bytes. In current code, these are handled correctly,
while there's a small mistake for loop condition to include meaningless
statement.
This commit adds two local variables for them and improve the loop
condition.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the callback function of asynchronous MIDI port, the intension of some
local variables are not clear.
This commit improves them. The 'len' variable is used to calculate the
number of MIDI bytes including in the transaction. The 'consume' variable
is used to return the actual number of consumed bytes in ALSA MIDI buffer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the callback function of asynchronous MIDI port, some local variables
are declared 'unsigned int', while they're assigned to int value of return
from snd_rawmidi_transmit_peek().
This commit fixes the type.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The given buffer to callback function is cleared in caller side.
This commit removes buffer initialization in callee side.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HDA codec driver issues snd_hda_codec_reset() at the error path of
PCM build. This was needed in the earlier code base, but the recent
rewrite to use the standard bus binding made this a deadlock:
modprobe D 0000000000000005 0 720 716 0x00000080
Call Trace:
[<ffffffff816a5dbe>] schedule+0x3e/0x90
[<ffffffff816a61a5>] schedule_preempt_disabled+0x15/0x20
[<ffffffff816a7ae5>] __mutex_lock_slowpath+0xb5/0x120
[<ffffffff816a7b6b>] mutex_lock+0x1b/0x30
[<ffffffff8148656b>] device_release_driver+0x1b/0x30
[<ffffffff81485c15>] bus_remove_device+0x105/0x180
[<ffffffff814822b9>] device_del+0x139/0x260
[<ffffffffa05e0ec5>] snd_hdac_device_unregister+0x25/0x30 [snd_hda_core]
[<ffffffffa074fa6a>] snd_hda_codec_reset+0x2a/0x70 [snd_hda_codec]
[<ffffffffa075007b>] snd_hda_codec_build_pcms+0x18b/0x1b0 [snd_hda_codec]
[<ffffffffa074a44e>] hda_codec_driver_probe+0xbe/0x140 [snd_hda_codec]
[<ffffffff81486ac4>] driver_probe_device+0x1f4/0x460
[<ffffffff81486dc0>] __driver_attach+0x90/0xa0
[<ffffffff81484844>] bus_for_each_dev+0x64/0xa0
[<ffffffff814862de>] driver_attach+0x1e/0x20
[<ffffffff81485e7b>] bus_add_driver+0x1eb/0x280
[<ffffffff81487680>] driver_register+0x60/0xe0
[<ffffffffa074a0da>] __hda_codec_driver_register+0x5a/0x60 [snd_hda_codec]
[<ffffffffa070a01e>] realtek_driver_init+0x1e/0x1000 [snd_hda_codec_realtek]
[<ffffffff810002f3>] do_one_initcall+0xb3/0x200
[<ffffffff816a1fc5>] do_init_module+0x60/0x1f8
[<ffffffff810ee5c3>] load_module+0x1653/0x1bd0
[<ffffffff810eed48>] SYSC_finit_module+0x98/0xc0
[<ffffffff810eed8e>] SyS_finit_module+0xe/0x10
[<ffffffff816aa032>] entry_SYSCALL_64_fastpath+0x16/0x75
The simple fix is just to remove this call, since we don't need to
think about unbinding at there any longer.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=948758
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By setting the MODALIAS variable in uevents, userspace helpers will be
enabled to load modules via the module alias associated with a device.
This information is required to automatically load HDA codec drivers
instead of having to explicitly request the various modules in the HDA
core code.
[Note that currently the legacy HDA controller driver tries to bind
codec modules manually. It's for supporting the fallback generic
drivers. This new udev modalias support was added rather for ASoC
HDA ext drivers, since this addition itself won't hurt the legacy HDA
-- tiwai]
[Use the common helper function to generate the modalias -- tiwai]
Signed-off-by: Thierry Reding <treding@nvidia.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Finally we have a proper infrastructure to generate the modaliases
automatically, let's move to hda_device_id from the legacy
hda_codec_preset that contains basically the same information.
The patch function hook is stored in driver_data field, which is long,
and we need an explicit cast. Other than that, the conversion is
mostly straightforward. Each entry is even simplified using a macro,
and the lengthy (and error-prone) manual modaliases got removed.
As a result, we achieved a quite good diet:
14 files changed, 407 insertions(+), 595 deletions(-)
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch just adds modalias sysfs entry to each hdaudio bus entry.
[rewritten to call the common helper function by tiwai]
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch provide a new common helper function,
snd_hdac_codec_modalias(), to give the codec modalias name string.
This function will be used by multiple places in the later patches.
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For generating modalias entries automatically, move the definition of
struct hda_device_id to linux/mod_devicetable.h and add the handling
of this record in file2alias helper. The new modalias is represented
with combination of vendor id, device id, and api version as
"hdaudio:vNrNaN".
This patch itself doesn't convert the existing modaliases. Since they
were added manually, this patch won't give any regression by itself at
this point.
[Modified the modalias format to adapt the api_version field, and drop
invalid ANY_ID definition by tiwai]
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More drm-misc for 4.4.
- fb refcount fix in atomic fbdev
- various locking reworks to reduce drm_global_mutex and dev->struct_mutex
- rename docbook to gpu.tmpl and include vga_switcheroo stuff, plus more
vga_switcheroo (Lukas Wunner)
- viewport check fixes for atomic drivers from Ville
- DRM_DEBUG_VBL from Ville
- non-contentious header fixes from Mikko Rapeli
- small things all over
* tag 'topic/drm-misc-2015-10-19' of git://anongit.freedesktop.org/drm-intel: (31 commits)
drm/fb-helper: Fix fb refcounting in pan_display_atomic
drm/fb-helper: Set plane rotation directly
drm: fix mutex leak in drm_dp_get_mst_branch_device
drm: Check plane src coordinates correctly during page flip for atomic drivers
drm: Check crtc viewport correctly with rotated primary plane on atomic drivers
drm: Refactor plane src coordinate checks
drm: Swap w/h when converting the mode to src coordidates for a rotated primary plane
drm: Don't leak fb when plane crtc coodinates are bad
ALSA: hda - Spell vga_switcheroo consistently
drm/gem: Use kref_get_unless_zero for the weak mmap references
drm/vgem: Drop vgem_drm_gem_mmap
drm: Fix return value of drm_framebuffer_init()
drm/gem: Use container_of in drm_gem_object_free
drm/gem: Check locking in drm_gem_object_unreference
drm/gem: Drop struct_mutex requirement from drm_gem_mmap_obj
drm/i810_drm.h: include drm/drm.h
r128_drm.h: include drm/drm.h
savage_drm.h: include <drm/drm.h>
gpu/doc: Convert to markdown harder
gpu/doc: Add vga_switcheroo documentation
...
