Commit Graph

18921 Commits

Author SHA1 Message Date
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
Takashi Sakamoto 555e8a8f7f ALSA: fireworks: Add command/response functionality into hwdep interface
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>
2014-05-26 14:28:58 +02:00
Takashi Sakamoto 594ddced82 ALSA: fireworks: Add hwdep interface
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>
2014-05-26 14:28:41 +02:00
Takashi Sakamoto aa02bb6e60 ALSA: fireworks: 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 14:28:27 +02:00
Takashi Sakamoto 53111cdc53 ALSA: fireworks/firewire-lib: Add a quirk of data blocks for MIDI in out-stream
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>
2014-05-26 14:28:14 +02:00
Takashi Sakamoto a63d3ff105 ALSA: fireworks: Add MIDI interface
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>
2014-05-26 14:28:01 +02:00
Takashi Sakamoto 6a22683e89 ALSA: fireworks: Add proc interface for debugging purpose
This commit adds proc interface to output infomation for debugging.
 - firmware information
 - sampling rate and clock source
 - physical metering (linear value)

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:27:47 +02:00
Takashi Sakamoto b84b1a27b4 ALSA: fireworks/firewire-lib: Add a quirk to reset data block counter at bus reset
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>
2014-05-26 14:26:44 +02:00
Takashi Sakamoto d9cd0065c8 ALSA: fireworks/firewire-lib: Add a quirk for fixed interval of reported dbc
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>
2014-05-26 14:25:15 +02:00
Takashi Sakamoto 697022391e ALSA: fireworks/firewire-lib: Add a quirk for wrong dbs in tx packets
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>
2014-05-26 14:25:00 +02:00
Takashi Sakamoto c8bdf49b99 ALSA: fireworks/firewire-lib: Add a quirk for the meaning of dbc
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>
2014-05-26 14:24:47 +02:00
Takashi Sakamoto 7ab566453f ALSA: fireworks/firewire-lib: Add a quirk for empty packet with TAG0
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>
2014-05-26 14:24:33 +02:00
Takashi Sakamoto 315fd41fe9 ALSA: fireworks: Add connection and stream management
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>
2014-05-26 14:24:19 +02:00
Takashi Sakamoto bde8a8f23b ALSA: fireworks: Add transaction and some commands
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>
2014-05-26 14:24:03 +02:00
Takashi Sakamoto b5b0433601 ALSA: fireworks: Add skelton for Fireworks based devices
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>
2014-05-26 14:23:36 +02:00
Takashi Sakamoto 1017abed18 ALSA: firewire-lib: Add some AV/C general commands
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>
2014-05-26 14:23:13 +02:00
Takashi Sakamoto 00a7bb81c2 ALSA: firewire-lib: Add support for deferred transaction
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>
2014-05-26 14:22:56 +02:00
Takashi Sakamoto b04479fb85 ALSA: firewire-lib: Add a new function to check others' connection
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>
2014-05-26 14:22:46 +02:00
Takashi Sakamoto 44aff6980a ALSA: firewire-lib: Add handling output connection by CMP
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>
2014-05-26 14:22:37 +02:00
Takashi Sakamoto c68a1c6584 ALSA: firewire-lib: Add 'direction' member to 'cmp_connection' structure
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>
2014-05-26 14:22:14 +02:00
Takashi Sakamoto a7fa0d047f ALSA: firewire-lib: Rename macros, variables and functions for CMP
Referring to IEC 61883-1, oMPR and iMPR, oPCR and iPCR have some fields with
the same role in the same position. This patch renames some macros, variables
and function arguments with "i" in its prefix to reuse them between oMPR and
iMPR, oPCR and iPCR.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:21:57 +02:00
Takashi Sakamoto c8de6dbbbb ALSA: firewire-lib: Restrict calling flush_context_completion() when context exists
Currently, drivers can bring XRUN state for PCM substreams when error to
queue packets or detecting discontinuity of packet. The application may try to
recover this state by calling snd_pcm_prepare().

Depending on each driver, .prepare() includes restart streaming. Then there
is a state that PCM substreams are running but isochronous contexts are
stopped. In this case, when .pointer() is called, it refers to error pointer.

This commit is for a prevention of this bug.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:21:56 +02:00
Takashi Sakamoto 7b2d99fa6b ALSA: firewire-lib/dice/speakers: Add common PCM constraints for AMDTP streams
This commit adds common PCM constraints according to current firewire-lib
implementation.

