Commit Graph

26841 Commits

Author SHA1 Message Date
John Hsu
babd658503 ASoC: nau8540: fix tab conversion problem
Fix the tab converting to space problem.

Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-18 18:05:19 +01:00
Mauro Carvalho Chehab
a5f8661df0 ALSA: usb-audio: replace /proc/bus/usb by /dev/bus/usb
The /proc/bus/usb devices don't exist anymore, since when we
got rid of usbfs. Those devices are now seen at
/dev/bus/usb.

Signed-off-by: Mauro Carvalho Chehab <mchehab@s-opensource.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-17 10:23:55 +02:00
Douglas Anderson
7952b4baff ASoC: rt5514: Unconfuse the rt5514 at probe / resume time
The rt5514 can get confused and incorrectly detect a start bit if the
SCL/SDA lines happen to both go low and then high again.  This
situation has been seen to happen at reboot time and is also
theoretically possible during suspend/resume if the rt5514 keeps power
but we shut down the i2c connection.

When this happens the rt5514 is confused about the state of the i2c
bus and won't recognize its own address.  That will lead to the rt5514
incorrectly NAKing the first transfer.

A single i2c transfer to any address should be enough to get the
rt5514 out of this funky state.

It is currently believed that this problem should be fixed in the
rt5514 driver itself because it seems that the i2c controller in the
rt5514 is easily confused.  Most i2c devices wouldn't detect a start
bit in this case.

Signed-off-by: Douglas Anderson <dianders@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-14 18:11:37 +01:00
Douglas Anderson
0a78b248c3 ASoC: rt5514: Avoid relying on uninitialized "val" value
In rt5514_i2c_probe() if the regmap_read(RT5514_VENDOR_ID2) fails then
"val" may be left as uninitialized.  Current code relies on "val" not
being RT5514_DEVICE_ID, but that's potentially unsafe.

Let's check for errors from regmap_read() and also explicitly init the
value do we're not passing a possibly uninitialized int to printk.

Signed-off-by: Douglas Anderson <dianders@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-14 18:11:37 +01:00
Douglas Anderson
d0c02e14e4 ASoC: rt5514: Mark rt5514_i2c_driver as static
There's no reason for rt5514_i2c_driver to be non-static.

Signed-off-by: Douglas Anderson <dianders@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-14 18:11:37 +01:00
Takashi Sakamoto
ae369e559f ALSA: firewire-tascam: support drain callback for MIDI playback substream
ALSA driver for TASCAM FireWire series transfers MIDI messages in system
workqueue. In current design of the driver, applications should wait for
sequence of transmission when they close ALSA rawmidi character devices.
However, when considering design of rawmidi interface, it's preferable
to wait in drain ioctl.

This commit adds support for the drain ioctl to wait for the end of
the transmission.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-14 14:50:43 +02:00
Takashi Sakamoto
1f94205d22 ALSA: firewire-tascam: move message parameters for async midi port
Units on TASCAM FireWire series handle MIDI messages with support for
running status. Drivers for the series should remember current running
status and transfer valid MIDI messages. For this purpose, current
ALSA driver for the series has some members in its top-level structure.
This is due to better abstraction of async midi port. Nowadays, the
abstraction was localized just for the driver.

This commit moves the members to structure for async midi port.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-14 14:50:40 +02:00
Takashi Sakamoto
3e7dc65ca0 ALSA: firewire-tascam: initialize parameters at open of rawmidi character devices
In current design of ALSA driver for TASCAM FireWire series, initialization
of members in asymc midi port structure is done at device probing. Some of
the members should be initialized every time to use rawmidi devices because
they're changed in sequence of transmission for MIDI messages.

This commit adds a new function to initialize them. Invariant parameters
during object lifetime are kept as is.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-14 14:50:38 +02:00
Takashi Sakamoto
98a00d3602 ALSA: firewire-tascam: use fixed-length array for message cache to async midi port
ALSA driver for TASCAM FireWire series internally allocates 4 byte buffer
for asynchronous transaction to transfer MIDI messages. However, the buffer
can be allocated with memory object of parent structure.

This commit adds 4 byte array as a member of the structure and obsoletes
the redundant allocation. This is deallocated with memory object of parent
structure.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-14 14:50:36 +02:00
Takashi Sakamoto
27badc4c10 ALSA: firewire-tascam: use the same address for asynchronous transaction for MIDI message
Units on TASCAM FireWire series receive MIDI messages by asynchronous
transactions on IEEE 1394 bus. Although the transaction is sent to a
certain register, current ALSA driver for this series has a redundant design.