- dmc fixes from Animesh (not yet all) for deeper sleep states
- piles of prep patches from Ville to make mmio functions type-safe
- more fbc work from Paulo all over
- w/a shuffling from Arun Siluvery
- first part of atomic watermark updates from Matt and Ville (later parts had to
be dropped again unfortunately)
- lots of patches to prepare bxt dsi support ( Shashank Sharma)
- userptr fixes from Chris
- audio rate interface between i915/snd_hda plus kerneldoc (Libin Yang)
- shrinker improvements and fixes (Chris Wilson)
- lots and lots of small patches all over
* tag 'drm-intel-next-2015-10-10' of git://anongit.freedesktop.org/drm-intel: (134 commits)
drm/i915: Update DRIVER_DATE to 20151010
drm/i915: Partial revert of atomic watermark series
drm/i915: Early exit from semaphore_waits_for for execlist mode.
drm/i915: Remove wrong warning from i915_gem_context_clean
drm/i915: Determine the stolen memory base address on gen2
drm/i915: fix FBC buffer size checks
drm/i915: fix CFB size calculation
drm/i915: remove pre-atomic check from SKL update_primary_plane
drm/i915: don't allocate fbcon from stolen memory if it's too big
Revert "drm/i915: Call encoder hotplug for init and resume cases"
Revert "drm/i915: Add hot_plug hook for hdmi encoder"
drm/i915: use error path
drm/i915/irq: Fix misspelled word register in kernel-doc
drm/i915/irq: Fix kernel-doc warnings
drm/i915: Hook up ring workaround writes at context creation time on Gen6-7.
drm/i915: Don't warn if the workaround list is empty.
drm/i915: Resurrect golden context on gen6/7
drm/i915/chv: remove pre-production hardware workarounds
drm/i915/snb: remove pre-production hardware workaround
drm/i915/bxt: Set time interval unit to 0.833us
...
This driver also supports adau1328, thus add adau1328 to ad193x_spi_id.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the SKL I2S machine driver using Realtek ALC286S codec
in I2S mode.
Signed-off-by: Omair M Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use resource managed API then we can remove snd_dmaengine_pcm_unregister()
and snd_soc_unregister_component() calls in .probe error path and .remove.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Let's leave space for the NUL char otherwise the static checkers
complain that we go beyond the end of the array.
Fixes: 53b3ffee78 ('ALSA: firewire-tascam: change device probing processing')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device has no mixer (and identifies itself as such), so just skip
the mixer definition.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the Zoom R16/24 (tx_length_quirk set), when calculating the maximum
sample frequency, consideration must be made for the fact that four bytes
of the packet contain a length descriptor and consequently must not be
counted as part of the audio data.
This is corroborated by the wMaxPacketSize for this device, which is 108
bytes according for the USB playback endpoint descriptor. The frame size
is 8 bytes (2 channels of 4 bytes each), and the 108 bytes thus work out
as 13 * 8 + 4, i.e. corresponding to 13 frames plus the additional 4 byte
length descriptor.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Zoom R16/24 have a nonstandard playback format where each isochronous
packet contains a length descriptor in the first four bytes. (Curiously,
capture data does not contain this and requires no quirk.)
The quirk involves adding the extra length descriptor whenever outgoing
isochronous packets are generated, both in pcm.c (outgoing audio) and
endpoint.c (silent data).
In order to make the quirk as unintrusive as possible, for
pcm.c:prepare_playback_urb(), the isochronous packet descriptors are
initially set up in the same way no matter if the quirk is enabled or not.
Once it is time to actually copy the data into the outgoing packet buffer
(together with the added length descriptors) the isochronous descriptors
are adjusted in order take the increased payload length into account.
For endpoint.c:prepare_silent_urb() it makes more sense to modify the
actual function, partly because the function is less complex to start with
and partly because it is not as time-critical as prepare_playback_urb()
(whose bulk is run with interrupts disabled), so the (minute) additional
time spent in the non-quirk case is motivated by the simplicity of having
a single function for all cases.
The quirk is controlled by the new tx_length_quirk member in struct
snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c
and endpoint.c from quirks.c in a similar manner to the txfr_quirk member
in the same structs.
In contrast to txfr_quirk however, the quirk is enabled directly in
quirks.c:create_standard_audio_quirk() by checking the USB ID in that
function. Another option would be to introduce a new
QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk
very plain to see in the quirk table, but it was felt that the additional
code needed to implement it this way would just make the implementation
more complex with no real gain.
Tested with a Zoom R16, both by doing capture and playback separately
using arecord and aplay (8 channel capture and 2 channel playback,
respectively), as well as capture and playback together using Ardour, as
well as Audacity and Qtractor together with jackd.
The R24 is reportedly compatible with the R16 when used as an audio
interface. Both devices share the same USB ID and have the same number of
inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the
patch.
Regression tested using an Edirol UA-5 in both class compliant (16-bit)
and "advanced" (24 bit, forces the use of quirks) modes.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Tested-by: Panu Matilainen <pmatilai@laiskiainen.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireOne is based on OXFW971 and ALSA OXFW driver can support it.
These are values of identical registers.