1.Maximum width for each sample is limited by 24.
AM824 in IEC 61883-6 can deliver 24bit data.

2. Minimum time for period is 5msec.
Apply the old value. For shorter latency, further works are needed.

3. In blocking mode, frames per period/buffer is aligned to 32.
Each packet can include some frames depending on its sampling rate. In
blocking mode, the number equals to SYT_INTERVAL. Currently firewire-lib
can schedule snd_pcm_period_elapsed() for each packet. So, for accurate
PCM interrupt, the number of frames per period/buffer should be aligned
to SYT_INTERVAL.
Currently firewire-lib is lack of better rules to achieve this. So LCM of
each value of SYT_INTERVALs (=32) is applied. This can be improved for
further work.

[Fixed the compile error due to the missing "&" by tiwai]

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:21:46 +02:00
Takashi Sakamoto 10550bea44 ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE
In previous commit, AMDTP functionality in firewire-lib supports mapping
for PCM data channels. With this mapping, firewire-lib can obsolete
a flag, CIP_HI_DUALWIRE, but Dice driver still keeps dual wire mode.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:15:10 +02:00
Takashi Sakamoto 77d2a8a4f6 ALSA: firewire-lib: Add support for channel mapping
Some devices arrange the position of PCM/MIDI data channel in AMDTP packet.
This commit allows drivers to set channel mapping.

To be simple, the mapping table is an array with fixed length. Then the number
of channels for PCM is restricted by 64 channels.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:14:41 +02:00
Takashi Sakamoto 7b3b0d8583 ALSA: firewire-lib: Add support for duplex streams synchronization in blocking mode
Generally, the devices can synchronize to handle 'presentation timestamp'
in CIP packets. This commit adds functionality to pick up this timestamp from
in-packets transmitted by the device, then use it for out packets.

In current implementation, this module generated the timestamp by itself. This
is 'SYT Match' mode. Then drivers with this module acts as synchronization
master. This commit allows this module to act as synchronization slave.

This commit restricts this mechanism is only available in blocking mode because
handling the timestamp in non-blocking mode is more complicated than in
blocking mode.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:13:59 +02:00
Takashi Sakamoto ccccad8646 ALSA: firewire-lib: Give syt value as parameter to handle_out_packet()
For duplex streams with synchronization, drivers should pick up
'presentation timestamp' from in-packets and use the timestamp for
out-packets. This commit is preparation for this.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:13:44 +02:00
Takashi Sakamoto 83d8d72dff ALSA: firewire-lib: Add support for MIDI capture/playback
For capturing/playbacking MIDI messages, this commit adds one MIDI conformant
data channel. This data channel has multiplexed 8 MIDI data streams. So this
data channel can transfer messages from/to 8 MIDI ports.

And this commit allows to set PCM format even if AMDTP streams already start.
I suppose the case that PCM substreams are going to be joined into AMDTP
streams when AMDTP streams are already started for MIDI substreams. Each
driver must count how many PCM/MIDI substreams use AMDTP streams to stop
AMDTP streams.

There are differences between specifications about MIDI conformant data.

About the multiplexing, IEC 61883-6:2002, itself, has no information. It
describes labels and bytes for MIDI messages and refers to MMA/AMEI RP-027
for 'successfull implementation'. MMA/AMEI RP-027 describes 8 MPX-MIDI data
streams for one MIDI conformant data channel. IEC 61883-6:2005 adds
'sequence multiplexing' and apply this way and describe incompatibility
between 2002 and 2005.

So this commit applies IEC 61883-6:2005. When we find some devices compliant
to IEC 61883-6:2002, then this difference should be handles as device quirk
in additional work.

About the number of bytes in an MIDI conformant data, IEC 61883-6:2002 describe
0,1,2,3 bytes. MMA/AMEI RP-027 describes 'MIDI1.0-1x-SPEED', 'MIDI1.0-2x-SPEED',
'MIDI1.0-3x-SPEED' modes and the maximum bytes for each mode corresponds to 1,
2, 3 bytes. The 'MIDI1.0-2x/3x-SPEED' modes are accompanied with 'negotiation
procedure' and 'encapsulation details' but there is no specifications for them.

