Current R-Car sound SSI/SRC/DVC selection has feature limit.
(It is assuming that SSI/SRC are using same index number)
So that enabling SSI/SRC flexible selection,
this patch modifies DMA settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Refactor the function to facilitate the migration to
multiple codecs.
Fix a trailing space in the header as well.
No functional change.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since multiple codecs DAI will be usable in the future, remove
explicit unique codec_dai and cpu_dai parameters.
Replace them with snd_soc_pcm_runtime pointer that will contain
every instances.
No functionale change.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
'status' is not used in the function. Remove it.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Tested-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "list" field has been omitted from struct azx, but its
initialization remained mistakenly in hda_tegra.c, which leads to a
compile error:
sound/pci/hda/hda_tegra.c: In function 'hda_tegra_create':
sound/pci/hda/hda_tegra.c:481:22: error: 'struct azx' has no member
named 'list'
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Fixes: 9a34af4a33 ('ALSA: hda - Move more PCI-controller-specific stuff from generic code')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'private_data' is not used in the function. Remove it.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I received a report this morning from one of the Novena developers that
the behaviour of the iMX6 ASoC codec driver (using imx-pcm-dma.c) was
sub-optimal under high system load.
While there are issues relating to system load remaining, upon reviewing
the ASoC imx-pcm-dma.c driver, it was noticed that it not using the
residue support, because SDMA doesn't support it. This has the effect
that SDMA has to make multiple calls into the ASoC and ALSA code, one
for each period.
Since ALSA's snd_pcm_elapsed() does not need to be called multiple times
and it is entirely sufficient to call it once to update ALSA with the
current buffer position via the pointer method, we can do better here.
We can also avoid stopping the DMA entirely, just like real cyclic DMA
implementations behave. While this means that we replay some old samples,
this is a nicer behaviour than having audio stop and restart.
The changes to achieve this are relatively minor - imx-sdma.c can track
where the DMA is to the nearest descriptor boundary - it does this
already when deciding how many callbacks to issue. In doing this,
buf_tail always points at the descriptor which will complete next.
The residue is defined by the bytes remaining to the end of the buffer,
when the buffer is viewed as a single block of memory [start...end].
So, when we start out, there's a full buffer worth of residue, and this
counts down as we approach the end of the buffer, eventually becoming
zero at the end, before returning to the full buffer worth when we
wrap back to the start.
Moving the walking of the descriptors into the interrupt handler means
that we can update the BD_DONE flag at interrupt time, thus avoiding
a delayed tasklet stopping the cyclic DMA.
This means that the residue can be calculated from (total descriptors -
buf_tail) * descriptor size. This is what the change below does. We
update imx-pcm-dma.c to remove the NO_RESIDUE flag since we now provide
the residue.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Peach-pi board has MAX98091 CODEC. Extend snow machine driver to support
this board.
Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Reviewed-by: Doug Anderson <dianders@chromium.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The MAX98091 CODEC is the same as MAX98090 CODEC, but with an extra
microphone. Existing driver for MAX98090 CODEC already has support
for MAX98091 CODEC. Adding proper compatible string so that MAX98091
CODEC can be specified from device tree.
Signed-off-by: Wonjoon Lee <woojoo.lee@samsung.com>
Signed-off-by: Doug Anderson <dianders@chromium.org>
Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The implementation of rt5677_pll_calc() has the same logic of rl6231_pll_calc().
The only difference is the lower boundary checking for freq_in.
This patch calls rl6231_pll_calc() instead of open-coded.
The k_bp of struct rt5677_pll_code is always false, thus also remove the
code to check pll_code.k_bp.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In case of S24_LE/U24_LE modes we expect 24bits on the bus while the samples
are stored and transferred in memory on 32bits (lower 3 bytes of the 4
bytes).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The codec need to be configured to 24bit mode in case of S24_LE format.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Correct the hw_params callback to configure the codec correctly in case of
S24_3LE format since in case of S24_3LE the codec has been configured to
16bit format mode.
S24_LE is not defined as supported format for the codec.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The recent commit [6194b99d: ALSA: hda - Kill the rest of snd_print*()
usages] changed the callback map_slaves(), but one call was forgotten
to be replaced due to the cast, which leads to kernel Oops due to
invalid function. This patch replaces it with a proper function.
Fixes: 6194b99de9 ('ALSA: hda - Kill the rest of snd_print*() usages')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cs4265_volatile_register reutrns a bool. The function now returns
true or false vs 1 and 0.
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current DVC can be enabled only when playback,
but, this came from misunderstanding.
It is not correct.
DVC <-> DMA relationship is...
Playback: MEM -> DMAC -> SRC -> DVC -> DMACp -> SSI
Capture: SSI -> DMACp -> SRC -> DVC -> DMAC -> MEM
DVC can be used for both Playback/Capture
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current R-Car sound 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>
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>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The similar fixup as T440 is needed for supporting the dock on T540.
Reported-by: Jim Minter <jminter@redhat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another quirk to make the headset mic work on some new Dell machines.
Cc: Hui Wang <hui.wang@canonical.com>
BugLink: https://bugs.launchpad.net/bugs/1297581
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: 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>
It turned out there is no need to enable microphone detection in MAX98090
codec. Headset microphone is anyway detected by a GPIO signal from another
chip and headset button presses cannot be detected either because a signal
needed for it is not connected.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Pass actual jack type bitmask to snd_soc_jack_new() in order to report
also microphone detections and not only headphone. While at it change also
jack name and pass also SND_JACK_LINEOUT type.
Reported-by: Jin Yao <yao.jin@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Headset jack has only mono microphone input.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Mic detect GPIO is active low when headset microphone is detected. Found
both by debugging and checking the schematics.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move "MICBIAS" as a supply widget to "Headset Mic" instead of keeping it
between input pin "IN34" and "Headset Mic".
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Cirrus Logic CS4265 Stereo I2C CODEC.
Signed-off-by: Paul Handrigan <paul.handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is cosmetical - it makes the pin quirk table look better.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is cosmetical - it makes the new pin quirk table look better.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a missed break in a case statement
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>
Two bug reporters with Dell XPS 15 report that they need to use the
dell-headset-multi model to get the headset mic working.
The two bug reporters have different PCI SSID (1028:05fd and 1028:05fe)
but this pin quirk matches both.
BugLink: https://bugs.launchpad.net/bugs/1331915
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use clk_prepare_enable/clk_disable_unprepare to make the driver
work properly with common clock framework.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use clk_prepare_enable/clk_disable_unprepare to make the driver
work properly with common clock framework.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use dmaengine instead of legacy s3c24xx DMA API for s3c24xx and s3c2412
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a newline and, while at it, remove a space and redundant braces.
Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As added in previosu patch along with stream to piep conversion si required for
compressed audio too
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Merrifield DSP used various pipelines to identify the streams and processing
modules. Add these defination in the pcm driver and also add a table for device
entries to firmware pipeline id conversion
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We need to call the proper init function in case it has been
overridden, as it might restore things that the generic routing
doesn't know anything about. E.g. AMD cards have special verbs
that need resetting.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=77901
Fixes: 5a61358433 ('ALSA: hda - hdmi: Add ATI/AMD multi-channel audio support')
Signed-off-by: Pierre Ossman <pierre@ossman.eu>
Cc: <stable@vger.kernel.org> [v3.13+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A recent refactoring broke the possibility to manually specify
model name as a module parameter. This patch restores the desired
functionality.
Fixes: c21c8cf77f ('ALSA: hda - Add fixup_forced flag')
Reported-by: Kent Baxley <kent.baxley@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP mute led support output mute led and input mute led.
ALC280:
GPIO3 to control output mute led.
Mic1 vref to control input mute led.
ALC282:
Line1 vref to control output mute led.
Mic1 vref to control input mute led.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed coding style issues of "Missing Blank line after declaration"
Signed-off-by: Shahina Shaik <sharab.shaik@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fixed Coding style issues of lines over 80 characters.
Signed-off-by: Shahina Shaik <sharab.shaik@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fixed a brace coding style issue in the tlv320aic32x4.c
Signed-off-by: Shahina Shaik <sharab.shaik@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Tis will be used to add table based support for pcm front ends in subsequent
patches
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This will be used to update current driver as well as in support for the mrfld
patches
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit 31ee2bfd72 ("ASoC: fsl: select SND_SOC_IMX_PCM_DMA
where needed") started selecting SND_SOC_IMX_PCM_DMA and
SND_SOC_IMX_PCM_FIQ for two drivers when building for i.MX.
This has turned out too aggressive, as FIQ is only available
for i.mx2 through i.mx5, but not i.mx6 or vybrid.
Further, two more drivers have become user-selectable in the
meantime, and they both depend on DMA for the imx platform
as well.
This changes the selection of FIQ to depend on the TZIC or
AVIC interrupt controllers that actually export the imx
specific FIQ interfaces, and adds the missing select statements
for SAI and ESAI.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Both codecs used by snow, max98090 and max98095 require the use
of I2C, so we can only select this driver if I2C is there, otherwise
we get a build error like:
codecs/max98090.c:2494:1: warning: data definition has no type or storage class [enabled by default]
module_i2c_driver(max98090_i2c_driver);
^
codecs/max98095.c:2443:1: warning: data definition has no type or storage class [enabled by default]
module_i2c_driver(max98095_i2c_driver);
^
This adds one more I2C dependency to the hundreds we already
have.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The platform field in the snd_soc_dapm_widget and snd_soc_dapm_context structs
is now unused can be removed. New code that wants to get the platform for a
widget or dapm context should use snd_soc_dapm_to_platform(w->dapm) or
snd_soc_dapm_to_platform(dapm).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DAI DAPM context was added in commit be09ad90 ("ASoC: core: Add platform DAI
widget mapping") and the only user was removed again in commit ae10e7e8f ("ASoC:
core: Only add platform DAI widgets once."). Now that we have a per component
DAPM context it is unlikely that we'll need the DAI DAPM context again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds stream_event() and seq_notifier() callbacks similar to those
found in the snd_soc_codec_driver and snd_soc_platform driver struct to the
snd_soc_component_driver struct. This is meant to unify the handling of these
callbacks across different types of components and will eventually allow their
removal from the CODEC and platfrom driver structs.