$ ./firewire-request /dev/fw1 read 0xfffff0050000
result: 97100105
$ ./firewire-request /dev/fw1 read 0xfffff0090020
result: 39373100
This commit adds an entry for this model. This model has physical controls
and its MIDI control messages are transferred to second MIDI data stream
multiplexed in one MIDI conformant data channel.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, sequence multiplexing is applied to MIDI conformant data
channel. As a result, eight MIDI data streams are included in the channel.
Although ALSA AM824 data block processing layer implements this
multiplexing, current OXFW driver doesn't utilize it due to wrong
calculation of MIDI ports.
This commit fixes this bug to add proper calculation. Although this commit
allows to use 8 MIDI data streams, the number of available MIDI ports is
limited by the number of ALSA MIDI ports added by the driver.
Fixes: df075feefbd3('ALSA: firewire-lib: complete AM824 data block processing layer')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current OXFW driver calculates the number of MIDI ports just before adding
ALSA MIDI ports. It's convenient for some devices with quirks to move
these codes before handling quirks.
This commit implements this idea.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When asynchronous MIDI port is closed before callbacked, the callback
function causes NULL pointer dereference to missing MIDI substream.
This commit fixes this bug.
Fixes: e8a40d9bcb23('ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages')
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The contents of Config ROM in firewire device structure are already
aligned to CPU-endianness. Thus, no need to convert it again.
This commit removes needless conversions
Fixes: 9edf723fd858('ALSA: firewire-digi00x: add skeleton for Digi 002/003 family')
Fixes: c0949b278515('ALSA: firewire-tascam: add skeleton for TASCAM FireWire series')
Reported-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, metering is supported for BeBoB based models
customized by M-Audio. The data in transaction is aligned to
big-endianness, while in the driver code u16 typed variable is assigned
to the data. This causes sparse warnings.
bebob_maudio.c:651:31: warning: cast to restricted __be16
bebob_maudio.c:651:31: warning: cast to restricted __be16
bebob_maudio.c:651:31: warning: cast to restricted __be16
bebob_maudio.c:651:31: warning: cast to restricted __be16
This commit fixes this bug by using __be16 variable for the data.
Fixes: 3149ac489ff8('ALSA: bebob: Add support for M-Audio special Firewire series')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, snd_efw_command_get_phys_meters() was added to handle
metering data. The given buffer is used to save transaction result and to
convert between endianness. But this causes sparse warnings.
fireworks_command.c:269:25: warning: incorrect type in argument 1 (different base types)
fireworks_command.c:269:25: expected unsigned int [usertype] *p
fireworks_command.c:269:25: got restricted __be32 [usertype] *
This commit fixes this bug.
Fixes: bde8a8f23bbe('ALSA: fireworks: Add transaction and some commands')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, u32 data was assigned to __be32 variable instead of an
int variable. This is not enough solution because it still causes sparse
warnings.
dice.c:80:23: warning: incorrect type in assignment (different base types)
dice.c:80:23: expected restricted __be32 [usertype] value
dice.c:80:23: got unsigned int
dice.c:81:21: warning: restricted __be32 degrades to integer
dice.c:81:46: warning: restricted __be32 degrades to integer
This commit fixes this bug.
Fixes: 7c2d4c0cf5ba('ALSA: dice: Split transaction functionality into a file')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some local variables in some functions are typed as unsigned int, while
__be32 value is assigned to them. This causes sparse warnings.
dice-stream.c:50:17: warning: incorrect type in assignment (different base types)
dice-stream.c:50:17: expected unsigned int [unsigned] channel
dice-stream.c:50:17: got restricted __be32 [usertype] <noident>
dice-stream.c:74:17: warning: incorrect type in assignment (different base types)
dice-stream.c:74:17: expected unsigned int [unsigned] channel
dice-stream.c:74:17: got restricted __be32 [usertype] <noident>
This commit fixes this bug.
Fixes: 288a8d0cb04f('ALSA: dice: Change the way to start stream')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently everyone and their dog has their own favourite spelling
for vga_switcheroo. This makes it hard to grep dmesg for log entries
relating to vga_switcheroo. It also makes it hard to find related
source files in the tree.
vga_switcheroo.c uses pr_fmt "vga_switcheroo". Use that everywhere.
Signed-off-by: Lukas Wunner <lukas@wunner.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: http://patchwork.freedesktop.org/patch/msgid/9b0175319ce78d831acfcf11e4c6c760f826b0e3.1444663039.git.lukas@wunner.de
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>
A former commit moves oxfw-related codes to a sub-directory, while it
forgot to remove an entry from Makefile in parent directory.
Fixes: 1a4e39c2e5ca('ALSA: oxfw: Move to its own directory')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When committed to upstream, these four modules had wrong entries for
Makefile. This forces them to be loadable modules even if they're set
as built-in.
This commit fixes this bug.
Fixes: b5b04336015e('ALSA: fireworks: Add skelton for Fireworks based devices')
Fixes: fd6f4b0dc167('ALSA: bebob: Add skelton for BeBoB based devices')
Fixes: 1a4e39c2e5ca('ALSA: oxfw: Move to its own directory')
Fixes: 14ff6a094815('ALSA: dice: Move file to its own directory')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent commit [7fbe824a0f0e: ALSA: hda - Update mixer name for the
lower codec address] tried to improve the mixer chip name assignment
in the order of codec address. However, this fix was utterly bogus;
it checks the field set in each codec, thus this value is reset at
each codec creation, of course. For really handling this priority,
the assignment has to be remembered in the common place, namely in
hda_bus, instead of hda_codec.
Fixes: 7fbe824a0f ('ALSA: hda - Update mixer name for the lower codec address')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
My static checker complains because tscm->spec->midi_capture_ports is
either 2 or 4 but the tscm->tx_midi_substreams[] array has 4 elements so
this is possibly off by one. I have looked at the code and I think it
should be >= instead of > as well.
Fixes: 107cc0129a ('ALSA: firewire-tascam: add support for incoming MIDI messages by asynchronous transaction')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We recently tried to add some new code to support turning the LED on and
off but the code in snd_tscm_transaction_reregister() is unreachable.