So this commit implements 'MIDI1.0-1x-SPEED' mode for playback, but allows
to pick up 1-3 bytes for capturing.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:12:44 +02:00
Takashi Sakamoto 2b3fc456fe ALSA: firewire-lib: Add support for AMDTP in-stream and PCM capture
For capturing PCM, this commit adds the functionality to handle in-stream.
This is also applied for dual-wire mode.

Currently, capturing 32bit samples are supported.

When the sequence of in-packet has discontinuity of dbc, in-stream isn't handled
and amdtp_streaming_error() returns true.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:12:35 +02:00
Takashi Sakamoto 4b7da117e5 ALSA: firewire-lib: Split some codes into functions to reuse for both streams
Some codes can be reused to handle in-stream. This commit adds new functions.
This commit also renames some functions to keep naming consistency.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:11:57 +02:00
Takashi Sakamoto 3ff7e8f0d4 ALSA: firewire-lib: Add 'direction' member to 'amdtp_stream' structure
This patch adds 'direction' member to amdtp_stream structure to indicate its
direction. This patch also adds 'direction' argument to amdtp_stream_init()
function to determine its direction.

The amdtp_stream_init() function is exported and used by firewire-speakers and
dice so this patch also affects them.

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>
2014-05-26 14:11:42 +02:00
Takashi Sakamoto b445db440c ALSA: firewire-lib: Add macros instead of fixed value for AMDTP
This patch adds some macros instead of fixed value for AMDTP according to
IEC 61883-1/6. These macros will also be used by followed patches.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:11:22 +02:00
Takashi Sakamoto be4a28940a ALSA: firewire-lib: Rename functions, structure, member for AMDTP
This patch renames some functions, a structure and its member to reuse them
in both AMDTP in/out stream.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:11:10 +02:00
Hui Wang e191893830 ALSA: hda - add an instance to use snd_hda_pick_pin_fixup
Just two members in the alc269_pin_fixup_tbl[] can cover more than
10 Dell laptop models.

Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 11:06:22 +02:00
Hui Wang c687200b9d ALSA: hda - drop def association and sequence from pinconf comparing
A lot a machine have the same codec, but they have different default
pinconf setting just because the def association and sequence is
different, as a result they can't share a hda_pintbl[], to overcome
it, we don't compare def association and sequence in the pinconf
matching.

Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 11:05:55 +02:00
Hui Wang 621b5a047e ALSA: hda - get subvendor from codec rather than pci_dev
It is safer for non-pci situation.

Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 11:05:26 +02:00
David Henningsson 20531415ad ALSA: hda - Add a new quirk match based on default pin configuration
Normally, we match on pci ssid only. This works but needs new code
for every machine. To catch more machines in the same quirk, let's add
a new type of quirk, where we match on
 1) PCI Subvendor ID (i e, not device, just vendor)
 2) Codec ID
 3) Pin configuration default

If all these three match, we could be reasonably certain that the
quirk should apply to the machine even though it might not be the
exact same device.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 11:03:53 +02:00
David Henningsson c21c8cf77f ALSA: hda - Add fixup_forced flag
The "fixup_forced" flag will indicate whether a specific fixup
(or nofixup) has been set by the user, to override the driver's
default.
This flag will help future patches.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 11:03:38 +02:00
Daniel Mack a860d95f74 ALSA: snd-usb: mixer: remove error messages on failed kmalloc()
If kmalloc() fails, warnings will be loud enough. We can safely just
return -ENOMEM in such cases.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-25 09:09:01 +02:00
Daniel Mack 6bc170e4e8 ALSA: snd-usb: mixer: coding style fixups
Shorten some over-long lines, multi-line comments, spurious whitespaces,
curly brakets etc.  No functional change.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-25 09:08:46 +02:00
Takashi Iwai 77f07800cb ALSA: hda - Fix onboard audio on Intel H97/Z97 chipsets
The recent Intel H97/Z97 chipsets need the similar setups like other
Intel chipsets for snooping, etc.  Especially without snooping, the
audio playback stutters or gets corrupted.  This fix patch just adds
the corresponding PCI ID entry with the proper flags.

Reported-and-tested-by: Arthur Borsboom <arthurborsboom@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-23 09:09:26 +02:00
Sylwester Nawrocki a6aba536ab ASoC: samsung: Handle errors when getting the op_clk clock
Ensure i2s->op_clk is not used when clk_get() for this clock fails.
This prevents working with an incorrectly configured clock in some
conditions.

Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-22 17:57:27 +01:00
Takashi Iwai 0c1d121016 ASoC: Updates for v3.16
Lots of cleanup work going on in the core this release but very little
 visible to external users except for the new drivers that have been
 added.
 
  - 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.
  - New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
    ST STA350.
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Merge tag 'asoc-v3.16' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: Updates for v3.16

Lots of cleanup work going on in the core this release but very little
visible to external users except for the new drivers that have been
added.

 - 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.
 - New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
   ST STA350.
2014-05-22 17:50:00 +02:00
Benoit Taine 6f51f6cf68 ALSA: Replace DEFINE_PCI_DEVICE_TABLE macro use
We should prefer `const struct pci_device_id` over
`DEFINE_PCI_DEVICE_TABLE` to meet kernel coding style guidelines.
This issue was reported by checkpatch.

A simplified version of the semantic patch that makes this change is as
follows (http://coccinelle.lip6.fr/):

// <smpl>

@@
identifier i;
declarer name DEFINE_PCI_DEVICE_TABLE;
initializer z;
@@

- DEFINE_PCI_DEVICE_TABLE(i)
+ const struct pci_device_id i[]
= z;

// </smpl>

It has been tested by compilation.

Signed-off-by: Benoit Taine <benoit.taine@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-22 17:46:56 +02:00
Mark Brown cee429e5c5 Merge remote-tracking branches 'asoc/topic/ux500', 'asoc/topic/wm8731', 'asoc/topic/wm8804', 'asoc/topic/wm8955' and 'asoc/topic/wm8985' into asoc-next 2014-05-22 00:24:04 +01:00
Mark Brown 04f87446c2 Merge remote-tracking branches 'asoc/topic/rt5651', 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/sirf', 'asoc/topic/sta350' and 'asoc/topic/tlv320dac33' into asoc-next 2014-05-22 00:24:00 +01:00
Mark Brown 6f821c6449 Merge remote-tracking branches 'asoc/topic/nuc900', 'asoc/topic/omap', 'asoc/topic/pxa', 'asoc/topic/rcar', 'asoc/topic/rt5640' and 'asoc/topic/rt5645' into asoc-next 2014-05-22 00:23:57 +01:00
Mark Brown 6630f30ed5 Merge remote-tracking branches 'asoc/topic/headers', 'asoc/topic/intel', 'asoc/topic/jz4740', 'asoc/topic/max98090', 'asoc/topic/max98095', 'asoc/topic/mc13783' and 'asoc/topic/multicodec' into asoc-next 2014-05-22 00:23:54 +01:00
Mark Brown 3a6a489fd8 Merge remote-tracking branches 'asoc/topic/devm', 'asoc/topic/fsl', 'asoc/topic/fsl-esai', 'asoc/topic/fsl-sai', 'asoc/topic/fsl-spdif' and 'asoc/topic/fsl-ssi' into asoc-next 2014-05-22 00:23:51 +01:00
Mark Brown 0c5dacf2ca Merge remote-tracking branches 'asoc/topic/cs42l56', 'asoc/topic/cs42xx8' and 'asoc/topic/davinci' into asoc-next 2014-05-22 00:23:49 +01:00
Mark Brown b03a1c7029 Merge remote-tracking branches 'asoc/topic/ad1980', 'asoc/topic/adsp', 'asoc/topic/ak4104', 'asoc/topic/ak4642', 'asoc/topic/alc5623', 'asoc/topic/arizona', 'asoc/topic/atmel' and 'asoc/topic/cache' into asoc-next 2014-05-22 00:23:45 +01:00
Mark Brown 497c11a946 Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next 2014-05-22 00:23:45 +01:00
Mark Brown b79e16cb4a Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next 2014-05-22 00:23:44 +01:00
Mark Brown e3ac3f2510 Merge remote-tracking branch 'asoc/topic/enum' into asoc-next 2014-05-22 00:23:44 +01:00
Mark Brown 566d4eeff8 Merge remote-tracking branch 'asoc/topic/dt' into asoc-next 2014-05-22 00:23:43 +01:00
Mark Brown 8e8fbd8f58 Merge remote-tracking branch 'asoc/topic/dapm-init' into asoc-next 2014-05-22 00:23:42 +01:00
Mark Brown 6bf88ab2ec Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next 2014-05-22 00:23:42 +01:00
Mark Brown 1450da3cf6 Merge remote-tracking branch 'asoc/topic/core' into asoc-next 2014-05-22 00:23:41 +01:00
Mark Brown 0f4019e6f4 Merge remote-tracking branch 'asoc/topic/component' into asoc-next 2014-05-22 00:23:41 +01:00
Mark Brown 228704bbdd Merge remote-tracking branch 'asoc/fix/max98090' into asoc-linus 2014-05-22 00:23:37 +01:00
Mark Brown 95b9cff321 ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
 Haswell/Baytrail drivers from Intel.  It's a bit larger than is good for
 this point in the cycle but it's all for a newly added driver so not so
 worrying as it might otherwise be.  Some of it's integration problems,
 some of it's the sort of problem usually turned up in stress tests.
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Merge tag 'asoc-v3.15-rc5-intel' into asoc-linus