The new callbacks are slightly different from the old ones in that they take a
snd_soc_component as a parameter rather than a snd_soc_dapm_context. This was
done since otherwise casting from the DAPM context to the component would
typically be the first thing to do in the callback. And the interface becomes
slightly cleaner by passing a snd_soc_component to all callbacks in the
snd_soc_component_driver struct.
The patch also already removes the stream_event() callback from the
snd_soc_codec_driver and snd_soc_platform_driver structs as it is currently
unused.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The snd_soc_platform dapm field is not accessed outside of the ASoC core. Switch
it over to using the snd_soc_component DAPM context.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds full DAPM support at the component level. Previously there was
only full DAPM support for CODECs and partial DAPM support (e.g. no Mixers nor
MUXs) for platforms. Having DAPM support at the component level will allow all
types of components to use DAPM and also help in consolidating the DAPM support
between CODECs and platforms.
Since the DAPM context is directly embedded into the snd_soc_codec and
snd_soc_platform struct and the 'dapm' field is directly referenced in a lot of
drivers moving the field just right now is not possible without causing code
churn. The approach this patch takes is to add two new fields to the component
struct. One field which is the pointer to the actual DAPM context used by the
component and one DAPM context that will be used as the default if no other
context was specified. For CODECs and platforms the pointer is initialized to
point to the CODEC or platform DAPM context. All generic code when referencing
a component's DAPM struct will go via the pointer. This will make it possible to
eventually seamlessly move the DAPM context from snd_soc_codec and
snd_soc_platform struct over once all direct references have been eliminated.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently the DAPM code directly looks at the CODEC driver struct to get a
handle to the set_bias_level() callback. This patch adds a new set_bias_level()
callback to the DAPM context struct. The DAPM code will use this new callback
instead of the CODEC callback. For CODECs the new callback is set up to call the
CODEC specific set_bias_level callback(). Not looking directly at the CODEC
driver struct will allow non CODEC DAPM contexts to implement a set_bias_level()
callback.
This is also similar to how the seq_notifier() and stream_event() callbacks are
currently handled.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently only pins in CODEC DAPM contexts are automatically marked as
non-connected if the card has the fully_routed flag set. This makes sense since
widgets which qualify for auto-disconnection are only found in CODEC DAPM
contexts. But with componentisation this is going to change, so consider all
widgets for auto-disconnection.
Also it is probably faster to walk the widgets list only once rather than once
for each CODEC.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Split snd_soc_component_register() into snd_soc_component_initialize() and
snd_soc_component_add(). Using a 2-stage registration approach has the advantage
that it is possible to modify the component after it has been initialized, but
before it is made visible to the system. This e.g. allows CODECs or platforms to
overwrite some of the default settings made in snd_soc_component_initialize().
Similar snd_soc_component_unregister() is split into two steps as well,
snd_soc_component_delete(), which removes the component from the system, and
snd_soc_component_cleanup(), which frees all the resources allocated by the
component.
Furthermore this patch makes sure that if a component is visible on two list
(e.g. the component list and the CODEC list) it is added or removed to both
lists atomically.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The component struct already has a name and id field which are initialized to
the same values as the same fields in the CODEC and platform structs. So remove
them from the CODEC and platform structs and used the ones from the component
struct instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move the name_prefix from the CODEC struct to the component struct. This will
eventually allow to specify prefixes for all types of components. It is also
necessary to make the DAPM code component type independent (i.e. a DAPM context
does not need to know whether it belongs to a CODEC or a platform or something
else).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
One-bit signed bitfields have two possible values: 0 and -1. This sometimes
leads to unexpected results (e.g. foo.bar = 1; foo.bar == 1 => false) which is
why it is recommended to make one-bit bitfields unsigned.
This fixes the following sparse warnings:
sound/soc/sh/fsi.c:267:25: error: dubious one-bit signed bitfield
sound/soc/sh/fsi.c:268:22: error: dubious one-bit signed bitfield
sound/soc/sh/fsi.c:269:20: error: dubious one-bit signed bitfield
sound/soc/sh/fsi.c:270:28: error: dubious one-bit signed bitfield
sound/soc/sh/fsi.c:271:26: error: dubious one-bit signed bitfield
sound/soc/sh/fsi.c:272:25: error: dubious one-bit signed bitfield
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
dmaengine_prep_slave_single() expects a enum dma_transfer_direction and not a
enum dma_data_direction. Since the integer representations of both DMA_TO_DEVICE
and DMA_MEM_TO_DEV aswell as DMA_FROM_DEVICE and DMA_DEV_TO_MEM have the same
value the code worked fine even though it was using the wrong type.
Fixes the following warning from sparse:
sound/soc/sh/rcar/core.c:227:49: warning: mixing different enum types
sound/soc/sh/rcar/core.c:227:49: int enum dma_data_direction versus
sound/soc/sh/rcar/core.c:227:49: int enum dma_transfer_direction
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
omap_pcm_platform_register() is declared in omap-pcm.h and defined in
omap-pcm.c. To make sure that the function signature matches for both omap-pcm.c
should include omap-pcm.h
Fixes the following warning from sparse:
sound/soc/omap/omap-pcm.c:235:5: warning: symbol
'omap_pcm_platform_register' was not declared. Should it be static?
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The cs42xx8_of_match table is not used outside of the driver, hence it can and
should be made static.
Fixes the following warning from sparse:
sound/soc/codecs/cs42xx8.c:425:27: warning: symbol 'cs42xx8_of_match' was
not declared. Should it be static?
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
SOC_ENUM_SINGLE_DECL() already adds the const qualifier, so there is no need to
manually specify it.
Fixes the following warnings from sparse:
sound/soc/codecs/tlv320aic31xx.c:253:1: warning: duplicate const
sound/soc/codecs/tlv320aic31xx.c:255:1: warning: duplicate const
sound/soc/codecs/tlv320aic31xx.c:257:1: warning: duplicate const
sound/soc/codecs/tlv320aic31xx.c:260:1: warning: duplicate const
sound/soc/codecs/tlv320aic31xx.c:262:1: warning: duplicate const
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now the DSP is capable of reporting the delay, report it to upper layers
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The argument was called mad_substream which is no longer apt as older driver
is not used anymore so rename as arg
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This helps us to handle pcm and compress ops seperately and per dai
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Connect "Headset Mic" to "MICBIAS1" supply widget of RT5640 in order to
enable bias voltage for headset microphones.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
DSP initialization complete message IPC_IA_FW_INIT_CMPLT is a large message
carrying firmware details in mailbox. Read and show those details during
init in order to be able to get that information to QA reports.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When SSC work as master, it will generate the frame sync signal.
On old SoCs, it only supports frame sync length less or equal to
16bits, on newer SoCs, it supports frame sync length extension,
which can support frame size larger than 16 bits.
So, add this to make it supports playback 24/32 bits audio clips.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
On the wm5110 CODEC both the left and right channel must be powered
when an output is being used as a mono output, although no audio is
routed to the right output channel. This patch adds additional DAPM
routes to link the right channel to the left in the case where an output
is marked as mono. Audio must always be brought in on the left channel
for mono operation.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch replaces the string "micbias1" to "MICBIAS1".
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
we need to release dapm widget list after dpcm_path_get in
soc_dpcm_runtime_update. otherwise, there will be potential memory
leak. add dpcm_path_put to fix it.
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Passing unsigned int pointers as u32 ponters may be dangerous on 64-bit
system.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Do not fail if the exact BLCK rate can not be produced, just print a
warning. Check that sysclk frequency is set before implicitly setting
the BCLK divider.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fixes build with SND_DAVINCI_SOC or SND_OMAP_SOC alone and adds build
dependecy to SND_DAVINCI_SOC or SND_OMAP_SOC.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
of_get_child_by_name expects of_node_put be called when done.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
max98090.c doesn't free the threaded interrupt it requests. This causes
an oops when doing "cat /proc/interrupts" after snd-soc-max98090.ko is
unloaded.
Fix this by requesting the interrupt by using devm_request_threaded_irq().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Cc: Stable <stable@vger.kernel.org> # 3.10+
Signed-off-by: Mark Brown <broonie@linaro.org>
We now have a generic helper function to cast from a DAPM context to a CODEC.
Make use of it in the places which previously open-coded it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In DIT (S/PDIF) mode, program the transmitted user bits to reflect the
configured sample rate, along with some other details.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Document the newly added regulators to the DT binding document.
Also, "static const char const *x" is not identical to "static const
char * const x", which sparse now complains about. Fix it.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The chip has two power supplies, VA and VDD. Enable them both as long
as the codec is in use.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Quite a few build coverage fixes in here among the usual small driver
fixes includling the sigmadsp change from Lars - moving the driver to
separate modules per bus (which is basically just code motion) avoids
issues with some combinations of buses being enabled.
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Merge tag 'asoc-v3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.16
Quite a few build coverage fixes in here among the usual small driver
fixes includling the sigmadsp change from Lars - moving the driver to
separate modules per bus (which is basically just code motion) avoids
issues with some combinations of buses being enabled.
The ALSA control code expects that the range of assigned indices to a control is
continuous and does not overflow. Currently there are no checks to enforce this.
If a control with a overflowing index range is created that control becomes
effectively inaccessible and unremovable since snd_ctl_find_id() will not be
able to find it. This patch adds a check that makes sure that controls with a
overflowing index range can not be created.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Each control gets automatically assigned its numids when the control is created.