This commit use the same address for the transaction.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-14 14:50:33 +02:00
Takashi Sakamoto
38d5826142 ALSA: firewire-tascam: send fixed-length transaction for async midi port
TASCAM FireWire series uses asynchronous transactions with fixed length
payload for MIDI messaging. On the other hand, ALSA driver for the series
has a redundant design to handle different length of payload.

This commit removes the redundant abstraction.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-14 14:50:31 +02:00
Takashi Sakamoto
9bae2150d0 ALSA: firewire-tascam: remove callback function from async midi port
As a result of localization of async midi port, ALSA driver for TASCAM
FireWire series can call helper function directly instead of callback
registration.

This commit removes the redundant design.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-14 14:50:29 +02:00
Takashi Sakamoto
531f471834 ALSA: firewire-lib/firewire-tascam: localize async midi port
In Linux kernel 4.4, firewire-lib got a feature called as 'async midi port'
for transmission of MIDI message via IEEE 1394 asynchronous communication,
however actual consumer of this feature is ALSA driver for TASCAM FireWire
series only. When adding this feature, I assumed that ALSA driver for
Digi00x might also be a consumer, actually it's not.

This commit moves the feature from firewire-lib to firewire-tascam module.
Two minor kernel APIs are removed.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-14 14:50:26 +02:00
Takashi Iwai
1900d947b5 Merge branch 'for-linus' into for-next
Back-merge to prepare for applying more FireWire updates.
2017-04-14 09:01:04 +02:00
Takashi Sakamoto
1e0f8f68f7 ALSA: usb-line6: constify snd_kcontrol_new strucutre array
In kernel APIs of ALSA control interface, drivers can create a control
element set by a call of snd_ctl_new1() with a template. This template
is known to have const qualifier in general cases.

This commit adds the qualifier to template array, for safer program and
runtime. Application of this change moves the symbol from .data section
to .rodata section.

Cc: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-14 08:57:56 +02:00
Takashi Sakamoto
dfb00a5693 ALSA: firewire-lib: fix inappropriate assignment between signed/unsigned type
An abstraction of asynchronous transaction for transmission of MIDI
messages was introduced in Linux v4.4. Each driver can utilize this
abstraction to transfer MIDI messages via fixed-length payload of
transaction to a certain unit address. Filling payload of the transaction
is done by callback. In this callback, each driver can return negative
error code, however current implementation assigns the return value to
unsigned variable.

This commit changes type of the variable to fix the bug.

Reported-by: Julia Lawall <Julia.Lawall@lip6.fr>
Cc: <stable@vger.kernel.org> # 4.4+
Fixes: 585d7cba5e ("ALSA: firewire-lib: add helper functions for asynchronous transactions to transfer MIDI messages")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-14 08:57:25 +02:00
Takashi Iwai
4e7655fd4f ALSA: seq: Don't break snd_use_lock_sync() loop by timeout
The snd_use_lock_sync() (thus its implementation
snd_use_lock_sync_helper()) has the 5 seconds timeout to break out of
the sync loop.  It was introduced from the beginning, just to be
"safer", in terms of avoiding the stupid bugs.

However, as Ben Hutchings suggested, this timeout rather introduces a
potential leak or use-after-free that was apparently fixed by the
commit 2d7d54002e ("ALSA: seq: Fix race during FIFO resize"):
for example, snd_seq_fifo_event_in() -> snd_seq_event_dup() ->
copy_from_user() could block for a long time, and snd_use_lock_sync()
goes timeout and still leaves the cell at releasing the pool.

For fixing such a problem, we remove the break by the timeout while
still keeping the warning.

Suggested-by: Ben Hutchings <ben.hutchings@codethink.co.uk>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-13 14:13:25 +02:00
Ander Conselvan De Oliveira
a87a4d23e8 ALSA: hda: Move common haswell init to a helper
Geminilake vendor nid is different from other Skylake variants, but rest
of the initialization code is same.

So a variable is added in hdmi_spec to store the platform specific vendor
nid and move the initialization code to a helper function to be used by
both platform specific init.

Fixes: 126cfa2f5e ("ALSA: hda: Add Geminilake HDMI codec ID")
Signed-off-by: Ander Conselvan De Oliveira <ander.conselvan.de.oliveira@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Jaikrishna Nemallapudi <jaikrishnax.nemallapudi@intel.com>
Cc: Senthilnathan Veppur <senthilnathanx.veppur@intel.com>
Cc: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-13 10:23:22 +02:00
Jeeja KP
b8c722ddd5 ASoC: Intel: Skylake: Add support for deferred DSP module bind
Module at the end of DSP pipeline that needs to be connected to a module
in another pipeline are represented as a PGA(leaf node) and in PGA event
handler these modules are bound/unbounded. Modules other than PGA leaf
can be connected directly or via switch to a module in another pipeline.
Example: reference path.