ASoC: Intel fixes for v3.15

This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel.  It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be.  Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.

# gpg: Signature made Wed 14 May 2014 12:40:27 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"
2014-05-22 00:23:36 +01:00
Mark Brown dd97254f5c ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
 relevance outside of the driver.
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Merge tag 'asoc-v3.15-rc5-drivers' into asoc-linus

ASoC: Driver fixes for v3.15

A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.

# gpg: Signature made Wed 14 May 2014 12:49:57 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"
2014-05-22 00:23:31 +01:00
Mark Brown 266bd275b9 ASoC: Core fixes for v3.15
A few things here:
 
  - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
    have audio paths which shouldn't be present causing spurious powerups
    and potential audible issues for users.
  - Ensure the suspend->off transition doesn't have spurious transitions
    to prepare added to the sequence.
  - Fix incorrect skipping of PCM suspension for active audio streams.
  - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
    this and Timur no longer has the boards that he was using.
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Merge tag 'asoc-v3.15-rc5-core' into asoc-linus

ASoC: Core fixes for v3.15

A few things here:

 - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
   have audio paths which shouldn't be present causing spurious powerups
   and potential audible issues for users.
 - Ensure the suspend->off transition doesn't have spurious transitions
   to prepare added to the sequence.
 - Fix incorrect skipping of PCM suspension for active audio streams.
 - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
   this and Timur no longer has the boards that he was using.

# gpg: Signature made Wed 14 May 2014 12:59:19 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"
2014-05-22 00:23:30 +01:00
Tushar Behera 1d55417e12 ASoC: samsung: Add devm_clk_get to pcm.c
clk_get in probe function can be safely replaced with devm_clk_get.

Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-22 00:20:11 +01:00
Tushar Behera 7253e354e7 ASoC: samsung: Use devm_snd_soc_register_component
Replaced snd_soc_register_component with its devres equivalent,
devm_snd_soc_register_component.

Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-22 00:20:11 +01:00
Tushar Behera 55313bd3b0 ASoC: samsung: Use devm_snd_soc_register_platform
Replaced snd_soc_register_platform with devm_snd_soc_register_platform
in samsung_asoc_dma_platform_register(). This makes the function
samsung_asoc_dma_platform_unregister() redundant. This is removed and
all its users are updated.

Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-22 00:20:11 +01:00
Tushar Behera c583883ecd ASoC: samsung: Use devm_snd_soc_register_card
Replace snd_soc_register_card with devm_snd_soc_register_card.
With this change, we can delete the empty remove functions.

Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-22 00:20:11 +01:00
Kailang Yang 13fd08a339 ALSA: hda/realtek - Add support headset mode for ALC233
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-21 11:13:17 +02:00
Toralf Förster 2d3a277822 ALSA: lola: fix format type mismatch in sound/pci/lola/lola_proc.c
Signed-off-by: Toralf Förster <toralf.foerster@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-21 11:12:15 +02:00
Toralf Förster e7fc496066 ALSA: hda - fix format type mismatch in sound/pci/hda/patch_sigmatel.c
Signed-off-by: Toralf Förster <toralf.foerster@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-21 11:11:50 +02:00
Takashi Iwai e9bd7d5ce8 ALSA: hda - Disable AA-mix on Sony Vaio S13
The analog-loopback causes the speaker noises even if it's set to zero
volume.  As a simple workaround, just get rid of the loopback mixer.

Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=873704
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-21 11:06:49 +02:00
Gabriele Mazzotta 5e6db6699b ALSA: hda - White noise fix for XPS13 9333
Disable the AA-loopback path to get rid of the constant white noise
that can be heard when headphones are used.

Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-21 11:00:06 +02:00