The allocation is done by incrementing the numid by the amount of allocated
numids per allocation. This means that excessive creation and destruction of
controls (e.g. via SNDRV_CTL_IOCTL_ELEM_ADD/REMOVE) can cause the id to
eventually overflow. Currently when this happens for the control that caused the
overflow kctl->id.numid + kctl->count will also over flow causing it to be
smaller than kctl->id.numid. Most of the code assumes that this is something
that can not happen, so we need to make sure that it won't happen
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A control that is visible on the card->controls list can be freed at any time.
This means we must not access any of its memory while not holding the
controls_rw_lock. Otherwise we risk a use after free access.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are two issues with the current implementation for replacing user
controls. The first is that the code does not check if the control is actually a
user control and neither does it check if the control is owned by the process
that tries to remove it. That allows userspace applications to remove arbitrary
controls, which can cause a user after free if a for example a driver does not
expect a control to be removed from under its feed.
The second issue is that on one hand when a control is replaced the
user_ctl_count limit is not checked and on the other hand the user_ctl_count is
increased (even though the number of user controls does not change). This allows
userspace, once the user_ctl_count limit as been reached, to repeatedly replace
a control until user_ctl_count overflows. Once that happens new controls can be
added effectively bypassing the user_ctl_count limit.
Both issues can be fixed by instead of open-coding the removal of the control
that is to be replaced to use snd_ctl_remove_user_ctl(). This function does
proper permission checks as well as decrements user_ctl_count after the control
has been removed.
Note that by using snd_ctl_remove_user_ctl() the check which returns -EBUSY at
beginning of the function if the control already exists is removed. This is not
a problem though since the check is quite useless, because the lock that is
protecting the control list is released between the check and before adding the
new control to the list, which means that it is possible that a different
control with the same settings is added to the list after the check. Luckily
there is another check that is done while holding the lock in snd_ctl_add(), so
we'll rely on that to make sure that the same control is not added twice.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The user-control put and get handlers as well as the tlv do not protect against
concurrent access from multiple threads. Since the state of the control is not
updated atomically it is possible that either two write operations or a write
and a read operation race against each other. Both can lead to arbitrary memory
disclosure. This patch introduces a new lock that protects user-controls from
concurrent access. Since applications typically access controls sequentially
than in parallel a single lock per card should be fine.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current dma name search loop didn't care about SSI index
This patch fixes it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ad32d0c7b0
(ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr)
added rsnd_gen_dma_addr() to calculate DMA addr,
but, it is necessary only for Gen2.
This patch ignores Gen1 case.
Kernel will be panic without this patch.
Special thanks to Simon
Reported-by: Simon Horman <horms@verge.net.au>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Simon Horman <horms+renesas@verge.net.au>
Signed-off-by: Mark Brown <broonie@linaro.org>
Building the pxa/mmp audio driver without support for the mmp
sram driver enabled results in this link error:
sound/built-in.o: In function `mmp_pcm_free_dma_buffers':
:(.text+0x3e734): undefined reference to `sram_get_gpool'
sound/built-in.o: In function `mmp_pcm_new':
:(.text+0x3e7c0): undefined reference to `sram_get_gpool'
The sram driver is cannot be manually enabled and needs to
be turned on by selecting MMP_SRAM from each module that
needs it, which is what this patch does.
Ideally, MMP should move over to the generic SRAM support, but
for the moment, we can avoid the build error.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
wm1811_micd_stop takes the accdet_lock mutex, and is called from two
places, one of which is already holding the accdet_lock. This obviously
causes a lock up.
This patch fixes this issue by removing the lock from wm1811_micd_stop
and ensuring that it is always locked externally.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
do_div() requires that the first parameter is a 64-bit integer,
which but clkrate was defined as an unsigned long. This caused
the following warnings:
CC sound/soc/fsl/fsl_ssi.o
sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_set_bclk':
sound/soc/fsl/fsl_ssi.c:593:3: warning: comparison of distinct pointer types lacks a cast
sound/soc/fsl/fsl_ssi.c:593:3: warning: right shift count >= width of type
sound/soc/fsl/fsl_ssi.c:593:3: warning: passing argument 1 of '__div64_32' from incompatible pointer type
include/asm-generic/div64.h:35:17: note: expected 'uint64_t *' but argument is of type 'long unsigned int *'
Signed-off-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit a0b148b4 (ASoC: wm8985: Use devm_regulator_bulk_get()) removed the last
user of pointer wm8985 to struct wm8985_priv. Thus remove it. Detected by
Coverity CID 1222150.
Signed-off-by: Christian Engelmayer <cengelma@gmx.at>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add support for the output sample rates 96kHz and 192kHz.
Tested with a Cubox-i imx6 system and an Onkyo TX-SR607 receiver.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
1) Apply better indentations
2) Drop braces for single statement.
3) Use simpler ternary to reduce code.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Transmitters and receivers may support a 192kHz sample rate.
Tested with a Cubox-i imx6 system and an Onkyo TX-SR607 receiver.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use the zeroing function instead of dma_alloc_coherent & memset(,0,)
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a machine is booted with nomodeset option, i915 driver skips the
whole initialization. Meanwhile, HD-audio tries to bind wth i915 just
by request_symbol() without knowing that the initialization was
skipped, and eventually it hits WARN_ON() in i915_request_power_well()
and i915_release_power_well() wrongly but still continues probing,
even though it doesn't work at all.
In this patch, both functions are changed to return an error in case
of uninitialized state instead of WARN_ON(), so that HD-audio driver
can give up HDMI controller initialization at the right time.
Acked-by: Daniel Vetter <daniel.vetter@ffwll.ch>
Cc: <stable@vger.kernel.org> [3.15]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of changes are small and easy cleanup or fixes.
- a few HD-audio Realtek codec fixes and quirks
- Intel HDMI audio fixes for Broadwell and Haswell / ValleyView
- FireWire sound stack cleanups
- a couple of sequencer core fixes
- compress ABI fix for 64bit
- Conversion to modern ktime*() API
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Merge tag 'sound-fix-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Most of changes are small and easy cleanup or fixes:
- a few HD-audio Realtek codec fixes and quirks
- Intel HDMI audio fixes for Broadwell and Haswell / ValleyView
- FireWire sound stack cleanups
- a couple of sequencer core fixes
- compress ABI fix for 64bit
- conversion to modern ktime*() API"
* tag 'sound-fix-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (23 commits)
ALSA: hda/realtek - Add more entry for enable HP mute led
ALSA: hda - Add quirk for external mic on Lifebook U904
ALSA: hda - fix a fixup value for codec alc293 in the pin_quirk table
ALSA: intel8x0: Use ktime and ktime_get()
ALSA: core: Use ktime_get_ts()
ALSA: hda - verify pin:converter connection on unsol event for HSW and VLV
ALSA: compress: Cancel the optimization of compiler and fix the size of struct for all platform.
ALSA: hda - Add quirk for ABit AA8XE
Revert "ALSA: hda - mask buggy stream DMA0 for Broadwell display controller"
ALSA: hda - using POS_FIX_LPIB on Broadwell HDMI Audio
ALSA: hda/realtek - Add support of ALC667 codec
ALSA: hda/realtek - Add more codec rename
ALSA: hda/realtek - New vendor ID for ALC233
ALSA: hda - add two new pin tables
ALSA: hda/realtek - Add support of ALC891 codec
ALSA: seq: Continue broadcasting events to ports if one of them fails
ALSA: bebob: Remove unused function prototype
ALSA: fireworks: Remove meaningless mutex_destroy()
ALSA: fireworks: Remove a constant over width to which it's applied
ALSA: fireworks: Improve comments about Fireworks transaction
...
More HP machine need mute led support.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fixup value for codec alc293 was set to
ALC269_FIXUP_DELL1_MIC_NO_PRESENCE by a mistake, if we don't fix it,
the Dock mic will be overwriten by the headset mic, this will make
the Dock mic can't work.
Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
do_posix_clock_monotonic_gettime() is a leftover from the initial
posix timer implementation which maps to ktime_get_ts() and returns
the monotonic time in a timespec.
Use ktime based ktime_get() and use the ktime_delta_us() function to
calculate the delta instead of open coding the timespec math.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
do_posix_clock_monotonic_gettime() is a leftover from the initial
posix timer implementation which maps to ktime_get_ts().
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch will verify the pin's coverter selection for an active stream
when an unsol event reports this pin becomes available again after a display
mode change or hot-plug event.
For Haswell+ and Valleyview: display mode change or hot-plug can change the
transcoder:port connection and make all the involved audio pins share the 1st
converter. So the stream using 1st convertor will flow to multiple pins
but active streams using other converters will fail. This workaround
is to assure the pin selects the right conveter and an assigned converter is
not shared by other unused pins.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Controls for shaping the ultrasonic frequency response were introduced
in this commit:
commit 720630c002ffc7b0fa2ed5b3f4bfb36fd8f87ca6
ASoC: wm5102: Add controls to allow shaping of ultrasonic response
However, they mistakenly used snd_kcontrol_chip instead of
snd_soc_kcontrol_codec, which has replaced it now the framework is
moving to componentisation. This patch fixes this.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit 432481220 (ASoC: fsl-ssi: Use regmap) removed struct ccsr_ssi.
Unfortunately, the structure is still used. This causes
mpc85xx_smp_defconfig and mpc85xx_defconfig builds to fail with
sound/soc/fsl/fsl_dma.c:926:50:
error: invalid use of undefined type 'struct ccsr_ssi'
dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0);
ound/soc/fsl/fsl_dma.c:927:50:
error: invalid use of undefined type 'struct ccsr_ssi'
dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0);
Fix by using constants, similar to original commit.
Cc: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Guenter Roeck <linux@roeck-us.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
Index of dma name should use -1, not +1 when capture case.
Thank you Dan.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Bios does not set up the pin config default correctly (everything
is set to zero). Reporter claims that 6stack-dig and 6stack-automute
solve the problem.