To support the deferred DSP module bind, following changes are done:
o When the path is enabled, the destination module that needs to be
bound may not be initialized. If the module is not initialized, add
these modules in a deferred bind list.
o When the destination module is initialized, check for these modules
in deferred bind list. If found, bind them.
o When the destination module is deleted, Unbind the modules.
o When the source module is deleted, remove the entry from the deferred
bind list.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-12 16:57:52 +01:00
Fabio Estevam
f2a3ee0125 ASoC: fsl_esai: Remove unneeded definition
There is no need for defining FSL_ESAI_RATES locally as the standard
SNDRV_PCM_RATE_8000_192000 definition can be used instead.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-12 16:44:50 +01:00
Bhumika Goyal
49c41e1f23 ALSA: line6: constify snd_kcontrol_new structures
Declare snd_kcontrol_new strcutures as const as they are only passed as
an argument to the function snd_ctl_new1. This argument is of type const,
so snd_kcontrol_new structures having this property can be made const too.
Done using Coccinelle:

@r disable optional_qualifier@
identifier x;
position p;
@@
static struct snd_kcontrol_new x@p={...};

@ok@
identifier r.x;
position p;
@@
snd_ctl_new1(&x@p,...)

@bad@
position p != {r.p,ok.p};
identifier r.x;
@@
x@p

@depends on !bad disable optional_qualifier@
identifier r.x;
@@
+const
struct snd_kcontrol_new x;

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-12 15:35:16 +02:00
Bhumika Goyal
8fdaebbb83 ALSA: usb-audio: constify snd_kcontrol_new structures
Declare snd_kcontrol_new strcutures as const as they are only passed as
an argument to the function snd_ctl_new1. This argument is of type const,
so snd_kcontrol_new structures having this property can be made const too.
Done using Coccinelle:

@r disable optional_qualifier@
identifier x;
position p;
@@
static struct snd_kcontrol_new x@p={...};

@ok@
identifier r.x;
position p;
@@
snd_ctl_new1(&x@p,...)

@bad@
position p != {r.p,ok.p};
identifier r.x;
@@
x@p

@depends on !bad disable optional_qualifier@
identifier r.x;
@@
+const
struct snd_kcontrol_new x;

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-12 15:34:44 +02:00
Takashi Sakamoto
f91c9d7610 ALSA: firewire-lib: cache maximum length of payload to reduce function calls
During packet streaming, maximum length of payload for isochronous packet
is invariable, therefore no need to recalculate. Current ALSA IEC 61883-1/6
engine calls a function to calculate it 8,000 or more times per second
for incoming packet processing.

This commit adds a member to have maximum length of payload into 'struct
amdtp_stream', to reduces the function calls. At first callback from
isochronous context, the length is calculated and stored for later
processing.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-12 15:34:21 +02:00
Subhransu S. Prusty
12ee4022f6 ALSA: hda: Add Geminilake id to SKL_PLUS
Geminilake is Skylake family platform. So add it's id to skl_plus check.

Fixes: 126cfa2f5e ("ALSA: hda: Add Geminilake HDMI codec ID")
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Cc: Senthilnathan Veppur <senthilnathanx.veppur@intel.com>
Cc: Vinod Koul <vinod.koul@intel.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-12 07:16:48 +02:00
Takashi Iwai
b7c5ac88e8 Merge branch 'for-linus' into for-next 2017-04-12 07:16:07 +02:00
John Stultz
e6d56d21bd ASoC: hisilicon: Use devm_snd_soc_register_component
Per feedback from Mark Brown, this patch updates the hi6210-i2s
driver to use devm_snd_soc_register_component which simplifies
the logic a bit.

Signed-off-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-11 21:20:55 +01:00
John Stultz
16c1c089b2 ASoC: hisilicon: Address style nit to use break in final default of switch statement
Mark Brown suggested a style change to use break in the final
default of a switch statement, so this patch addresses that.

Signed-off-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-11 21:20:51 +01:00
John Stultz
da13d7462b ASoC: hisilicon: Add error returns even for cases that shouldn't happen.
This patch addresses feedback from Mark Brown, adding a few
extra error returns in cases that shouldn't happen

Signed-off-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-11 21:20:44 +01:00
Fabio Estevam
6f2daf82fa ASoC: tas2552: Return the real error code
In the case of error in tas2552_codec_probe() we should better
propagate the real error code instead of always returning '-EIO'.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-11 19:35:58 +01:00
Mousumi Jana
b6e38b2944 ASoC: topology: Fix to store enum text values
Add missing enum texts store in soc_enum.