Fixes: e65e2cb99e ('ALSA: firewire-tascam: Turn on/off FireWire LED')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch rechecks the jack detect status after resuming from S3.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The index_cache is per instance run time state but rt298_index_def is not.
Make rt298_index_def const and make a copy of memory for index_cache rather
than directly use the rt298_index_def.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SX_TLV controls are intended for situations where the register behind
the control has some non-zero value indicating the minimum gain
and then gains increasing from there and eventually overflowing through
zero.
Currently every CODEC implementing these controls specifies the minimum
as the non-zero value for the minimum and the maximum as the number of
gain settings available.
This means when the info callback subtracts the minimum value from the
maximum value to calculate the number of gain levels available it is
actually under reporting the available levels. This patch fixes this
issue by adding a new snd_soc_info_volsw_sx callback that does not
subtract the minimum value.
Fixes: 1d99f2436d ("ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV")
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Tested-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Probing from Gen1 is not error. This patch fixup it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some of the default value on rt298_index_def are incorrect. Change
them to the correct value.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PCM timer is not always used. For embedded device, we need an interface
to disable it when it is not needed, to shrink the kernel size and
memory footprint, here add CONFIG_SND_PCM_TIMER for it.
When both CONFIG_SND_PCM_TIMER and CONFIG_SND_TIMER is unselected,
about 25KB saving bonus we can get.
Please be noted that when disabled, those stubs who using pcm timer
(e.g. dmix, dsnoop & co) may work incorrectlly.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Nocturn needs the MIDI_RAW_BYTES quirk, like other Novation devices.
Tested that the Nocturn shows up in aconnect, and that it can be used
as a control surface (using the xtor synthesizer patch editor).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In most cases, we prefer the onboard codec as the primary device, thus
it's better to set it as the mixer name. Currently, however, the
mixer name is updated per the device instantiation order, and user
gets often HDMI/DP or other seen as a mixer chip name. Also, if a
codec name is renamed by the driver, the old chip name might be left
still as the mixer name.
This patch addresses these issues by remembering the chip address that
was referred as the mixer name. When a codec with the same or lower
address gives its name, renew the mixer name accordingly, as it's
either the update of the codec name or we get likely the more
appropriate chip as the reference.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few multiple codec drivers do renaming the chip_name string but all
these are open-coded and some of them have even no error check. Let's
make common helpers to do it properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cirrus codecs have also fine power controls on each widget, thus it
gets benefit from the recent widget power-saving feature. As we
haven't seen any obvious regressions with tests on some MacBooks,
let's try to enable it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We cap the upper bound of "idx" but not the negative side. Let's make
it unsigned to fix this.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Gen1 SRU support was created for preparation of Gen2 SRC support,
but no-one is using this feature (sampling rate convert) on Gen1.
BockW had used SRU before, but it was pass through mode.
This means it is same as SSI. And BockW "platform base" code was
removed from upstream code. It is now supported via DT, but it doesn't
use SRU. More detail, r8a7778.dtsi has "rcar_sound,src" entry, but
no-one is using this feature today. SRU probing has no relation to this
removing. This means there is no effect for DT compatibility, no issues
on upstream kernel.
Gen2 SRC was created from Gen1 SRU, these are similar but not same IP.
Keeping Gen1 SRU in current driver is a little bit difficult,
and no-one is using it today. Gen1 sound is still supported via SSI.
Gen1 SRU support will be removed in the next kernel version.
This patch announces it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Rounding must take place before multiplication with the frame size, since
each packet contains a whole number of frames.
We must also properly consider the data interval, as a larger data
interval will result in larger packets, which, depending on the sampling
frequency, can result in packet sizes that are less than integral
multiples of the packet size for a lower data interval.
Detailed explanation and rationale:
The code before this commit had the following expression on line 613 to
calculate the maximum isochronous packet size:
maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
>> (16 - ep->datainterval);
Here, ep->freqmax is the maximum assumed sample frequency, calculated from the
nominal sample frequency plus 25%. It is ultimately derived from ep->freqn,
which is in the units of frames per packet, from get_usb_full_speed_rate()
or usb_high_speed_rate(), as applicable, in Q16.16 format.
The expression essentially adds the Q16.16 equivalent of 0.999... (i.e.
the largest number less than one) to the sample rate, in order to get a
rate whose integer part is rounded up from the fractional value. The
multiplication with (frame_bits >> 3) yields the number of bytes in a
packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back
to an integer, taking into consideration the bDataInterval field of the
endpoint descriptor (which describes how often isochronous packets are
transmitted relative to the (micro)frame rate (125us or 1ms, for USB high
speed and full speed, respectively)). For this discussion we will initially
assume a bDataInterval of 0, so the second line of the expression just
converts the Q16.16 value to an integer.
In order to illustrate the problem, we will set frame_bits 64, which
corresponds to a frame size of 8 bytes.
The problem here is twofold. First, the rounding operation consists
of the addition of 0x0.ffff and subsequent conversion to integer, but as the
expression stands, the conversion to integer is done after multiplication
with the frame size, rather than before. This results in the resulting
maxsize becoming too large.
Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is
0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000.
The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 .
However, if we do the number of bytes calculation in a less obscure way it's
more apparent what the true corresponding packet size is: we get
ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612,
and the 8000 is the number of isochronous packets per second on a high
speed USB connection (125 us microframe interval).
This is fixed by performing the complete rounding operation prior to
multiplication with the frame rate.
The second problem is that when considering the ep->datainterval, this
must be done before rounding, in order to take the advantage of the fact
that if the number of bytes per packet is not an integer, the resulting
rounded-up integer is not necessarily a factor of two when the data
interval is increased by the same factor.
For instance, assuming a freqency of 41 kHz, the resulting
bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or
0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0),
this means that 6 frames per packet are needed, whereas with a data
interval of 2 we need 10.25, i.e. 11 frames needed.
Rephrasing the maxsize expression to:
maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
(frame_bits >> 3);
for the above 96 kHz example we instead get
((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value.
We can also do the calculation with a non-integer sample rate which is when
rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn =
0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)):
Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down)
True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56
New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56
This is also corroborated by the wMaxPacketSize check on line 616. Assume
that wMaxPacketSize = 104, with ep->maxpacksize then having the same value.