Alsa-info at http://www.alsa-project.org/db/?f=376c0804cbdde90bcd2cb94799407cb1cacf5d05
BugLink: https://bugs.launchpad.net/bugs/1319291
Reported-by: Stefano Statuti <stefano.statuti@hotmail.it>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the SigmaDSP module is built-in, but the I2C core is build as a module
we'll get a undefined reference:
sound/built-in.o: In function `sigma_action_write_i2c':
:(.text+0x5d8d4): undefined reference to `i2c_master_send'
This can happen if a audio driver that is using the regmap SigmaDSP interface is
built into the kernel, but core I2C support is build as a module. To fix this
split the SigmaDSP module into three modules, one module providing the core
infrastructure and two small modules implementing the regmap and I2C interfaces.
This allows e.g. the core infrastructure and regmap support to be built into the
kernel while I2C support can still be build as a module.
Fixes: dab464b60 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add controls to allow custom shaping of the ultrasonic response. This
custom shaping can be turned on/off at runtime, although, it should be
noted that settings will not affect a currently open audio stream,
they will be applied when the next audio stream is started.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add the clock prepare and unprepare call to the driver initialization
phase. This will remove a warning once the PXA architecture is migrated
to the clock infrastructure.
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
The calculation code does
u64 = (u32 - u32) * 100000;
The 64 bits are of no help here as the type is casted only after the
multiplication, and therefore the result may overflow, possibly causing
inoptimal or wrong clock setup in an unfortunate case (the maximum
result value of the first substraction is currently 47999).
Fix the code to cast before multiplication.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Acked-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We should not copy the return value into this val since it's supposed to
get the value of the register not the success result of regmap_read().
Thus fix it.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit c9e065c27f ("ASoC: dapm: Make sure to always update the DAPM graph
in _put_volsw()") stopped updating register values in those cases where
initial after boot state of kcontrol appears to not change but where
register value still needs update because it is not in sync with the
kcontrol state.
Fix this by doing snd_soc_test_bits() unconditionally as it was before but
by using separate flags for kcontrol and register state changes. This allow
both DAPM graph to be updated when disabling auto-muted control and update
register if it is out-of-sync in respect of kcontrol state.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This reverts commit 7189eb9b8f.
It will use LPIB to get the DMA position on Broadwell HDMI Audio.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Broadwell HDMI can't use position buffer reliably, force to use LPIB
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the SigmaDSP module is built-in, but the I2C core is build as a module
we'll get a undefined reference:
sound/built-in.o: In function `sigma_action_write_i2c':
:(.text+0x5d8d4): undefined reference to `i2c_master_send'
This can happen if a audio driver that is using the regmap SigmaDSP interface is
built into the kernel, but core I2C support is build as a module. To fix this
split the SigmaDSP module into three modules, one module providing the core
infrastructure and two small modules implementing the regmap and I2C interfaces.
This allows e.g. the core infrastructure and regmap support to be built into the
kernel while I2C support can still be build as a module.
Fixes: dab464b60 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
These two new pin tables can fix headset mic problems for several
new Dell machines.
And also delete some machines from old quirk table since the existing
pin talbes already cover them.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
From e7a94bb7fb871c73cc85712d89c1f48d0271c1be Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Thu, 5 Jun 2014 12:31:28 +0200
Subject: [PATCH] ASoC: MMP audio needs sram support
Building the pxa/mmp audio driver without support for the mmp
sram driver enabled results in this link error:
sound/built-in.o: In function `mmp_pcm_free_dma_buffers':
:(.text+0x3e734): undefined reference to `sram_get_gpool'
sound/built-in.o: In function `mmp_pcm_new':
:(.text+0x3e7c0): undefined reference to `sram_get_gpool'
The sram driver is cannot be manually enabled and needs to
be turned on by selecting MMP_SRAM from each module that
needs it, which is what this patch does.
Ideally, MMP should move over to the generic SRAM support, but
for the moment, we can avoid the build error.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Haojian Zhuang <haojian.zhuang@gmail.com>
Cc: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
New codec support for ALC891.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At this time, majority of changes come from ASoC world while we got a
few new drivers in other places for FireWire and USB. There have been
lots of ASoC core cleanups / refactoring, but very little visible to
external users.
ASoC
- Support for specifying aux CODECs in DT
- Removal of the deprecated mux and enum macros
- More moves towards full componentisation
- Removal of some unused I/O code
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers
- Several drivers exposed directly in Kconfig for use with simple-card
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
ST STA350, Analog Devices ADAU1361, ADAU1381, ADAU1761 and ADAU1781,
and Realtek RT5677
HD-audio:
- Clean up Dell headset quirks
- Noise fixes for Dell and Sony laptops
- Thinkpad T440 dock fix
- Realtek codec updates (ALC293,ALC233,ALC3235)
- Tegra HD-audio HDMI support
FireWire-audio:
- FireWire audio stack enhancement (AMDTP, MIDI), support for incoming
isochronous stream and duplex streams with timestamp synchronization
- BeBoB-based devices support
- Fireworks-based device support
USB-audio:
- Behringer BCD2000 USB device support
Misc:
- Clean up of a few old drivers, atmel, fm801, etc
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Merge tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into next
Pull sound updates from Takashi Iwai:
"At this time, majority of changes come from ASoC world while we got a
few new drivers in other places for FireWire and USB. There have been
lots of ASoC core cleanups / refactoring, but very little visible to
external users.
ASoC:
- Support for specifying aux CODECs in DT
- Removal of the deprecated mux and enum macros
- More moves towards full componentisation
- Removal of some unused I/O code
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers
- Several drivers exposed directly in Kconfig for use with
simple-card
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651
and ST STA350, Analog Devices ADAU1361, ADAU1381, ADAU1761 and
ADAU1781, and Realtek RT5677
HD-audio:
- Clean up Dell headset quirks
- Noise fixes for Dell and Sony laptops
- Thinkpad T440 dock fix
- Realtek codec updates (ALC293,ALC233,ALC3235)
- Tegra HD-audio HDMI support
FireWire-audio:
- FireWire audio stack enhancement (AMDTP, MIDI), support for
incoming isochronous stream and duplex streams with timestamp
synchronization
- BeBoB-based devices support
- Fireworks-based device support
USB-audio:
- Behringer BCD2000 USB device support
Misc:
- Clean up of a few old drivers, atmel, fm801, etc"
* tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (480 commits)
ASoC: Fix wrong argument for card remove callbacks
ASoC: free jack GPIOs before the sound card is freed
ALSA: firewire-lib: Remove a comment about restriction of asynchronous operation
ASoC: cache: Fix error code when not using ASoC level cache
ALSA: hda/realtek - Fix COEF widget NID for ALC260 replacer fixup
ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptop
ALSA: firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio data
ASoC: add RT5677 CODEC driver
ASoC: intel: The Baytrail/MAX98090 driver depends on I2C
ASoC: rt5640: Add the function "get_clk_info" to RL6231 shared support
ASoC: rt5640: Add the function of the PLL clock calculation to RL6231 shared support
ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651
ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error.
ASoC: Add helper functions to cast from DAPM context to CODEC/platform
ALSA: bebob: sizeof() vs ARRAY_SIZE() typo
ASoC: wm9713: correct mono out PGA sources
ALSA: synth: emux: soundfont.c: Cleaning up memory leak
ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320
ASoC: fsl-ssi: Use regmap
ASoC: fsl-ssi: reorder and document fsl_ssi_private
...
Sometimes PORT_EXIT messages are lost when a process is exiting.
This happens if you subscribe to the announce port with client A,
then subscribe to the announce port with client B, then kill client A.
Client B will not see the PORT_EXIT message because client A's port is
closing and is earlier in the announce port subscription list. The
for each loop will try to send the announcement to client A and fail,
then will stop trying to broadcast to other ports. Killing B works fine
since the announcement will already have gone to A. The CLIENT_EXIT
message does not get lost.
How to reproduce problem:
*** termA
$ aseqdump -p 0:1
0:1 Port subscribed 0:1 -> 128:0
*** termB
$ aseqdump -p 0:1
*** termA
0:1 Client start client 129
0:1 Port start 129:0
0:1 Port subscribed 0:1 -> 129:0
*** termB
0:1 Port subscribed 0:1 -> 129:0
*** termA
^C
*** termB
0:1 Client exit client 128
<--- expected Port exit as well (before client exit)
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently mutex_destroy() is called in module's cleanup function. But after
cleaned up, this mutex is automatically released. So this function call
is meaningless.
[fixed a typo in changelog by tiwai]
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The constants of enum snd_efw_grp_type is for struct snd_efw_phys_grp.type.
But this member is 1 byte. Although the value is between 0x00-0xff, a constant
has 0x10000. This constant is meaningless.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To reverse a pointer for the ring buffer, subtraction by buffer
size is better than assignment to the beginning of the buffer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All assignment for local variables in these functions are not related to
critical section.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_seq_event_dup returns -ENOMEM in some buffer-full conditions,
but usually returns -EAGAIN. Make -EAGAIN trigger the overflow
condition in snd_seq_fifo_event_in so that the fifo is cleared
and -ENOSPC is returned to userspace as stated in the alsa-lib docs.
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have in the past added 'depends on I2C' for some of the PXA boards
after hitting randconfig build bugs. I have seens a couple of new
bugs in this area during the linux-next cycle for 3.16, after it
became possible to build some more PXA machines with I2C disabled.
To shut this up for good, this adds the dependency to every board
that uses I2C as the interface to the codec. I have gone through
all board files and verified that they all either use AC97 or
I2C, and this annotates the latter. Some of these already enable
I2C from mach-pxa/Kconfig, but since that can change it's better
to be explicit here.
The link error that can result otherwise happens when CONFIG_I2C
is set to 'm' and the codec driver is built-in as a result of being
selected by the platform specific glue.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The commit [e1d4d3c8: ASoC: free jack GPIOs before the sound card is
freed] introduced snd_soc_card remove callbacks to a few drivers, but
they are implemented with a wrong argument type. The callback should
receive snd_soc_card pointer instead of snd_soc_pcm_runtime.
