IF YOU WOULD LIKE TO GET AN ACCOUNT, please write an
email to Administrator. User accounts are meant only to access repo
and report issues and/or generate pull requests.
This is a purpose-specific Git hosting for
BaseALT
projects. Thank you for your understanding!
Только зарегистрированные пользователи имеют доступ к сервису!
Для получения аккаунта, обратитесь к администратору.
In a rare combination of Kconfig settings, the 88pm860x-codec
module may be selected as a loadable module, while it's also being
used by the ttb-dkb code that is built-in, resulting in a link
error:
sound/built-in.o: In function `ttc_pm860x_init':
:(.text+0x3e888): undefined reference to `pm860x_hs_jack_detect'
:(.text+0x3e898): undefined reference to `pm860x_mic_jack_detect'
Changing ttb-tkb to a tristate option tells Kconfig that 88pm86x
actually needs to be built-in if ttc-dkb is also built-in.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This was overlooked in the late change to remove the I2S padding bits
from S24_LE mode. The patch also limits S32_LE mode to 384kHz, the
maximum according to the datasheets.
Signed-off-by: Peter Rosin <peda@axentia.se>
Signed-off-by: Mark Brown <broonie@kernel.org>
The rt5677 codec has gained code that requires SPI to work correctly,
but there is no provision in Kconfig to prevent the driver from
being used when SPI is disabled or a loadable module, resulting
in this build error:
sound/built-in.o: In function `rt5677_spi_write':
:(.text+0xa7ba0): undefined reference to `spi_sync'
sound/built-in.o: In function `rt5677_spi_driver_init':
:(.init.text+0x253c): undefined reference to `spi_register_driver'
ERROR: "spi_sync" [sound/soc/codecs/snd-soc-rt5677-spi.ko] undefined!
ERROR: "spi_register_driver" [sound/soc/codecs/snd-soc-rt5677-spi.ko] undefined!
This makes the SPI portion of the driver depend on the SPI subsystem,
and disables the function that uses SPI for firmware download if SPI
is disabled. The latter may not be the correct solution, but I could
not come up with a better one.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Fixes: af48f1d08a54741 ("ASoC: rt5677: Support DSP function for VAD application")
Signed-off-by: Mark Brown <broonie@kernel.org>
An earlier bug fix of mine made the SND_DM365_VOICE_CODEC symbol
tristate to avoid creating an undefined reference from the
davinci-vcif.c driver to the davinci_soc_platform_register
function that may be in a module.
However, this may now lead to a different error on randconfig
kernels:
"warning: SND_DM365_VOICE_CODEC creates inconsistent choice state"
This happens because we now have a choice statement with
one bool and one tristate option, and the latter might not
support being set to 'y' because of dependencies.
This new change turns the other option into 'tristate' as well,
which avoids the problem.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Fixes: 19926c6de0c3 ("ASoC: davinci: vcif must be a module if SND_DAVINCI_SOC is")
Signed-off-by: Mark Brown <broonie@kernel.org>
Protect the call with a mutex, as this may be called in parallel
(either from the PCM rate change and the clock change).
Acked-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Define snd_ak4114_suspend() and snd_ak4114_resume() functions to
handle PM properly, stopping and restarting the work at PM.
Currently only ice1712/juli.c deals with the PM and ak4114, so fix the
calls there appropriately.
The same PM functions are defined in ak4113.c, too, although they
aren't currently called yet (ice1712/quartet.c may be enhanced to
support PM later).
Acked-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... just to follow the standard coding style.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When ak4114 work calls its callback and the callback invokes
ak4114_reinit(), it stalls due to flush_delayed_work(). For avoiding
this, control the reentrance by introducing a refcount. Also
flush_delayed_work() is replaced with cancel_delayed_work_sync().
The exactly same bug is present in ak4113.c and fixed as well.
Reported-by: Pavel Hofman <pavel.hofman@ivitera.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of them are rather relevant with the definitions in driver.h,
and there are only a few lines, so just rip it off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just reformatting the comments and typos fixed, no functional
changes. Particularly,
- avoid the kerneldoc marker "/**",
- reduce multiple comment lines into single lines,
- corrected wrongly referred function names
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8904 and WM8918 has the same data type, while the WM8912
has different data type. So, use the data in dt ids table to
distinguish them.
Signed-off-by: Alexander Morozov <linux@meltdown.ru>
[voice.shen@atmel.com: add code to distinguish device type]
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The PLL introduces jitter, which in turn introduces noice if used
to clock the DAC. Thus, avoid the PLL output, and use the PLL input
to drive the DAC clock, if possible.
This is described for the PCM5142/PCM5242 chips in the answers to the
forum post "PCM5142/PCM5242 DAC clock source" at the TI E2E community
pages (1).
(1) http://e2e.ti.com/support/data_converters/audio_converters/f/64/t/389994
Signed-off-by: Peter Rosin <peda@axentia.se>
Signed-off-by: Mark Brown <broonie@kernel.org>
Using the PLL in master mode requires using an external connection
between one of the GPIO pins (configured as PLL/4 output) and the
SCK pin. It also requires the external clock to be fed to some other
GPIO pin instead of the SCK pin.