As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to
(104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize
calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111
(with decimals 111.99988). Clearly, we should get back the 104 here,
which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 .
(The error has not been a problem because it only results in maxsize being
a bit too big which just wastes a couple of bytes, either as a result of
the first maxsize calculation, or because the resulting calculation will
hit the wMaxPacketSize value before the packet is too big, resulting in
fixing the size to wMaxPacketSize even though the packet is actually not
too long.)
Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Compiling the hdac extended core on arm fails with below error:
sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_writel':
>> sound/hda/ext/hdac_ext_bus.c:29:2: error: implicit declaration of
>> function
+'writel' [-Werror=implicit-function-declaration]
writel(value, addr);
^
sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_readl':
>> sound/hda/ext/hdac_ext_bus.c:34:2: error: implicit declaration of
>> function
+'readl' [-Werror=implicit-function-declaration]
return readl(addr);
This is fixed by explicitly including io.h
Fixes: 99463b3a39 - ('ALSA: hda: provide default bus io ops extended hdac')
Reported-by: kbuild test robot <lkp@intel.com>
Suggested-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It is not necessary to set registers volatile. So, return false
for default case of rt298_volatile_register.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, this driver picks up model name with be32_to_cpu() macro
to align characters. This is wrong operation because the result is
different depending on CPU endiannness.
Additionally, vendor released several versions of firmware for this
series. It's not better to assign model-dependent information to
device entry according to the version field.
This commit fixes these bugs. The name of model is picked up correctly
and used to identify model-dependent information.
Cc: Stefan Richter <stefanr@s5r6.in-berlin.de>
Fixes: c0949b2785 ('ALSA: firewire-tascam: add skeleton for TASCAM FireWire series')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series has some LEDs on its surface. These LEDs can be
turned on/off by receiving asynchronous transactions to a certain
address. One of the LEDs is labels as 'FireWire'. It's better to light it
up when this driver starts to work. Besides, the LED for 'FireWire' is
turned off at bus reset.
This commit implements this idea.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commits, this driver got functionalities to transfer/receive
MIDI messages to/from TASCAM FireWire series.
This commit adds some ALSA MIDI ports to enable userspace applications
to use the functionalities.
I note that this commit doesn't support virtual MIDI ports which console
models support. A physical controls can be assigned to a certain MIDI
ports including physical and virtual. But the way is not clear.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series use asynchronous transaction to receive MIDI
messages. The transaction should be sent to a certain address.
This commit supports the outgoing MIDI messages. The messages in the
transaction includes some quirks:
* One MIDI message is transferred in one quadlet transaction, except for
system exclusives.
* MIDI running status is not allowed, thus transactions always include
status byte.
* The basic data format is the same as transferring MIDI messages
supported in previous commit.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series use asynchronous transaction to transfer MIDI
messages. The transaction is sent to a registered address.
This commit supports the incoming MIDI messages. The messages in the
transaction include some quirks:
* Two quadlets are used for one MIDI message and one timestamp.
* Usually, the first byte of the first quadlet includes MIDI port and MSB
4 bit of MIDI status. For system exclusive message, the first byte
includes MIDI port and 0x04, or 0x07 in the end of the message.
* The rest of the first quadlet includes MIDI bytes up to 3.
* Several set of MIDI messages and timestamp can be transferred in one
block transaction, up to 8 sets.
I note that TASCAM FireWire series ignores ID bytes of system exclusive
message. When receiving system exclusive messages with ID bytes on physical
MIDI bus, the series transfers the messages without ID bytes on IEEE 1394
bus, and vice versa.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commits, asynchronous transactions are supported for physical
controls. This commit adds a pair of MIDI ports for them.
This driver already adds diferrent number of ALSA MIDI ports for physical
MIDI ports, and the number of in/out ports are different. As seeing as
'amidi' program in alsa-utils package, a pair of in/out MIDI ports is
expected with the same name. Therefore, this commit adds a pair of new
ports to the first.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, asynchronous transaction for incoming MIDI messages
from physical controls is supported. The physical controls may be
controlled by receiving MIDI messages at a certain address.
This commit supports asynchronous transaction for this purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Digi 00x series has two types of model; rack and console. The console
models have physical controls. The model can transmit control messages.
These control messages are transferred by asynchronous transactions to
registered address.
This commit supports the asynchronous transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds MIDI functionality to capture/playback MIDI messages
from/to physical MIDI ports. These messages are transferred in isochronous
packets.
When no substreams request AMDTP streams to run, this driver starts the
streams at current sampling rate. When other substreams start at different
sampling rate, the streams are stopped temporarily, then start again at
requested sampling rate. This operation can generate missing MIDI bytes,
thus it's preferable to start PCM substreams at favorite sampling rate in
advance.
Digi 002/003 console also has a set of MIDI port for physical controls.
These ports are added in later commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In Digi 002/003 protocol, MIDI messages are transferred in the last data
channel of data blocks. Although this data channel has a label of 0x80,
it's not fully MIDI conformant data channel especially because the Counter
field always zero independently of included MIDI bytes. The 4th byte of
the data channel in LSB tells the number of included MIDI bytes. This byte
also includes the number of MIDI port. Therefore, the data format in this
data channel is:
* 1st: 0x80 as label
* 2nd: MIDI bytes
* 3rd: 0 or MIDI bytes
* 4th: the number of MIDI byte and the number of MIDI port
This commit adds support of MIDI messages in data block processing layer.
Like AM824 data format, this data channel has a capability to transfer
more MIDI messages than the capability of phisical MIDI bus. Therefore, a
throttle for data rate is required to prevent devices' internal buffer to
overflow.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Original code for 'DoubleOhThree' encoding was written with '__u8' type,
while the type is usually used to export something to userspace.