Fixes: e1d4d3c854 ('ASoC: free jack GPIOs before the sound card is freed')
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more updates from the last week of development, nothing too
exciting. Highlights include:
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers.
- New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and
ADAU1781, and Realtek RT5677.
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Merge tag 'asoc-v3.16-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Final updates for v3.16
A few more updates from the last week of development, nothing too
exciting. Highlights include:
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers.
- New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and
ADAU1781, and Realtek RT5677.
This is the same change as commit fb6b8e7144 "ASoC: tegra: free jack
GPIOs before the sound card is freed", but applied to all other ASoC
machine drivers where code inspection indicates the same problem exists.
That commit's description is:
==========
snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to
generate an initial jack status report. If sound card initialization
fails, that work item needs to be cancelled, so it doesn't run after the
card has been freed. Specifically, freeing the card calls
snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets
jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which
is called from the work queue item.
snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine
drivers do call this function in the platform driver remove() callback.
However, this happens after the sound card is freed, at least when the
card is freed due to errors late during snd_soc_instantiate_card(). This
leaves a window where the work item can execute after the card is freed.
In next-20140522, sound card initialization does fail for unrelated
reasons, and hits the problem described above.
To solve this, fix the Tegra ASoC machine drivers to clean up the Jack
GPIOs during the snd_soc_card's .remove() callback, which is executed
before the overall card object is freed. also, guard the cleanup call
based on whether we actually setup up the GPIOs in the first place.
Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove
function to match where the GPIOs get set up. However, there is no such
callback.
==========
Note that I have not even compile-tested this in most cases, since most
of the drivers rely on specific mach-* support I don't have enabled, and
don't support COMPILE_TEST. Testing by the relevant board maintainers
would be useful.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The comment for fcp_avc_transaction() describes it doesn't support this type
of operation. But it was already supported by this commit.
00a7bb81c2
ALSA: firewire-lib: Add support for deferred transaction
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A quite large set of SoC updates this cycle. In no particular order:
- Multi-cluster power management for Samsung Exynos, adding support for
big.LITTLE CPU switching on EXYNOS5
- SMP support for Marvell Armada 375 and 38x
- SMP rework on Allwinner A31
- Xilinx Zynq support for SOC_BUS, big endian
- Marvell orion5x platform cleanup, modernizing the implementation and
moving to DT.
- _Finally_ moving Samsung Exynos over to support MULTIPLATFORM, so
that their platform can be enabled in the same kernel binary as most
of the other v7 platforms in the tree. \o/ The work isn't quite complete,
there's some driver fixes still needed, but the basics now work.
New SoC support added:
- Freescale i.MX6SX
- LSI Axxia AXM55xx SoCs
- Samsung EXYNOS 3250, 5260, 5410, 5420 and 5800
- STi STIH407
Plus a large set of various smaller updates for different platforms. I'm
probably missing some important one here.
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Merge tag 'soc-for-3.16' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc into next
Pull part one of ARM SoC updates from Olof Johansson:
"A quite large set of SoC updates this cycle. In no particular order:
- Multi-cluster power management for Samsung Exynos, adding support
for big.LITTLE CPU switching on EXYNOS5
- SMP support for Marvell Armada 375 and 38x
- SMP rework on Allwinner A31
- Xilinx Zynq support for SOC_BUS, big endian
- Marvell orion5x platform cleanup, modernizing the implementation
and moving to DT.
- _Finally_ moving Samsung Exynos over to support MULTIPLATFORM, so
that their platform can be enabled in the same kernel binary as
most of the other v7 platforms in the tree. \o/
The work isn't quite complete, there's some driver fixes still
needed, but the basics now work.
New SoC support added:
- Freescale i.MX6SX
- LSI Axxia AXM55xx SoCs
- Samsung EXYNOS 3250, 5260, 5410, 5420 and 5800
- STi STIH407
plus a large set of various smaller updates for different platforms.
I'm probably missing some important one here"
* tag 'soc-for-3.16' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (281 commits)
ARM: exynos: don't run exynos4 l2x0 setup on other platforms
ARM: exynos: Fix "allmodconfig" build errors in mcpm and hotplug
ARM: EXYNOS: mcpm rename the power_down_finish
ARM: EXYNOS: Enable mcpm for dual-cluster exynos5800 SoC
ARM: EXYNOS: Enable multi-platform build support
ARM: EXYNOS: Consolidate Kconfig entries
ARM: EXYNOS: Add support for EXYNOS5410 SoC
ARM: EXYNOS: Support secondary CPU boot of Exynos3250
ARM: EXYNOS: Add Exynos3250 SoC ID
ARM: EXYNOS: Add 5800 SoC support
ARM: EXYNOS: initial board support for exynos5260 SoC
clk: exynos5410: register clocks using common clock framework
ARM: debug: qcom: add UART addresses to Kconfig help for APQ8084
ARM: sunxi: allow building without reset controller
Documentation: devicetree: arm: sort enable-method entries
ARM: rockchip: convert smp bringup to CPU_METHOD_OF_DECLARE
clk: exynos5250: Add missing sysmmu clocks for DISP and ISP blocks
ARM: dts: axxia: Add reset controller
power: reset: Add Axxia system reset driver
ARM: axxia: Adding defconfig for AXM55xx
...
It is not an error to have no cache so we shouldn't return an error code
and cause our callers to fail, just silently do nothing instead. Thanks
to Jarkko for identify the problematic commit.
Reported-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Reported-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The conversion to a fixup table for Replacer model with ALC260 in
commit 20f7d928 took the wrong widget NID for COEF setups. Namely,
NID 0x1a should have been used instead of NID 0x20, which is the
common node for all Realtek codecs but ALC260.
Fixes: 20f7d928fa ('ALSA: hda/realtek - Replace ALC260 model=replacer with the auto-parser')
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correcion of wrong fixup entries add in commit ca8f0424 to replace
static model quirk for PB V7900 laptop (will model).
[note: the removal of ALC260_FIXUP_HP_PIN_0F chain is also needed as a
part of the fix; otherwise the pin is set up wrongly as a headphone,
and user-space (PulseAudio) may be wrongly trying to detect the jack
state -- tiwai]
Fixes: ca8f04247e ('ALSA: hda/realtek - Add the fixup codes for ALC260 model=will')
Signed-off-by: Ronan Marquet <ronan.marquet@orange.fr>
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to AM824 in IEC 61883-6:2002, 2 bits in LSB of label for Raw Audio
data means Valid Length Code (VBL). Ths value is:
- b00 for 24 bits sample (label is 0x40)
- b01 for 20 bits sample (label is 0x41)
- b10 for 16 bits sample (label is 0x42)
But current firewire-lib apply 24 bits label for both of 16/24 bits samples.
As long as developers investigate BeBoB/Fireworks/OXFW/Dice, all of them
have a behaviour to ignore the label. They can generate correct sound even
if firewire-lib gives wrong label (i.e. 0xff). On BeBoB, this is not only
for Raw Audio data channel, but also for IEC 60958 Conformant data channel.
So there is little possibility of regression.
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch adds the function "get_clk_info" to RL6231 shared support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch adds the function of the PLL clock calculation to RL6231 shared
support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch adds the RL6231 class device shared support for RT5640, RT5645 and
RT5651. The function of the DMIC clock calculation can be shared by RL6231
shared support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since we cannot make sure the 'reg_size' will always be none zero here,
and then if 'reg_size' equals to zero, the kzalloc() will return ZERO_SIZE_PTR,
which equals to ((void *)16).
So this patch fix this with just doing the 'reg_size' zero check before calling
kzalloc().
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ARRAY_SIZE() was intended here instead of sizeof(). The
"bridgeco_freq_table" array holds integers so the original condition is
never true.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewd-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mono output PGA input only has four possible sources, so
omit the rest.
Signed-off-by: Matt Reimer <mreimer@sdgsystems.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is a risk for memory leak in when something unexpected happens
and the function returns.
This was largely found by using a static code analysis program called cppcheck.
[fixed a typo of kfree() by tiwai]
Signed-off-by: Rickard Strandqvist <rickard_strandqvist@spectrumdigital.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eukrea-i.MX51 board was converted to use DT, ie we no longer have a
MACH_EUKREA_MBIMXSD51_BASEBOARD symbol.
Transformation of other boards planned for the near future, so this
patch removes all these dependencies and restricts build of this
driver to ARCH_MXC.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch replaces the ssi specific functions write_ssi, read_ssi and
write_ssi_mask by standard regmap function calls.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reorder all variables in struct fsl_ssi_private to have groups that make
sense together. The patch also updates the struct documentation.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The baudclock may be used and set by different streams.
Allow only the first stream to set the bitclock rate. Other streams have
to try to get to the correct rate without modifying the bitclock rate
using the SSI internal clock modifiers.
The variable baudclk_streams is introduced to keep track of the active
streams that are using the baudclock. This way we know if the baudclock
may be set and whether we may enable/disable the clock.
baudclock enable/disable is moved to hw_params()/hw_free(). This way we can
keep track of the baudclock in those two functions and avoid a running
clock while it is not used. As hw_params()/hw_free() may be called
multiple times for the same stream, we have to use baudclk_streams
variable to know whether we may enable/disable the clock.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In i2s master mode the fsl_ssi driver depends on someone calling
.set_tdm_slot correctly. In this mode though only a DC value of
2 is allowed, so set it in this case and no longer depend on
.set_tdm_slot.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The baudclock_locked variable is only used in functions which
are serialized anyway from the core. No need to have a lock
around the variable, so remove it.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The fsl_ssi driver uses the .set_sysclk callback to configure the
bitclock for master mode. This is unnecessary since the bitclock
is known in hw_params. This patch configures the bitclock from .hw_params.
