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18892 Commits (c1c3f2c906e35bcb6e4cdf5b8e077660fead14fe)

Author SHA1 Message Date
Mark Brown d8188f00e7 ASoC: intel: The Baytrail/MAX98090 driver depends on I2C
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-01 20:12:05 +01:00
Oder Chiou d92950e755 ASoC: rt5640: Add the function "get_clk_info" to RL6231 shared support
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>
2014-06-01 20:04:30 +01:00
Oder Chiou 71c7a2d675 ASoC: rt5640: Add the function of the PLL clock calculation to RL6231 shared support
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>
2014-06-01 20:04:30 +01:00
Oder Chiou 49ef7925c2 ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651
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>
2014-06-01 20:04:30 +01:00
Mark Brown 15f78ea67f Merge branches 'topic/rt5640', 'topic/rt5645' and 'topic/rt5651' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-rl6231 2014-06-01 20:04:24 +01:00
Xiubo Li b59dce53ef ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error.
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>
2014-06-01 20:02:17 +01:00
Dan Carpenter 33a5f989de ALSA: bebob: sizeof() vs ARRAY_SIZE() typo
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>
2014-06-01 18:16:04 +02:00
Mark Brown 287d414eac Merge branch 'topic/fsl' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-fsl-ssi
Conflicts:
	sound/soc/fsl/Kconfig
2014-06-01 14:02:07 +01:00
Matt Reimer a7f0b839cb ASoC: wm9713: correct mono out PGA sources
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>
2014-06-01 13:52:51 +01:00
Rickard Strandqvist 14577c2516 ALSA: synth: emux: soundfont.c: Cleaning up memory leak
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>
2014-06-01 14:33:09 +02:00
Alexander Shiyan 7b8751abdd ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320
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>
2014-06-01 12:00:22 +01:00
Markus Pargmann 4324812201 ASoC: fsl-ssi: Use regmap
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>
2014-06-01 11:55:08 +01:00
Markus Pargmann 737a6b418a ASoC: fsl-ssi: reorder and document fsl_ssi_private
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>
2014-06-01 11:55:08 +01:00
Markus Pargmann d429d8e332 ASoC: fsl-ssi: Fix baudclock handling
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>
2014-06-01 11:55:08 +01:00
Sascha Hauer b5dd91b3dc ASoC: fsl-ssi: Set framerate divider correctly for i2s master mode
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>
2014-06-01 11:55:08 +01:00
Sascha Hauer d8ced4793f ASoC: fsl-ssi: remove unnecessary spinlock
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>
2014-06-01 11:55:07 +01:00
Sascha Hauer 8dd51e23a1 ASoC: fsl-ssi: set bitclock in master mode from hw_params
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>
2014-06-01 11:55:07 +01:00
Markus Pargmann 85e59af240 ASoC: fsl-ssi: make fsl,mode property optional
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>
2014-06-01 11:55:07 +01:00
Sascha Hauer fcdbadef37 ASoC: fsl-ssi: introduce SoC specific data
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>
2014-06-01 11:55:07 +01:00
Jarkko Nikula 4af72f4e69 ASoC: Intel: byt-rt5640: Use card PM ops from core
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>
2014-06-01 11:50:45 +01:00
Jarkko Nikula 8eb776ab17 ASoC: Intel: Use devm_snd_soc_register_card
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>
2014-06-01 11:50:45 +01:00
Andy Shevchenko a018c28550 ASoC: Intel: remove duplicate headers
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>
2014-06-01 11:50:45 +01:00
Jarkko Nikula 58dcc48816 ASoC: Intel: Clear stored Baytrail DSP DMA pointer before stream start
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>
2014-06-01 11:50:45 +01:00
Axel Lin 4641c771b6 ASoC: cs42l56: Fix new value argument in snd_soc_update_bits calls
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>
2014-06-01 11:49:25 +01:00
Imre Deak 9cf0e4520d ASoC: Intel: byt/hsw: Add missing kthread_stop to error/cleanup path
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>
2014-06-01 11:44:49 +01:00
Jarkko Nikula 9b351d4689 ASoC: Intel: Add Baytrail byt-max98090 machine driver
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>
2014-06-01 11:44:49 +01:00
Peter Ujfalusi e6c111fac4 ASoC: tlv320aci3x: Fix custom snd_soc_dapm_put_volsw_aic3x() function
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
2014-06-01 11:43:02 +01:00
Takashi Iwai 112cddcada ALSA: firewire: Fix dependency on PCM and rawmidi
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>
2014-05-30 15:22:06 +02:00
Takashi Iwai 598e306184 ALSA: hda/analog - Fix silent output on ASUS A8JN
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>
2014-05-30 12:07:12 +02:00
Paul Bolle 66470c973c ALSA: gus: remove checks for CONFIG_SND_DEBUG_ROM