This commit replaces the type with 'u8'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables interrupt transfer mode for MIDI ports on newer
Boss/Roland devices such as the GT-100/001 which support interrupt
transfer on both IN and OUT MIDI endpoints. Previously this wasn't being
enabled for these devices as the code was specifically looking for the
scenario where the IN endpoint supported interrupt transfer and the OUT
endpoint was bulk transfer. Newer devices support interrupt transfer for
both endpoints.
This has been tested on Boss devices GT-001, BR-80 and JS-8 and Roland
VS-20.
It would benefit from some regresison testing with other devices if
possible.
Signed-off-by: Keith A. Milner <maillist@superlative.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In firewire-lib, isochronous packet streaming is stopped when detecting
wrong value for FMT field of CIP headers. Although this is appropriate
to IEC 61883-1 and 6, some BeBoB based devices with vendors' customization
use invalid value to FMT field of CIP headers in the beginning of
streaming.
$ journalctl
snd-bebob fw1.0: Detect unexpected protocol: 01000000 8000ffff
I got this log with M-Audio FireWire 1814. In this line, the value of FMT
field is 0x00, while it should be 0x10 in usual AMDTP.
Except for the beginning, these devices continue to transfer packets with
valid value for FMT field, except for the beginning. Therefore, in this
case, firewire-lib should continue to process packets. The former
implementation of firewire-lib performs it.
This commit loosens the handling of wrong value, to continue packet
processing in the case.
Fixes: 414ba022a5 ('ALSA: firewire-lib: add support arbitrary value for fmt/fdf fields in CIP header')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The structures of type snd_bebob_clock_spec, snd_bebob_rate_spec,
snd_bebob_meter_spec, and snd_bebob_spec are never modified after they are
initialized. Make them all const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This ensures that the link is not requesting any clock and the
PLL can turn off. The link is powered when controller is brought
out of reset.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On runtime pm resume, we need to download the firmware, also on
suspend we need to ensure all the interrupts from controller and
DSP are disabled.
Also since we download the firmware on resume, we don't need to do
so on init, so remove that bit
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Like we have in legacy mode HDA driver, we need to check the
status bit and handle interrupt only when it is not zero or all
bits set. We typically see the status as all 1's when controller
resumes from suspend, So add the check here as well and don't
handle for these cases.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Skylake driver will set the SPA bit to 0 to turn off the DSP core.
Driver will poll the Current Power Active (CPA) bit to match the
Set Power Active (SPA) bit value. When CPA bit matches the value
of SPA bit, the achieved power state has reached.
In case of DSP power down, register that was polled is SPA
instead of CPA. This patch corrects the register to be polled
in case of DSP power down.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, when asynchronous transactions finish in error state and
retries, work scheduling and work running also continues. This
should be canceled at fatal error because it can cause endless loop.
This commit enables to cancel transferring MIDI messages when transactions
encounter fatal errors. This is achieved by setting error state.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Typically, the target devices have internal buffer to adjust output of
received MIDI messages for MIDI serial bus, while the capacity of the
buffer is limited. IEEE 1394 transactions can transfer more MIDI messages
than MIDI serial bus can. This can cause buffer over flow in device side.
This commit adds throttle to limit MIDI data rate by counting intervals
between two MIDI messages. Usual MIDI messages consists of two or three
bytes. This requires 1.302 to 1.953 mili-seconds interval between these
messages. This commit uses kernel monotonic time service to calculate the
time of next transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, when two MIDI trigger callbacks can be called immediately,
transactions for the second MIDI messages can be postpone till next trigger
callback. This is not good for real-time message transmission.
This commit schedules work again at response handling callback if the
MIDI substream still includes untransferred MIDI messages.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, when waiting for a response, callers can start another
transaction by scheduling another work. This is not good for error
processing of transaction, especially the first response is too late.
This commit serialize request/response transactions, by adding one
boolean member to represent idling state.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some models receive MIDI messages via IEEE 1394 asynchronous transactions.
In this case, MIDI messages are transferred in fixed-length payload. It's
nice that firewire-lib module has common helper functions.
This commit implements this idea. Each driver adds
'struct snd_fw_async_midi_port' in its instance structure. In probing,
it should call snd_fw_async_midi_port_init() to initialize the
structure with some parameters such as target address, the length
of payload in a transaction and a pointer for callback function
to fill the payload buffer. At 'struct snd_rawmidi_ops.trigger()'
callback, it should call 'snd_fw_async_midi_port_run()' to start
transactions. Each driver should ensure that the lifetime of MIDI
substream continues till calling 'snd_fw_async_midi_port_finish()'.
The helper functions support retries to transferring MIDI messages when
transmission errors occur. When transactions are successful, the helper
functions call 'snd_rawmidi_transmit_ack()' internally to consume MIDI
bytes in the buffer. Therefore, Each driver is expected to use
'snd_rawmidi_transmit_peek()' to tell the number of bytes to transfer to
return value of 'fill' callback.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_seq_oss_readq_put_event() seems to be missing a memory barrier which
might cause the waker to not notice the waiter and miss sending a
wake_up as in the following figure.
snd_seq_oss_readq_put_event snd_seq_oss_readq_wait
------------------------------------------------------------------------
/* wait_event_interruptible_timeout */
/* __wait_event_interruptible_timeout */
/* ___wait_event */
for (;;) { prepare_to_wait_event(&wq, &__wait,
state);
spin_lock_irqsave(&q->lock, flags);
if (waitqueue_active(&q->midi_sleep))
/* The CPU might reorder the test for
the waitqueue up here, before
prior writes complete */
if ((q->qlen>0 || q->head==q->tail)
...
__ret = schedule_timeout(__ret)
if (q->qlen >= q->maxlen - 1) {
memcpy(&q->q[q->tail], ev, sizeof(*ev));
q->tail = (q->tail + 1) % q->maxlen;
q->qlen++;
------------------------------------------------------------------------
There are two other place in sound/core/seq/oss/ which have similar
code. The attached patch removes the call to waitqueue_active() leaving
just wake_up() behind. This fixes the problem because the call to
spin_lock_irqsave() in wake_up() will be an ACQUIRE operation.