.set_dai_sysclk now sets a bitclock frequency which is preferred over
the default calculated bitclock frequency.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The simple soundcard binding has its own way for specifying the dai
format. To be able to use this binding we have to make the fsl,mode
property optional. As the property is used in existing devicetrees
keep the option around for compatibility reasons.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Introduce a SoC data struct which contains the differences between
the different SoCs this driver supports. This makes it easy to support
more differences without having to introduce a new switch/case each
time.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use card PM ops from ASoC core instead of defining custom PM ops here since
we are calling anyway common suspend/resume callbacks.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Simplify byt-rt5640.c and haswell.c machine drivers by using
devm_snd_soc_register_card(). Remove also needless dev_set_drvdata()
from byt_rt5640_probe() since snd_soc_register_card() does it too.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A few files contain duplicate headers. This patch removes the second entry of
duplicate in each file under question.
There is no functional changes.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Stored DSP DMA pointer must be cleared before starting the stream since
PCM pointer callback sst_byt_pcm_pointer() can be called before pointer is
updated. In that case last position of previous stream was wronly returned.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The new value argument needs proper shift to match the mask bit fields.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Brian Austin <brian.austin@cirrus.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Baytrail and Haswell SST IPC don't stop the kernel thread in error and
cleanup path thus leaving orphan kernel thread behind in such a case.
Also while at it, fix one error path in sst-haswell-ipc.c that doesn't free
hsw->msg.
[Jarkko: I edited the commit log a little]
Signed-off-by: Imre Deak <imre.deak@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add machine driver and ACPI probing for Baytrail SST with MAX98090 codec.
Jack detect code from Kevin Strasser <kevin.strasser@intel.com>, GPIO
resolving from Mika Westerberg <mika.westerberg@linux.intel.com> and fixes
and cleanups from Liam Girdwood <liam.r.girdwood@linux.intel.com>.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For some unknown reason the parameters for snd_soc_test_bits() were in wrong
order:
It was:
snd_soc_test_bits(codec, val, mask, reg); /* WRONG!!! */
while it should be:
snd_soc_test_bits(codec, reg, mask, val);
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Now snd-firewire-lib supports rawmidi in addition to PCM, thus we need
to give a proper dependency. For fixing and simplification, move the
selections of SND_PCM and SND_RAWMIDI into SND_FIREWIRE_LIB section.
Then each driver doesn't have to select them but only
SND_FIREWIRE_LIB.
Reported-by: Jim Davis <jim.epost@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS A8JN with AD1986A codec seems following the normal EAPD in the
normal order (0 = off, 1 = on) unlike other machines with AD1986A.
Apply the workaround used for Toshiba laptop that showed the same
problem.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=75041
Cc: <stable@vger.kernel.org> [3.11+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Checks for CONFIG_SND_DEBUG_ROM were added in v2.5.5 but a Kconfig
symbol SND_DEBUG_ROM was never added. These checks have always
evaluated to false. Remove them and the printk()s they hide.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Checks for CONFIG_BCM_CS4297A_CSWARM were added in v2.6.11. The related
Kconfig symbol was never added so these checks always evaluated to true.
Remove them.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
WM8804 can run with PLL frequencies of 256xfs and 128xfs for
most sample rates. At 192kHz only 128xfs is supported. The
existing driver selects 128xfs automatically for some lower
samples rates. By using an additional mclk_div divider, it
is now possible to control the behaviour. This allows using
256xfs PLL frequency on all sample rates up to 96kHz. It
should allow lower jitter and better signal quality. The
behavior has to be controlled by the sound card driver,
because some sample frequency share the same setting. e.g.
192kHz and 96kHz use 24.576MHz master clock. The only
difference is the MCLK divider.
Signed-off-by: Daniel Matuschek <daniel@matuschek.net>
Tested-by: Florian Meier <florian.meier@koalo.de>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This reverts commit c687200b9d.
Dropping the def association and sequence from pinconf comparing is a
bit risky, It will introduce a greater risk of catching unwanted
machines.
And in addition, so far no BIOS experts give us an explicit answer
whether it makes senses to compare these two fields or not.
For safety reason, we revert this commit.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There was a typo here so we return directly instead of freeing "hwinfo".
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewd-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We checked "err" earlier. These things seem to be left over code.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewd-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This issue was reported by coccicheck using the semantic patch
at scripts/coccinelle/api/resource_size.cocci
Signed-off-by: Benoit Taine <benoit.taine@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a ASoC machine driver to support the EVAL-ADAU1X81 board
connected to a Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a ASoC machine driver to support the EVAL-ADAU1X61 board
connected to a Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Analog Devices ADAU1381 and ADAU1781 audio
CODECs. The device is a low-power, 24-bit stereo audio CODEC with multiple
analog inputs and outputs, two digital microphone inputs and an I2S interface.
The device can be controlled either using I2C or SPI. The main difference
between the two variants is that the ADAU1781 has a freely programmable SigmaDSP
processor, while the ADAU1381 has a fixed function wind noise reduction filter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Analog Devices ADAU1361 and ADAU1761 CODECs.
The device is a a low-power, 24-bit stereo audio CODEC with multiple analog
input and outputs, one digital microphone input and an I2S interface. The device
can be controlled either via I2C or SPI. The main difference between the two
variants is that the ADAU1761 has a built-in SigmaDSP, while the ADAU1361 has
not.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ADAU1X61 and ADAU1X81 are very similar in the digital domain, but are quite
different in the analog domain. This patch adds support for the common parts of
the ADAU1X61 and ADAU1X81 CODECs.
The patch also restores some of the alphabetical order in the Makfile and
Kconfig.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently bebob driver apply Raw Audio Data channel (in IEC 61883-1:2002,
Multi Bit Linear Audio Data channel in IEC 61883-6:20005) to IEC 60958
Conformant Data channel because both fireworks and bebob based devices
can handle it by ignoring each label.
This patch improves a comment about this.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently mutex_unlock() is called in module's cleanup function. But after
cleaned up, this mutex is automatically released. So this function call
is meaningless.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some variables were declared without static even if they're not referred
to from external files. This commit make them local symbols for better
information-hiding by file unit.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds suffix to register values of each device, to supress 'sparse'
warnings. Additionally, this commit changes offset values with integer literal.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The second argument of snd_efw_command_get_sampling_rate() means sampling
rate and its type is 'unsigned int'. But 'int' variable is passed as parameter.
It's better to apply the same type for the variable as its argument.
Similally, the type of variable for snd_efw_command_get_clock_source() and
avc_bridgeco_get_plug_type() should be the same type as each argument.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Originally, I intent to this argument given with 'struct snd_pcm_runtime.rate'
or params_rate(). They return value of 'unsigned int'. So 'unsigned int' is
better for the type of this argument.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are two modes for Fireworks, IEC 61883 compliant or Windows.
So it's better to use enum type instead of int to express the intension,
even if C language specification defines to handle enum variables as usual
integer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The return value of memdup_user() should be passed to return correct error.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
__be32 value should not be assigned directly to bool value.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit fixes a big for loop count with array. The limitation of loop
count should be calcurated with the number of elements in the array, not
with the number of bytes.
Aditionally, this commit apply the same declaration as a prototype in header
for the array.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have defines for adsp messages best to consistently use them.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since commit e5d80e82e3 (ASoC: sgtl5000: Convert to use regmap directly) a
kernel oops is observed after a suspend/resume sequence.
The kernel oops happens inside sgtl5000_restore_regs() as codec->reg_cache is no
longer a valid pointer.
Add the remaining register entries into sgtl5000_reg_defaults[] and remove
sgtl5000_restore_regs() completely, which allows suspend/resume to work fine and
make the code simpler.
Tested on a im53-qsb board.
Reported-by: Shawn Guo <shawn.guo@freescale.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When CONFIG_PM is disabled, the CONFIG_SND_HDA_POWER_SAVE_DEFAULT symbol
does not get defined, which causes a build error for the hda-tegra driver:
hda/hda_tegra.c:80:25: error: 'CONFIG_SND_HDA_POWER_SAVE_DEFAULT' undeclared here (not in a function)
static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
^
/git/arm-soc/sound/pci/hda/hda_tegra.c:235:13: warning: 'hda_tegra_disable_clocks' defined but not used [-Wunused-function]
static void hda_tegra_disable_clocks(struct hda_tegra *data)
^
This works around the problem by not referencing that macro
when CONFIG_PM is disabled. Instead, we assume that it's disabled
unconditionally and cannot be enabled at runtime.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Cc: Dylan Reid <dgreid@chromium.org>
Cc: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
commit 8c5178fca4 ("ALSA: Add params_width() helpers") introduces
a helper to get the sample width. Updating Samsung related sound
drivers to use this helper.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Having the binary ones complement operator in the new bitmak value makes the
code hard to read.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When headphones are plugged in, force AFG and node 0x02
("Headphone Playback Volume") to D0 to avoid pop noises.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=76611
Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
If master clock is provided through device tree, then update
the master clock frequency during set_sysclk.
Documentation has been updated to reflect the change.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
If master clock is provided through device tree, then update
the master clock frequency during set_sysclk.
Documentation has been updated to reflect the change.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add "depends on I2C" to shut up the build errors from randconfig.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make including the omap-pcm.h outside sound/soc/omap more convenient.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Allow jack GPIO pins be defined also using GPIO descriptor-based interface
in addition to legacy GPIO numbers. This is done by adding two new fields to
struct snd_soc_jack_gpio: idx and gpiod_dev.
Legacy GPIO numbers are used only when GPIO consumer device gpiod_dev is
NULL and otherwise idx is the descriptor index within the GPIO consumer
device.
New function snd_soc_jack_add_gpiods() is added for typical cases where all
GPIO descriptor jack pins belong to same GPIO consumer device. For other
cases the caller must set the gpiod_dev in struct snd_soc_jack_gpio before
calling snd_soc_jack_add_gpios().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch does basic GPIO descriptor conversion to soc-jack. Even the GPIOs
are still passed and requested using legacy GPIO numbers the driver
internals are converted to use GPIO descriptor API.
Motivation for this is to prepare soc-jack so that it will allow registering
jack GPIO pins using both GPIO descriptors and legacy GPIO numbers.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Failure to terminate this match table can lead to boot failures
depending on where the compiler places the match table.
Signed-off-by: Stephen Boyd <sboyd@codeaurora.org>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DMAC src/dst addr needs to be set from driver when DT case.
(It was set from SoC/DMAEngine code when non-DT case)
This patch adds rsnd_gen_dma_addr() to set DMAC src/dst addr.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver is supporting to use DMAEngine.
But, DMA slave channel name "tx", "rx" is not enough
in DT case.
Becuase, it has many ports and path combination.
This patch adds rsnd_dma_of_name() to find
DMA channel name, for example
memory to SSI0 is "mem_ssi0",
SSI0 to memory is "ssi0_mem",
SSI0 to SRC0 is "ssi0_src0",
SRC0 to SSI0 is "src0_ssi0",
SRC0 to DVC0 is "src0_dvc0"...
Renesas sound want to use PIO transfer mode for some reasons.
It will be PIO tranfer mode if device node doesn't have
DMA settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver uses many modules (= SSI/SRC/DVC),
and each module had own name.
But, each module name can be used as several purpose,
like clock name, DMA name etc...
This patch uses common name for each module.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver is supporting Gen1/Gen2.
SRC probe can return error if it was unknown
generation.
Now, rsnd_src_non_ops is not needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
DT DMA support needs struct platform_device pointer,
and it can get struct device pointer from platform_device.
Save platform_device instead of device.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Driver needs to call of_node_put() after of_get_chile_by_name()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to
generate an initial jack status report. If sound card initialization
fails, that work item needs to be cancelled, so it doesn't run after the
card has been freed. Specifically, freeing the card calls
snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets
jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which
is called from the work queue item.
snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine
drivers do call this function in the platform driver remove() callback.
However, this happens after the sound card is freed, at least when the
card is freed due to errors late during snd_soc_instantiate_card(). This
leaves a window where the work item can execute after the card is freed.
In next-20140522, sound card initialization does fail for unrelated
reasons, and hits the problem described above.
To solve this, fix the Tegra ASoC machine drivers to clean up the Jack
GPIOs during the snd_soc_card's .remove() callback, which is executed
before the overall card object is freed. also, gGuard the cleanup call
based on whether we actually setup up the GPIOs in the first place.
Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove
function to match where the GPIOs get set up. However, there is no such
callback.
This change fixes all Tegra machine drivers. By code inspection, I
believe some non-Tegra machine drivers have the same issue. I'll send a
patch for that separately, once this is reviewed.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This makes sure a format string can never get processed into the worker
thread name from the device name.
Signed-off-by: Kees Cook <keescook@chromium.org>
Acked-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some platforms require that the codecs mclk is a fixed multiplication
factor of the audio stream rate. Add a optional property to the
binding to hold this factor and implement a hw_params() function to
make use of it.
Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the clock divisor and multiplier NI, MI values for audio
sampling frequencies 44100 and 48000 Hz and PCLK 19.2 MHz. This is useful
for the Odroid X2/U2 boards when the codec works in master mode and its
MCLK clock is fed from the I2S CDCLK output.
Signed-off-by: Chen Zhen <zhen1.chen@samsung.com>
[s.nawrocki@samsung.com: edited the commit description]
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In post commit, a quirk of this firmware about transactions is reported.
This commit apply a workaround for this quirk.
They often fail transactions due to gap_count mismatch. This state is changed
by generating bus reset.
The fw_schedule_bus_reset() is an exported symbol in firewire-core. But there
are no header for public. This commit moves its prototype from
drivers/firewire/core.h to include/linux/firewire.h.
This mismatch still affects bus management before generating this bus reset.
It still takes a time to call driver's probe() because transactions are still
often failed.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just powering on, these devices below wait to download firmware.
- Firewire Audiophile
- Firewire 410
- Firewire 1814
- ProjectMix I/O
But firmware version 5058 or later, flash memory in the device stores the
firmware. So this driver can enable these devices by sending a certain cue to
load the firmware.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The firmwares for M-Audio Firewire 410/1814 and ProjectMix I/O has a quirk to
ignore MIDI messages in data blocks more than 8. This commit uses a flag which
Fireworks uses for a similar quirk.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
M-Audio Firewire 1814 has a quirk, ProjectMix I/O also has. They transmit
empty packet with wrong value of dbc incremented by 8 at high sampling rate.
According to IEC 61883-1, this value should be the same as the one in
previous packet.
This commit adds a flag named as CIP_EMPTY_HAS_WRONG_DBC. With flag, the value
of dbc in empty packet is overwittern by an expected value.
This is an example of this quirk:
CIP Header 0 CIP Header 1 Payload size
010D0000 9004F759 210
010D0010 90040B59 210
010D0020 90042359 210
01020028 9004FFFF 2 <-
010D0030 90043759 210
010D0040 90044B59 210
010D0050 90046359 210
01020058 9004FFFF 2 <-
010D0060 90047759 210
010D0070 90048B59 210
010D0080 9004A359 210
01020088 9004FFFF 2 <-
010D0090 9004B759 210
010D00A0 9004CB59 210
010D00B0 9004E359 210
010200B8 9004FFFF 2 <-
010D00C0 9004F759 210
010D00D0 90040B59 210
010D00E0 90042359 210
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support some models which M-Audio produces
with DM1000 but its firmware is special. They are:
- Firewire 1814
- ProjectMix I/O
They have heavily customized firmware. The usual operations can't be applied to
them. For this reason, this commit adds a model specific member to 'struct
snd_bebob' and some model specific functions. Some parameters are write-only so
this commit also adds control interface for applications to set them.
M-Audio special firmware quirks:
- Just after powering on, they wait to download firmware. This state is
changed when receiving cue. Then bus reset is generated and the device is
recognized as a different model with the uploaded firmware.
- They don't respond against BridgeCo AV/C extension commands. So drivers
can't get their stream formations and so on.
- They do not start to transmit packets only by establishing connection but
also by receiving SIGNAL FORMAT command.
- After booting up, they often fail to send response against driver's request
due to mismatch of gap_count.
This module don't support to upload firmware.
Tested-by: Darren Anderson <darrena092@gmail.com> (ProjectMix I/O)
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support some models which M-Audio produces
with DM1000/DM1000E with usual firmware. They are:
- Firewire 410
- Firewire AudioPhile
- Firewire Solo
- Ozonic
- NRV10
- FirewireLightBridge
According to a person who worked in BridgeCo, some models are produced with
'Pre-BeBoB'. This means that these products were released before BeBoB was
officially produced, and later BeBoB specification was formed. So these models
have some quirks.
M-Audio usual firmware quirks:
- Just after powering on, 'Firewire 410' waits to download firmware. This
state is changed when receiving cue. Then bus reset is generated and the
device is recognized as a different model with the uploaded firmware.
- 'Firewire Audiophile' also waits to download firmware but its
vendor id/model id is the same as the one after loading firmware.
- The information of channel mapping for MIDI conformant data channel is
invalid against BridgeCo specification.
This commit adds some codes for these quirks but don't support to upload
firmware.
This commit also adds specific operations to get metering information. The
metering information also includes status of clock synchronization if the model
supports to switch source of clock.
The specification of FirewireLightBridge is unknown. So in this time, normal
operations are applied for this model.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support all of models which Focusrite
produces with DM1000/BeBoB. They are:
- Saffire
- Saffire LE
- SaffirePro 10 I/O
- SaffirePro 26 I/O
This commit adds Focusrite specific operations:
1. Get source of clock
2. Get/Set sampling frequency
3. Get metering information
The driver uses these functionalities to read/write specific address by async
transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support all of models which Yamaha produced
with DM1000/BeBoB. They are:
- GO44
- GO46
This commit adds Yamaha specific operations. To get source of clock, AV/C Audio
Subunit command is used.
I note that their appearances are similar to some models of TerraTec; 'Go44' is
similar to 'PHASE 24 FW' and 'GO46' is similar to 'PHASE X24 FW'. But their
combination of Audio/Music subunits is a bit different.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support all of models which Terratec produced
with DM1000/BeBoB. They are:
- PHASE 24 FW
- PHASE X24 FW
- PHASE 88 Rack FW
- EWS MIC2
- EWS MIC4
- Aureon 7.1 Firewire
For Phase series, this commit adds a Terratec specific operation. To get source
of clock. AV/C Audio Subunit command is used.
For EWS series and Aureon, this module uses normal operations.
Tested-by: Maximilian Engelhardt <maxi@daemonizer.de> (PHASE 24 FW)
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is for some devices which have its own operations or quirks.
Many functionality should be implemented in user land. Then this commit adds
functionality related to stream such as sampling frequency or clock source. For
help to debug, this commit adds the functionality to get metering information
if it's available.
To help these functionalities, this commit adds some AV/C commands defined in
'AV/C Audio Subunit Specification (1394TA).
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This interface is designed for mixer/control application. By using hwdep
interface, the application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to capture/playback PCM samples.
When AMDTP stream is already running for PCM or the source of clock is not
internal, available sampling rate is limited at current one.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to capture/playback MIDI messages.
When no AMDTP streams are running, this module starts AMDTP stream at current
sampling rate for MIDI substream.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds proc interface to get these information for debugging:
- firmware information
- stream formation
- current clock source and sampling rate
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Normal BeBoB firmware has a quirk. When receiving bus reset, it transmits
packets with discontinuous value in dbc field.