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>
2014-05-30 10:12:10 +02:00
Paul Bolle 55d0cc2998 sound: remove checks for CONFIG_BCM_CS4297A_CSWARM
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>
2014-05-30 10:11:55 +02:00
Daniel Matuschek 06109f47f2 ASoC: wm8804: Allow control of master clock divider in PLL generation
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>
2014-05-29 16:01:56 +01:00
Hui Wang 532895c58c ALSA: hda - move some alc662 family machines to hda_pin_quirk table
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-29 15:59:43 +02:00
Hui Wang d91a4c1be0 ALSA: hda - move some alc269 family machines to hda_pin_quirk table
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-29 15:59:35 +02:00
Hui Wang 37df09492c Revert "ALSA: hda - drop def association and sequence from pinconf comparing"
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>
2014-05-29 15:59:28 +02:00
Dan Carpenter 396178370b ALSA: fireworks: small leak on error path
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>
2014-05-29 15:56:18 +02:00
Dan Carpenter aeebbddda7 ALSA: fireworks: remove some stray checks
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>
2014-05-29 15:56:02 +02:00
Benoit Taine 82285f254c ALSA: au1x00: Use resource_size instead of computation
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>
2014-05-28 17:50:57 +02:00
Lars-Peter Clausen cb07ef36fe ASoC: Blackfin: ADAU1X81 eval board support
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>
2014-05-27 20:54:51 +01:00
Lars-Peter Clausen 5dcdbee9cf ASoC: Blackfin: ADAU1X61 eval board support
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>
2014-05-27 20:54:51 +01:00
Lars-Peter Clausen 2923af0246 ASoC: Add ADAU1381/ADAU1781 audio CODEC support
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>
2014-05-27 20:54:51 +01:00
Lars-Peter Clausen dab464b60b ASoC: Add ADAU1361/ADAU1761 audio CODEC support
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>
2014-05-27 20:54:50 +01:00
Lars-Peter Clausen 4101866c74 ASoC: Add ADAU1X61 and ADAU1X81 CODECs common code
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>
2014-05-27 20:54:50 +01:00
Takashi Iwai a58bdba749 Merge branch 'topic/firewire' into for-next
This is a merge of big firewire audio stack updates by Takashi Sakamoto.
2014-05-27 17:38:08 +02:00
Takashi Sakamoto 51fa31d462 ALSA: bebob: Improve comments about stream format
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>
2014-05-27 17:36:24 +02:00
Takashi Sakamoto 7862126a4f ALSA: bebob: Remove meaningless mutex_unlock()
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>
2014-05-27 17:36:11 +02:00
Takashi Sakamoto 9fb01cdb38 ALSA: bebob: Add static specifier to identifier with file scope
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>
2014-05-27 17:36:01 +02:00
Takashi Sakamoto 791c67b427 ALSA: bebob: Use different names for identifiers in the same file
To suppress 'sparse' warning.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-27 17:35:48 +02:00
Takashi Sakamoto 73616c4eec ALSA: fireworks/bebob: Improve indentation
According to coding rule.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-27 17:35:38 +02:00
Takashi Sakamoto 9b5f0edfd2 ALSA: fireworks/bebob: Add suffix for long long integer literal
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>
2014-05-27 17:35:30 +02:00
Takashi Sakamoto a6b598bf4b ALSA: fireworks/bebob: Use the same type of variables as function parameters
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>
2014-05-27 17:35:22 +02:00
Takashi Sakamoto 4a286d5528 ALSA: fireworks/bebob: Change type of argument for sampling rate
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>
2014-05-27 17:35:13 +02:00
Takashi Sakamoto 93219d0649 ALSA: fireworks: Use the same prototype for functions as actual declaration
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>
2014-05-27 17:35:04 +02:00
Takashi Sakamoto ba06b2cbad ALSA: fireworks: Fix wrong value as argument for PTR_ERR()
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>
2014-05-27 17:34:52 +02:00
Takashi Sakamoto 51212eea4f ALSA: firewire-lib: Fix sparse warning of incorrect type in assignment
__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>
2014-05-27 17:34:37 +02:00
Takashi Sakamoto f9503a68fb ALSA: firewire-lib: Use ARRAY_SIZE() instead of sizeof() for correct loop limit
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>
2014-05-27 17:34:27 +02:00
Charles Keepax 62c35b3bd2 ASoC: wm_adsp: Use adsp_err/warn instead of dev_err/warn
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>
2014-05-27 16:08:42 +01:00
Fabio Estevam 29aa37cddf ASoC: sgtl5000: Fix the cache handling
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>
2014-05-27 12:22:15 +01:00
Fabian Frederick 00a6d7b676 ALSA: sound/aoa/codecs/onyx.c: use static const for texts
'texts' is only used as source in strcpy

Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-27 11:58:55 +02:00
Arnd Bergmann 16c2395203 ALSA: hda: fix tegra build
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>
2014-05-27 07:36:18 +02:00
Tushar Behera 88ce1465ec ASoC: samsung: Use params_width()
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>
2014-05-26 17:04:20 +01:00
Axel Lin 772bc594da ASoC: sirf-audio-codec: Simplify the new bitmask value in regmap_update_bits
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>
2014-05-26 17:00:39 +01:00
Gabriele Mazzotta 033b0a7ca9 ALSA: hda - Pop noises fix for XPS13 9333
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>
2014-05-26 17:47:12 +02:00
Lars-Peter Clausen 2896b8b4d8 ASoC: davinci-evm: Replace instances of rtd->codec->card with rtd->card
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>
2014-05-26 16:34:55 +01:00
Tushar Behera e3048c3d2b ASoC: max98095: Add master clock handling
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>
2014-05-26 16:18:59 +01:00
Tushar Behera b10ab7b838 ASoC: max98090: Add master clock handling
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>
2014-05-26 16:16:54 +01:00
Takashi Iwai 5dc04f51c1 ASoC: alc5623: Fix Kconfig dependency
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>
2014-05-26 16:10:59 +01:00
Jyri Sarha 87c1936426 ASoC: omap-pcm: Move omap-pcm under include/sound
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>
2014-05-26 15:32:32 +01:00
Mark Brown 35bcc3c20d Merge branch 'topic/davinci' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-omap 2014-05-26 15:31:40 +01:00
Jarkko Nikula f025d3b9c6 ASoC: jack: Add support for GPIO descriptor defined jack pins
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>
2014-05-26 15:26:00 +01:00
Jarkko Nikula 50dfb69d1b ASoC: jack: Basic GPIO descriptor conversion
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>
2014-05-26 15:23:14 +01:00
Stephen Boyd 4c715c758c ASoC: pxa: pxa-ssp: Terminate of match table
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>
2014-05-26 14:38:50 +01:00
Kuninori Morimoto ad32d0c7b0 ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr
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>
2014-05-26 14:34:56 +01:00
Kuninori Morimoto 199e7688bd ASoC: rsnd: care DMA slave channel name for DT
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>
2014-05-26 14:34:55 +01:00
Kuninori Morimoto 8aefda5046 ASoC: rsnd: module name is unified
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>
2014-05-26 14:34:55 +01:00
Kuninori Morimoto 033e7ed85b ASoC: rsnd: remove rsnd_src_non_ops
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>
2014-05-26 14:34:55 +01:00
Kuninori Morimoto 9f464f8e07 ASoC: rsnd: save platform_device instead of device
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>
2014-05-26 14:34:54 +01:00
Kuninori Morimoto f451e48d8e ASoC: rsnd: DT node clean up by using the of_node_put()
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>
2014-05-26 14:34:54 +01:00
Stephen Warren fb6b8e7144 ASoC: tegra: free jack GPIOs before the sound card is freed
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>
2014-05-26 14:32:34 +01:00
Kees Cook 3538632089 ASoC: Intel: avoid format string leak to thread name
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>
2014-05-26 14:31:04 +01:00
Andrew Lunn 2942a0e285 ASoC: simple-card: Support setting mclk via a fixed factor
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>
2014-05-26 14:29:30 +01:00
Chen Zhen 2c81a10ae6 ASoC: max98090: Add NI/MI values for user pclk 19.2 MHz
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>
2014-05-26 14:28:57 +01:00
Fabio Estevam b20e53a826 ASoC: fsl_ssi: Add suspend/resume support
Doing a suspend/resume sequence while playing an audio track in the backgroung
causes broken audio right after resume:

root@freescale /$ aplay clarinet.wav &

root@freescale /home$ Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian,
 Rate 44100 Hz, Mono

root@freescale /home$ echo mem > /sys/power/state
PM: Syncing filesystems ... done.
Freezing user space processes ... (elapsed 0.002 seconds) done.
Freezing remaining freezable tasks ... (elapsed 0.002 seconds) done.
Suspending console(s) (use no_console_suspend to debug)
PM: suspend of devices complete after 37.082 msecs
PM: suspend devices took 0.040 seconds
PM: late suspend of devices complete after 4.234 msecs
PM: noirq suspend of devices complete after 4.618 msecs
Disabling non-boot CPUs ...
PM: noirq resume of devices complete after 4.013 msecs
PM: early resume of devices complete after 4.000 msecs
PM: resume of devices complete after 68.907 msecs
PM: resume devices took 0.070 seconds
Restarting tasks ... Suspended. Trying resume. Failed. Restarting stream. Done.
Suspended. Trying resume. Failed. Restarting stream. Done.
Suspended. Trying resume. Failed. Restarting stream. Done.
Suspended. Trying resume. Failed. Restarting stream. Done.
Suspended. Trying resume. Failed. Restarting stream. Done.
Suspended. Trying resume. Failed. Restarting stream. Done.
Suspended. Trying resume. Failed. Restarting stream. Done.
....

Add SNDRV_PCM_TRIGGER_RESUME/SUSPEND cases so that we can gracefully handle
system suspend/resume.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Shawn Guo <shawn.guo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-26 14:24:24 +01:00
Takashi Sakamoto 9b1ee0b2cb ALSA: firewire/bebob: Add a workaround for M-Audio special Firewire series
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>
2014-05-26 14:33:10 +02:00
Takashi Sakamoto a2b2a7798f ALSA: bebob: Send a cue to load firmware for M-Audio Firewire series
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>
2014-05-26 14:32:58 +02:00
Takashi Sakamoto c495a4a36e ALSA: bebob: Add a quirk of data blocks for MIDI messages for some M-Audio devices
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>
2014-05-26 14:32:46 +02:00
Takashi Sakamoto 9d59124cac ALSA: bebob/firewire-lib: Add a quirk of wrong dbc in empty packet for M-Audio special Firewire series
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>
2014-05-26 14:32:33 +02:00
Takashi Sakamoto 3149ac489f ALSA: bebob: Add support for M-Audio special Firewire series
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>
2014-05-26 14:32:21 +02:00
Takashi Sakamoto 9076c22ddd ALSA: bebob: Add support for M-Audio usual Firewire series
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>
2014-05-26 14:32:03 +02:00
Takashi Sakamoto 25784ec2d0 ALSA: bebob: Add support for Focusrite Saffire/SaffirePro series
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>
2014-05-26 14:31:50 +02:00
Takashi Sakamoto 8ac98a3585 ALSA: bebob: Add support for Yamaha GO series
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>
2014-05-26 14:31:38 +02:00
Takashi Sakamoto 326b9cacf4 ALSA: bebob: Add support for Terratec PHASE, EWS series and Aureon
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>
2014-05-26 14:31:25 +02:00
Takashi Sakamoto 1fc9522a08 ALSA: bebob: Prepare for device specific operations
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>
2014-05-26 14:31:15 +02:00
Takashi Sakamoto 618eabeae7 ALSA: bebob: Add hwdep interface
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>
2014-05-26 14:31:03 +02:00
Takashi Sakamoto fbbebd2c40 ALSA: bebob: Add PCM interface
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>
2014-05-26 14:30:46 +02:00
Takashi Sakamoto 248b78027d ALSA: bebob: Add MIDI interface
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>
2014-05-26 14:30:16 +02:00
Takashi Sakamoto ad9697bad7 ALSA: bebob: Add proc interface for debugging purpose
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>
2014-05-26 14:30:00 +02:00
Takashi Sakamoto b6bc812327 ALSA: bebob/firewire-lib: Add a quirk for discontinuity at bus reset
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>
2014-05-26 14:29:44 +02:00
Takashi Sakamoto eb7b3a056c ALSA: bebob: Add commands and connections/streams management
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>
2014-05-26 14:29:29 +02:00
Takashi Sakamoto fd6f4b0dc1 ALSA: bebob: Add skelton for BeBoB based devices
This commit adds a new driver for BeBoB based devices with no specific
operations. Currently this driver just create/remove card instance according
to callbacks.

BeBoB is 'BridgeCo enhanced Breakout Box'. This is installed to firewire
devices with DM1000/DM1100/DM1500 chipset. It gives common way for host
system to handle BeBoB based devices.

Current supported devices:
 - Edirol FA-66/FA-101
 - PreSonus FIREBOX/FIREPOD/FP10/Inspire1394
 - BridgeCo RDAudio1/Audio5
 - Mackie Onyx 1220/1620/1640 (Firewire I/O Card)
 - Mackie d.2 (Firewire Option)
 - Stanton FinalScratch 2 (ScratchAmp)
 - Tascam IF-FW DM
 - Behringer XENIX UFX 1204/1604
 - Behringer Digital Mixer X32 series (X-UF Card)
 - Apogee Rosetta 200/Rosetta 400 (X-FireWire card)
 - Apogee DA-16X/AD-16X/DD-16X (X-FireWire card)
 - Apogee Ensemble
 - ESI Quotafire610
 - AcousticReality eARMasterOne
 - CME MatrixKFW
 - Phonix Helix Board 12 MkII/18 MkII/24 MkII
 - Phonic Helix Board 12 Universal/18 Universal/24 Universal
 - Lynx Aurora 8/16 (LT-FW)
 - ICON FireXon
 - PrismSound Orpheus/ADA-8XR

Devices possible to be supported if identifying IDs:
 - Apogee Mini-Me Firewire/Mini-DAC Firewire
 - Behringer F-Control Audio 610/1616
 - Cakewalk Sonar Power Studio 66
 - CME UF400e
 - ESI Quotafire XL
 - Infrasonic DewX/Windy6
 - Mackie Digital X Bus x.200/400
 - Phonic Helix Board 12/18/24
 - Phonic FireFly 202/302
 - Rolf Spuler Firewire Guitar

Tested-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:29:12 +02:00