I found this issue when I was looking through the linux source code
for places calling waitqueue_active() before wake_up*(), but without
preceding memory barriers, after sending a patch to fix a similar
issue in drivers/tty/n_tty.c (Details about the original issue can be
found here: https://lkml.org/lkml/2015/9/28/849).
Signed-off-by: Kosuke Tatsukawa <tatsu@ab.jp.nec.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we have introduced the core fns we should make hda use these
helpers
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current codec helpers are local to hda code and needs to be moved to
core so that other users can use it.
The helpers to read/write the codec and to check the
power state of widgets is copied
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We were getting build warning about "Section mismatch".
dmi_platform_intel_broadwell is being referenced from the probe function
rt5645_i2c_probe(), but dmi_platform_intel_broadwell was marked with
__initdata.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Reviewed-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a driver for the SPDIF transceiver available on RK3066, RK3188 and
RK3288. Heavily based on the rockchip i2s driver.
Signed-off-by: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
dev_info is too noisy for tplg wiget loading, so move it to
debug level
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Quite a few fixes here but they're all very small and driver specific,
none of them really stand out if you aren't using the relevant hardware
but they're all useful if you do happen to have an affected device.
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Merge tag 'asoc-fix-v4.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.3
Quite a few fixes here but they're all very small and driver specific,
none of them really stand out if you aren't using the relevant hardware
but they're all useful if you do happen to have an affected device.
Load and Initialize Non HDA Link Table in Skylake driver
to get platform configuration.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Initialize and creates DSP controls if processing pipe capability
is supported by HW. Updates the dma_id, hw_params to module param
to be used when DSP module has to be configured.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The SKL driver does not code DSP topology in driver. It uses the
newly added ASoC topology core to parse the topology information
(controls, widgets and map) from topology binary.
Each topology element passed private data which contains
information that driver used to identify the module instance
within firmware and send IPCs for that module to DSP firmware
along with parameters.
This patch adds init routine to invoke topology load and callback
for topology creation.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For FE and BE, the PCM parameters come from FE and BE hw_params
values passed. For a FE we convert the FE params to DSP expected
module format and pass to DSP. For a BE we need to find the
gateway settings (i2s/PDM) to be applied. These are queried from
NHLT table and applied.
Further for BE based on direction the settings are applied as
either source or destination parameters.
These helpers here allow the format to be calculated and queried
as per firmware format.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Skylake driver topology model tries to model the firmware
rule for pipeline and module creation.
The creation rule is:
- Create Pipe
- Add modules to Pipe
- Connect the modules (bind)
- Start the pipes
Similarly destroy rule is:
- Stop the pipe
- Disconnect it (unbind)
- Delete the pipe
In driver we use Mixer, as there will always be ONE mixer in a
pipeline to model a pipe. The modules in pipe are modelled as PGA
widgets. The DAPM sequencing rules (mixer and then PGA) are used
to create the sequence DSP expects as depicted above, and then
widget handlers for PMU and PMD events help in that.
This patch adds widget event handlers for PRE/POST PMU and
PRE/POST PMD event for mixer and pga modules. These event
handlers invoke pipeline creation, destroy, module creation,
module bind, unbind and pipeline bind unbind
Event handler sequencing is implement to target the DSP FW
sequence expectations to enable path from source to sink pipe for
Playback/Capture.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To configure a module, driver needs to send input and output PCM
params for a module in DSP. The FE PCM params come from hw_params
ie from user, for a BE they also come from hw_params but from
BE-link fixups.
So based on PCM params required driver has to find a converter
module (src/updown/format) and then do the conversion and
calculate PCM params in these pipelines In this patch we add the
helper modules which allow driver to do these calculations.
Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SKL driver needs to instantiate pipelines and modules in the DSP.
The topology in the DSP is modelled as DAPM graph with a PGA
representing a module instance and mixer representing a pipeline
for a group of modules along with the mixer itself.
Here we start adding building block for handling these. We add
resource checks (memory/compute) for pipelines, find the modules
in a pipeline, init modules in a pipe and lastly bind/unbind
modules in a pipe These will be used by pipe event handlers in
subsequent patches
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver now can make use of mclk data, if provided, to set, enable
and disable the clock source. As part of this, the choice to
enable clock squaring is dealt with as part of dai_sysclk() call
rather than as platform data.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for DT bindings in the codec driver.
As part of this support, the mclk data can now be provided and
used to control the mclk during codec operation.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull in the i915/hda changes for N/CTS setting so I can apply the
follow-up documentation work for drm/i915.
Some conflicts because ofc we had to rework i915 while that N/CTS work
was going on. But not more than adjacent changes really.
Signed-off-by: Daniel Vetter <daniel.vetter@intel.com>
Since commit 3d7608e4c1 ("ARM: shmobile: bockw: remove legacy
board file and config"), Renesas R-Car SoCs are only supported in
generic DT-only ARM multi-platform builds. The driver doesn't need to
use platform data anymore, hence remove platform data configuration.
Move <sound/rcar_snd.h> to sound/soc/sh/rcar/, as it's no longer needed
by platform code.
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
Should only try to enable/disable the provided mclk, during bias
level changes, if it's not NULL. Also return value of
clk_prepare_enable() should be checked and dealt with accordingly.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use of_match_ptr() to handle non-DT kernel scenario where match
table should be NULL.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Most of the boards have their headphone jack directly connected to the
matching pins of the SoCs. Since most of the time we will have the same
routing path, it makes no sense to put that in the DTS, since it will only
be some useless duplication there.
It also fixes the following warning messages that were seen so far, on
boards where we were using the bindings in the documentation example.
sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack
sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Left -> direct -> Headphone Jack
sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack
sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Right -> direct -> Headphone Jack
Reported-by: Priit Laes <plaes@plaes.org>
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/nau8825.c:1096:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
CC: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of hardconding a platform data for dw_dmac let's use it's own
autoconfiguration feature. Thus, remove hardcoded values.
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: Mark Brown <broonie@kernel.org>
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make a copy of memory for index_cache rather than directly use the
rt286_index_def to avoid run time error.