This is described for the PCM5122 chip in the answers to the forum post
"PCM5122 DAC as I2S master troubles with PLL mode" at the TI E2E
community pages (1). The clocking functionality is also much better
described in the datasheet for the chip PCM5242, which seems to be
register compatible with PCM512x and PCM514x (which both have severely
lacking datasheets).
(1) http://e2e.ti.com/support/data_converters/audio_converters/f/64/t/267830
Signed-off-by: Peter Rosin <peda@axentia.se>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use register field names from the seemingly compatible PCM5242 datasheet,
as the PCM512x and PCM514x datasheets are severly lacking.
Signed-off-by: Peter Rosin <peda@axentia.se>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper functions to allow drivers to specify several disjoint
ranges for a variable. In particular, there is a codec (PCM512x) that
has a hole in its supported range of rates, due to PLL and divider
restrictions.
This is like snd_pcm_hw_constraint_list(), but for ranges instead of
points.
Signed-off-by: Peter Rosin <peda@axentia.se>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
of_match_ptr is already conditionally compiled based on
CONFIG_OF so further conditional compilation is not
required. Remove conditional compilation surrounding
of_match_ptr.
Signed-off-by: Andrew Jackson <Andrew.Jackson@arm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The user-space API definition for usb_stream stuff should be moved
to include/uapi/sound to be exposed publicly.
While we're at it, add the missing ifdef guard for double inclusion,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The soundscape driver uses the ISA inb/outb functions declared
in linux/io.h, so it needs to include this header to avoid
a build error:
sscape.c: In function 'sscape_write_unsafe':
sscape.c:203:2: error: implicit declaration of function 'outb' [-Werror=implicit-function-declaration]
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We never set the ->scratch pointer, so let's delete it.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fix warning while "make xmldocs".
Warning(.//sound/soc/soc-devres.c:70): No description
found for parameter 'platform_drv'
Warning(.//sound/soc/soc-devres.c:70): Excess function
parameter 'platform' description in 'devm_snd_soc_register_platform'
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Both playback and capture callbacks are identical, so let's merge
them.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code deals with the stream start / stop solely via
line6_pcm_acquire() and line6_pcm_release(). This was (supposedly)
intended to avoid the races, but it doesn't work as expected. The
concurrent acquire and release calls can be performed without proper
protections, thus this might result in memory corruption.
Furthermore, we can't take a mutex to protect the whole function
because it can be called from the PCM trigger callback that is an
atomic context. Also spinlock isn't appropriate because the function
allocates with kmalloc with GFP_KERNEL. That is, these function just
lead to singular problems.
This is an attempt to reduce the existing races. First off, separate
both the stream buffer management and the stream URB management. The
former is protected via a newly introduced state_mutex while the
latter is protected via each line6_pcm_stream lock.
Secondly, the stream state are now managed in opened and running bit
flags of each line6_pcm_stream. Not only this a bit clearer than
previous combined bit flags, this also gives a better abstraction.
These rewrites allows us to make common hw_params and hw_free
callbacks for both playback and capture directions.
For the monitor and impulse operations, still line6_pcm_acquire() and
line6_pcm_release() are used. They call internally the corresponding
functions for both playback and capture streams with proper lock or
mutex. Unlike the previous versions, these function don't take the
bit masks but the only single type value. Also they are supposed to
be applied only as duplex operations.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clearing prev_fsize in line6_pcm_acquire() is pretty racy.
This can be called at any time while the stream is being played.
Rather better to clear prev_fbuf and prev_fsize at the proper place
like the stream stop for capture, and just after copying the monitor /
impulse data inside the spinlock.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The impulse and monitor handling in submit_audio_out_urb() isn't
protected thus this can be racy with the capture stream handling.
This patch extends the range to protect via each stream's spinlock
(now the whole submit_audio_*_urb() are covered), and take the capture
stream lock additionally for the impulse and monitor handling part.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the check of multi configurations before snd_card_new() as a
short path, and reduce superfluous pointer references.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of allocating the private data individually in each driver's
probe at first, let snd_card_new() allocate the data that is called in
line6_probe(). This simplifies the primary probe functions.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The interface argument is used just for retrieving the assigned
device, which can be already found in line6->ifcdev. Drop them from
the callbacks. Also, pass the usb id to private_init so that the
driver can deal with it there. This is a preliminary work for the
further cleanup to move the whole allocation into driver.c.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A minor optimization; while pausing, the driver just copies the zero
that doesn't need any volume changes.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM stream buffer allocation and free are identical for both
playback and capture streams. Provide single helper functions.
These are used only in pcm.c, thus they can be even static.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codes to unlink and sync URBs are identical for both playback and
capture streams. Consolidate to single helper functions.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new line6_pcm_stream structure and group individual
fields of snd_line6_pcm struct to playback and capture groups.
This patch itself just does rename and nothing else. More
meaningful cleanups based on these fields shuffling will follow.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the problem still really remains, we should fix it instead of
papering over it like this...
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>