Sound data needs to be sent to R-Car sound SSI when playback.
But, there are 2 interfaces for it.
1st is SSITDR/SSIRDR which are mapped on SSI.
2nd is SSIn_BUSIF which are mapped on SSIU.
2nd SSIn_BUSIF is used when DMA transfer,
and it is always used if sound data came from via SRC.
But, we can use it when SSI+DMA case too.
(Current driver is assuming 1st SSITDR/SSIRDR for it)
2nd SSIn_BUSIF can be used as FIFO.
This is very helpful/useful for SSI+DMA.
But DMA address / DMA ID are not same between 1st/2nd cases.
This patch care about these settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current FSI driver is using DMAEngine directly,
but, ASoC is requesting to use common DMA transfer method,
like snd_dmaengine_pcm_trigger() or dmaengine_pcm_ops.
It is difficult to switch at this point, since Renesas
driver is also supporting PIO transfer.
This patch uses dmaengine_prep_dma_cyclic() instead
of dmaengine_prep_slave_single().
It is used in requested method,
and is good first step to switch over.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
fsi PIO/DMA handler are using each own pointer update method,
but these can be share.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current fsi driver is using SNDRV_DMA_TYPE_CONTINUOUS
for snd_pcm_lib_preallocate_pages_for_all().
But, it came from original dma-sh7760.c,
and no longer needed.
This patch exchange its parameter, and removed
original dma mapping and un-needed
dma_sync_single_xxx() from driver.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Avoid creating duplicate directories by prefixing codecs and platforms
with their separate identifiers. This avoids snd-soc-dummy (which can
appear both as a dummy platform and a dummy codec on the same card)
from clashing.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Permit ALSA to run without hardware interrupts from the audio interface.
Instead, ALSA will use a kernel timer to decide when to check the buffer
state, resulting in a lighter workload for the CPU.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is no hardware restriction requiring a minimum of 8 periods, or
a minimum of 2048 bytes in a period. Let's drop these values so that
userspace has more flexibility in choosing these parameters.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
We still see the occasional timeout waiting for busy to clear. As the
spec is contradictory, and we know that the current implementation
doesn't work, try an alternative interpretation from the spec. This
one appears to work - I have yet to find any issue with it during my
testing over several months.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
The spec requires that the mute bits must be set while the channel
is disabled. Ensure that this is the case by providing a helper
which ensures that the appropriate mute bit is set while the enable
bit is clear.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since we wish to disable capture inputs for some formats, we need to
ensure that we clear the enable bits in our cached record control
register. This seems to have been missed, resulting in the register
only accumulating enable bits.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a KIRKWOOD_RECCTL_ENABLE_MASK definition to complement the existing
PLAYCTL definition, and make use of it where we wish to clear both
enable bits.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Also add const to array
text data bss dec hex filename
10946 2904 3528 17378 43e2 sound/soc/codecs/wm0010.o-before
10891 2840 3512 17243 435b sound/soc/codecs/wm0010.o-after
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@linaro.org>
There are a bunch of users open coding the for_each_node_by_name() by
calling of_find_node_by_name() directly instead of using the macro. This
is getting in the way of some cleanups, and the possibility of removing
of_find_node_by_name() entirely. Clean it up so that all the users are
consistent.
Signed-off-by: Grant Likely <grant.likely@linaro.org>
Cc: Rob Herring <robh+dt@kernel.org>
Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Cc: "Rafael J. Wysocki" <rjw@rjwysocki.net>
Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Cc: Takashi Iwai <tiwai@suse.de>
ICH6_ prefix doesn't mean that it's specific to ICH6 chipset but
rather its generic for all HD-audio (or "Azalia") devices.
Use AZX_ prefix instead to align with other constants.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just move struct fields between struct azx and struct hda_intel, and
move some definitions from hda_priv.h to hda_intel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... and move most parts into hda_intel.c from the generic controller
code. This is a clean up, and there should be no functional change by
this patch.
Now, struct azx obtains the generic callbacks for getting the position
and the delay. As default NULL, posbuf is read. These replace the
old position_fix[], and each is implemented as a callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For Intel Haswell/Broadwell display HD-A controller, the 24MHz HD-A link BCLK
is converted from Core Display Clock (CDCLK): BCLK = CDCLK * M / N
And there are two registers EM4 and EM5 to program M, N value respectively.
The EM4/EM5 values will be lost and when the display power well is disabled.
BIOS programs CDCLK selected by OEM and EM4/EM5, but BIOS has no idea about
display power well on/off at runtime. So the M/N can be wrong if non-default
CDCLK is used when the audio controller resumes, which results in an invalid
BCLK and abnormal audio playback rate. So this patch saves and restores valid
M/N values on controller suspend/resume.
And 'struct hda_intel' is defined to contain standard HD-A 'struct azx' and
Intel specific fields, as Takashi suggested.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a USB-audio device is disconnected while PCM is still running, we
still see some race: the disconnect callback calls
snd_usb_endpoint_free() that calls release_urbs() and then kfree()
while a PCM stream would be closed at the same time and calls
stop_endpoints() that leads to wait_clear_urbs(). That is, the EP
object might be deallocated while a PCM stream is syncing with
wait_clear_urbs() with the same EP.
Basically calling multiple wait_clear_urbs() would work fine, also
calling wait_clear_urbs() and release_urbs() would work, too, as
wait_clear_urbs() just reads some fields in ep. The problem is the
succeeding kfree() in snd_pcm_endpoint_free().
This patch moves out the EP deallocation into the later point, the
destructor callback. At this stage, all PCMs must have been already
closed, so it's safe to free the objects.
Reported-by: Alan Stern <stern@rowland.harvard.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP Spectre 13 has the IDT 92HD95 codec, and BIOS seems to set the
default high-pass filter in some "safer" range, which results in the
very soft tone from the built-in speakers in contrast to Windows.
Also, the mute LED control is missing, since 92HD95 codec still has no
HP-specific fixups for GPIO setups.
This patch adds these missing features: the HPF is adjusted by the
vendor-specific verb, and the LED is set up from a DMI string (but
with the default polarity = 0 assumption due to the incomplete BIOS on
the given machine).
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=74841
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The static quirk code has been disabled for a while and it seems
working fine, so it's time to actually get rid of it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The static quirk code has been disabled for a while and it seems
working fine, so it's time to actually get rid of it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pass the codec object so that we can replace all the rest of
snd_print*() usages with the proper device-specific print helpers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A struct member variable is set to the same value more than once
This was found using a static code analysis program called cppcheck.
Signed-off-by: Rickard Strandqvist <rickard_strandqvist@spectrumdigital.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kernel should not enable/disable speakers and digital microphone whenever
jack is inserted/removed. This is more use-case than kernel specific
decision. For instance one may want to play VoIP ring tones using both
speakers and headphone but play music only from one of them.
Because of above reason remove "Ext Spk" and "Int Mic" update when jack
state is changed. Also this update was illogical anyway: "Ext Spk" was
enabled when jack was inserted and disabled when jack was removed.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