Fixes: c418a84a8c ("ASoC: Constify reg_default tables")
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AD1939 is missed from the table, so add it.
AD1936 and AD1937 are controlled by I2C interface, so remove them.
Fixes: e5224f58e3 ("ASoC: ad193x: add support to ad1934")
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The minimum volume level for the TAS2552 (control register value 0x00)
is -7dB however the driver declares it as -0.07dB.
Running amixer before the patch reports:
dBscale-min=-0.07dB,step=1.00dB,mute=0
Running amixer with the patch applied reports:
dBscale-min=-7.00dB,step=1.00dB,mute=0
Signed-off-by: Andreas Dannenberg <dannenberg@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Change the rockchip i2s object name (and thus module name) from the
rather generic snd-soc-i2s to the more specific snd-soc-rockchip-i2s
Signed-off-by: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
The AD1934 codec has no ADC feature. Hence it register mapping is slightly
different from the register mapping of other members of the AD193x family.
Some ASoC controls and widgets are related to the DAC feature so are not
relevant in the case of an AD1934 codec.
Signed-off-by: Cyrille Pitchen <cyrille.pitchen@atmel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, ESAI needs to save
all the values of registers before the system suspend and restore them after
the system resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power
source of modules during system suspend and resume procedure. Thus,
SSI needs to save all the values of registers before the system
suspend and restore them after the system resume.
The register SFCSR is volatile, but some bits in it need to be
recovered after suspend/resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, SPDIF needs to save
all the values of registers before the system suspend and restore them after
the system resume.
The SRPC register should be volatile, LOCK bit is set by the hardware.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, SAI needs to save
all the values of registers before the system suspend and restore them after
the system resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add snd_soc_pm_ops in machine driver to make the trigger suspend/resume
be called in suspend/resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
dw i2s controller can work in slave mode, codec being master.
dw i2s is made to support master/slave operation, by reading dwc
register.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
if the stream is decoupled and both link and host are used, while
releasing the stream, need to check if link and host stream are
not in use. This patch adds fix to check if the host/link stream
is in used before coupling it back when releasing the stream.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Bits in LOSIDV need to be set to map the stream id for specific link.
Fixing this by setting the required bits in the register.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently if bulk_enable() of supplies fails, the code still goes on
to try and put the device into active state, and set expected IO
voltage of the device. This doesn't really make sense so code now
returns on bulk_enable() failure, with an error message. Also,
to help with debug, failure to get supplies also provides an error
message.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When OSS emulation is loaded on ISA SB AWE32 chip, we get now kernel
warnings like:
WARNING: CPU: 0 PID: 2791 at fs/sysfs/dir.c:31 sysfs_warn_dup+0x51/0x80()
sysfs: cannot create duplicate filename '/devices/isa/sbawe.0/sound/card0/seq-oss-0-0'
It's because both emux synth and opl3 drivers try to register their
OSS device object with the same static index number 0. This hasn't
been a big problem until the recent rewrite of device management code
(that exposes sysfs at the same time), but it's been an obvious bug.
This patch works around it just by using a different index number of
emux synth object. There can be a more elegant way to fix, but it's
enough for now, as this code won't be touched so often, in anyway.
Reported-and-tested-by: Michael Shell <list1@michaelshell.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch adds the codec reset setting in the shutdown function to prevent
the weird sound of the headphone happened by rebooting.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The recent widget power saving introduced some unavoidable click
noises on old IDT 92HD73xx chips while it still seems working on the
compatible new chips. In the bugzilla, we tried lots of tests and
workarounds, but they didn't help much. So, let's disable the feature
for these specific chips as the least (but safest) fix.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=104981
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Its a bit odd that debugfs_create_bool() takes 'u32 *' as an argument,
when all it needs is a boolean pointer.
It would be better to update this API to make it accept 'bool *'
instead, as that will make it more consistent and often more convenient.
Over that bool takes just a byte.
That required updates to all user sites as well, in the same commit
updating the API. regmap core was also using
debugfs_{read|write}_file_bool(), directly and variable types were
updated for that to be bool as well.
Signed-off-by: Viresh Kumar <viresh.kumar@linaro.org>
Acked-by: Mark Brown <broonie@kernel.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The MacBookPro 12,1 has the same setup as the 11 for controlling the
status of the optical audio light. Simply apply the existing workaround
to the subsystem ID for the 12,1.
[sorted the fixup entry by tiwai]
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=105401
Signed-off-by: John Flatness <john@zerocrates.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Much like all the other Lenovo laptops, add a quirk to make
sound work with docking.
Reported-and-tested-by: lacknerflo@gmail.com
Signed-off-by: Laura Abbott <labbott@fedoraproject.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec supports 4 channel recording with TDM on AIF1.
This patch modifies the DAI capability to allow it.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds support for the DA7219 audio codec with built-in advanced
accessory detect features.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Designware I2S uses tx empty and rx available signals as the DMA
handshaking signals. during music playing, if XRUN occurs,
i2s_stop() function will be executed and both tx and rx irq are
masked, when music continues to be played, i2s_start() is executed
but both tx and rx irq are not unmasked which cause I2S stop
sending DMA handshaking signal to DMA controller, and it finally
causes music playing will be stopped once XRUN occurs for the first
time.
[On list discussion suggests this may be partly a race condition on slow
systems -- broonie]
Signed-off-by: Yitian Bu <yitian.bu@tangramtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The struct snd_soc_tplg_pcm_dai is renamed to snd_soc_tplg_pcm.
This struct will now be used to handle data related to PCMs
(FE DAI & DAI links). It's not for BE, because BE DAI mappings will be
provided by ACPI/FDT data.
Remove the unused struct snd_soc_tplg_pcm_cfg_caps. We are using
snd_soc_tplg_stream and snd_soc_stream_caps instead.
Bump ABI version to 4.
Signed-off-by: Vedang Patel <vedang.patel@intel.com>
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adds convenience defines for declaring a gain control that
has an input mux. These blocks are functionally equivalent to
the existing mixer blocks but can only have a single input
active at once.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai