R-Car sound uses Audio DMAC and Audio DMAC peri peri.
Audio DMAC peri peri transfers data inside circuit.
DMA transfer needs source / destination address,
and destination address can be set via dmaengine_slave_config().
The source address can be set when starting DMAEngine.
Because Audio DMAC peri peri always ignores its value,
current driver always used same source address for
Audio DMAC / Audio DMAC peri peri
(Audio DMAC peri peri source / destination address
is always fixed value)
But, This is not good match for DT booting.
This patch properly uses DMA start address
for Audio DMAC / Audio DMAC peri peri.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