This causes two situation, one is to abort streaming by firewire-lib as a
result of detecting the discontinuity. Another is to call driver's .update()
because of bus reset. These two is generated independently. (The former
depends on isochronous stream and the latter depends on IEEE1394 bus driver.)
When BeBoB driver works with XRUN-recoverable applications, this situation
looks like stream_start_duplex() call followed by stream_update_duplex() call
because applications will call snd_pcm_prepare() immediately at XRUN.
To update connections and streams at first, this commit use completion. When
queueing error occurs, stream_start_duplex() is forced to wait maximum
1000msec. During this, when .update() is called, the completion is waken and
stream_start_duplex() is processed without breaking connections.
At bus reset, stream_start_duplex() shouldn't break/establish connections and
stream_update_duplex() should update connections because a caller of
fw_iso_resources_allocate() is responsible for calling
fw_iso_resources_update() on bus reset.
This commit also adds a flag, which has an effect to skip checking continuity
for first packet. This flag is useful for BeBoB quirk to start handling packets
during streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds management functionality for connections and streams.
BeBoB uses CMP to manage connections and uses AMDTP for streams.
This commit also adds some BridgeCo's AV/C extension commands. There are some
BridgeCo's AV/C extension commands but this commit just uses below commands
to get device's capability and status:
1.Extended Plug Info commands
- Plug Channel Position Specific Data
- Plug Type Specific Data
- Cluster(Section) Info Specific Data
- Plug Input Specific Data
2.Extended Stream Format Information commands
- Extended Stream Format Information Command - List Request
For Extended Plug Info commands for Cluster Info Specific Data, I pick up
'section' instead of 'cluster' from document to prevent from misunderstanding
because 'cluster' is also used in IEC 61883-6.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds two functionality for hwdep interface, adds two parameters for
this driver, add a node for proc interface.
To receive responses from devices, this driver already allocate own callback
into initial memory space in host controller. This means no one can allocate
its own callback to the address. So this driver must give a way for user
applications to receive responses.
This commit adds a functionality to receive responses via hwdep interface. The
application can receive responses to read from this interface. To achieve this,
this commit adds a buffer to queue responses. The default size of this buffer is
1024 bytes. This size can be changed to give preferrable size to
'resp_buf_size' parameter for this driver. The application should notice rest
of space in this buffer because this driver don't push responses when this
buffer has no space.
Additionaly, this commit adds a functionality to transmit commands via hwdep
interface. The application can transmit commands to write into this interface.
I note that the application can transmit one command at once, but can receive
as many responses as possible untill the user-buffer is full.
When using these interfaces, the application must keep maximum number of
sequence number in command within the number in firewire.h because this driver
uses this number to distinguish the response is against the command by the
application or this driver.
Usually responses against commands which the application transmits are pushed
into this buffer. But to enable 'resp_buf_debug' parameter for this driver, all
responses are pushed into the buffer. When using this mode, I reccomend to
expand the size of buffer.
Finally this commit adds a new node into proc interface to output status of the
buffer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This interface is designed for mixer/control application. To use hwdep
interface, the application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to capture/playback PCM samples.
When AMDTP stream is already running for PCM or the source of clock is not
internal, available sampling rate is limited at current one.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks has a quirk to ignore MIDI messages in data blocks more than 8.
This commit adds a flag for this quirk and codes to skip 8 or more data
blocks to transfer MIDI messages.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to capture/playback MIDI messages.
When no AMDTP streams are running, this driver starts AMDTP stream for MIDI
stream at current sampling rate.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks has a quirk to reset data block counter at bus reset.
This commit adds a flag of CIP_SKIP_DBC_ZERO_CHECK. This flag has an effect
to skip checking dbc continuity when dbc is zero.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks firmware version 5.5 reports fix interval for dbc in each packet.
For example, AudioFire4:
CIP0 CIP1 Payload
00070000 900484FF 72
00070008 9004A8FF 72
00070008 90FFFFFF 02
00070010 9004D0FF 72
00070018 9004C4FF 72
00070020 9004E8FF 72
00070020 90FFFFFF 02
00070028 900410FE 72
The interval of each dbc should be 16 except for empty packet but it's still 8.
This commit adds a flag for this quirk and codes to refer to a fixed value.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One of Fireworks firmware, named as 'AudioFire9', seems to transmit
packets with wrong value of dbs. It's always 0x11 but actual size of
data block is different.
This commit adds a flag for this quirk and some codes to calculate
correct size.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks has a quirk for the value of dbc field in transmitted packets.
For Fireworks, dbc means the end of events in current packet. This is out
of specification.
For example, AudioFire4:
CIP0 CIP1 Payload
01070092 90FFFFFF 02
0107009A 9001E17B 3A <-
010700A2 9001F6E5 3A
010700A2 90FFFFFF 02
010700AA 9001104F 3A <-
010700B2 900125B9 3A
010700BA 90013B23 3A
010700BA 90FFFFFF 02
010700C2 9001548E 3A <-
010700CA 900169F8 3A
010700CA 90FFFFFF 02
010700D2 90018362 3A <-
010700DA 900198CC 3A
According to IEC 61883-1/6, a packet following to empty packet has the same
value for its dbc. But for Fireworks, it's incremented and empty packet has
the same value as previous packet in dbc field.
This commit adds a flag for Fireworks and some codes to checking dbc continuity.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks has a quirk to transmit empty packets with TAG0. This commit
adds handling this quirk for full duplex stream synchronization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks manages connections by CMP and can transmit/receive AMDTP streams
with a few quirks. This commit adds functionality to start/stop the streams.
Major Fireworks products don't support 'SYT-Match' clock source mode, except
for AudioFire12/8(till 2009 July) with firmware version 1.0. Already in
previous commit, this driver don't support such old firmwares. So this commit
adds support for non 'SYT-Match' clock source modes.
I note that this driver has a short gap for MIDI streams when starting PCM
stream. When AMDTP streams are running only for MIDI data and PCM data is
going to be joined at different sampling rate, then AMDTP streams are
stopped once and started again after changing sampling rate.
Unfortunately, Fireworks is not fully compliant to IEC 61883-1/6. Some commits
following to this commit add these quirks.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks uses own command and response. This commit adds functionality to
transact and adds some commands required for sound card instance and kernel
streaming.
There are two ways to deliver substance of this transaction:
1.AV/C vendor dependent command for command/response
2.Async transaction to specific addresses for command/response
By way 1, I confirm AudioFire12 cannot correctly response to some commands with
firmware version 5.0 or later. This is also confirmed by FFADO. So this driver
implement way 2.
The address for response gives an issue. When this driver allocate own callback
function into the address, then no one can allocate its own callback function.
This situation is not good for applications in user-land. This issue is solved
in later commit.
I note there is a command to change the address for response if the device
supports. But this driver uses default value. So users should not execute this
command as long as hoping this driver works correctly.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a new driver for devices based on Fireworks. This driver
just creates/removes card instance according to callbacks.
Fireworks is a board module which Echo Audio produced. This module
consists of three chipsets:
- Communication chipset for IEEE1394 PHY/Link and IEC 61883-1/6
- DSP or/and FPGA for signal processing
- Flash Memory to store firmwares
Current supported devices:
- Mackie Onyx 400F/1200F
- Echo AudioFire12/8(until 2009 July)
- Echo AudioFire2/4/Pre8/8(since 2009 July)
- Echo Fireworks 8/HDMI
- Gibson Robot Interface pack/GoldTop
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds three commands, which may be used by some firewire device
drivers. These commands are defined in 'AV/C Digital Interface Command Set
General Specification Version 4.2 (2004006, 1394TA)'.
1. PLUG INFO command (clause 10.1)
2. INPUT PLUG SIGNAL FORMAT command (clause 10.10)
3. OUTPUT PLUG SIGNAL FORMAT command (clause 10.11)
By the command 1, the drivers can get the number of plugs for AV/C unit or
subunit.
By the command 2 and 3, the drivers can get/set sampling frequency.
The 'firewire-speakers' already uses INPUT PLUG SIGNAL FORMAT command to set
sampling rate. So this commit also affects the driver.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices based on BeBoB use this type of AV/C transaction.
'Deferred Transaction' is defined in 'AV/C Digital Interface Command Set
General Specification' and is used by targets to make a response deferred
during processing it.
If a target may not be able to complete a command within 100msec since
receiving the command, then the target shall return INTERIM response,
to which final response will follow later. CONTROL/NOTIFY commands are
allowed for deferred transaction.
In the specification, devices allow to send INTERIM response just one time.
But this commit allows to handle several INTERIM response with two reasons.
One reason is to simplify codes, and another reason is to prepare for
devices which is out of specification.
There is an issue. In the specification, the interval between INTERIM
response and final response is 'Unspecified interval'. The specification
depends on each subunit specification for this interval.
But we promise to finish this function for caller. In this reason, I use
FCP_TIMEOUT_MS for this interval. Currently it's 125msec. When we find
devices which needs more time for this interval, then let us add some codes
to apply more interval for 'Unspecified interval'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Plug Control Registers have two fields related to the number of established
connections, one is 'Broadcast connection counter' and another is
'Point-to-point connection counter'. The driver can know there are established
connections or not to check these fields.
This commit is for considering about JACK/FFADO streaming. Currently, when
JACK/FFADO starts its streaming to the device, cmp_connection_establish() is
failed expectedly. This seems to be enough but there are some devices which
needs to change sampling frequency before trying to establish connections.
For such devices, this functionality is needed.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds some macros, codes with condition of direction and new functions
to handle output connection. Once cmp_connection_init() is executed with its
direction, CMP input and output connection can be handled by the same way.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds 'direction' member to 'cmp_connection' structure to indicate
the direction of connection. This patch also adds 'direction' argument to
cmp_connection_init() function to determine the direction.
The cmp_connection_init() function is exported and used in snd-firewire-speakers
so this patch also affect it.
This patch just add them. Actual implementation will be done by followed
patches.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