Signed-off-by: Mousumi Jana <mousumix.jana@intel.com>
Signed-off-by: Pardha Saradhi K <pardha.saradhi.kesapragada@intel.com>
Signed-off-by: Kranthikumar, GudishaX <gudishax.kranthikumar@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-11 19:35:28 +01:00
Bard Liao
97c415a6f6 ASoC: rt5665: move rt5665_set_jack_detect to .set_jack
Now, we can use .set_jack callback function on codec level. So we
don't need export rt5665_set_jack_detect.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-11 17:05:20 +01:00
Mark Brown
b8d4f7a30b Merge branch 'topic/jack' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-rt5665 2017-04-11 16:58:30 +01:00
Takashi Sakamoto
c6b0b9e65f ALSA: firewire-motu: add tracepoints for messages for unique protocol
MOTU units transfer/receive messages in each data block of their
isochronous packet payload. A part of content in the message is cleard for
MIDI message transmission, while the rest is unknown yet. Additional
features are required to assist users and developers to reveal the
details.

This commit adds tracepoints for the purpose. The tracepoints are designed
for MOTU's protocol version 2 and 3 (Protocol version 1 is not upstreamed
yet). In the tracepoints, events are probed to gather first two 24 bit
data chunks of each data block. The chunks are formatted into elements
of 64 bit array with padding in MSB.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-11 08:30:47 +02:00
Takashi Sakamoto
17909c1b30 ALSA: firewire-motu: add tracepoints for SPH in IEC 61883-1 fashion
Unique protocol is used for MOTU FireWire series. In this protocol,
data block format is not compliant to AM824 in IEC 61883-1/6. Each of
the data block consists of 24 bit data chunks, except for a first
quadlet. The quadlet is used for source packet header (SPH) described
in IEC 61883-1.

The sequence of SPH seems to represent presentation timestamp
corresponding to included data. Developers have experienced that invalid
sequence brings disorder of units in the series.

Unfortunately, current implementation of ALSA IEC 61883-1/6 engine and
firewire-motu driver brings periodical noises to the units at sampling
transmission frequency based on 44.1 kHz. The engine generates the SPH with
even interval and this mechanism seems not to be suitable to the units.
Further work is required for this issue and infrastructure is preferable
to assist the work.

This commit adds tracepoints for the purpose. In the tracepoints, events
are probed to gather the SPHs from each data blocks.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-11 08:30:45 +02:00
Takashi Sakamoto
b164d2fd6e ALSA: firewire_lib: add tracepoints for packets without CIP headers
Unique protocol is used for RME Fireface series. In this protocol,
payload format for isochronous packet is not compliant to CIP in
IEC 61883-1/6. The packet includes data blocks just with data channels,
without headers and any metadata.

In previous commits, ALSA IEC 61883-1/6 engine supports this protocol.
However, tracepoints are not supported yet, unlike implementation for
IEC 61883-1/6 protocol. This commit adds support of tracepoints for
the protocol.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-11 08:30:43 +02:00
Fabio Estevam
570c70a60f ASoC: sgtl5000: Allow LRCLK pad drive strength to be changed
Introduce the "lrclk-strength" property to allow LRCLK pad drive strength
to be changed via device tree.

When running a stress playback loop test on a mx6dl wandboard channel
swap can be noticed on about 10% of the times.

While debugging this issue I noticed that when probing the SGTL5000
LRCLK pin with the scope the swap did not happen. After removing
the probe the swap started to happen again.

After changing the LRCLK pad drive strength to the maximum value the
issue is gone.

Same fix works on a mx6dl Colibri board as well.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Tested-by: Max Krummenacher <max.krummenacher@toradex.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-10 20:21:46 +01:00
Bard Liao
d7344010d1 ASoC: jack: add snd_soc_codec_set_jack
There are many codecs with the capability of jack detection. Usually,
we create a jack on machine driver but there is no common function for
machine driver to deliver the jack pointer to codec driver.
snd_soc_codec_set_jack can be used for delivering the jack pointer to
codec driver and enable the jack detection function. To make it work,
codec driver need to define a callback function to receive the jack
pointer and do all necessary procedure for enabling jack detection.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-10 19:46:09 +01:00
Andy Green
0bf750f4cb ASoC: hisilicon: Add hi6210 i2s audio driver
Add driver for hi6210 i2s controller found on hi6220 boards.

Signed-off-by: Andy Green <andy.green@linaro.org>
[jstultz: Forward ported to mainline, fairly major rework
 based on suggestions from Mark Brown]
Signed-off-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-10 19:40:13 +01:00
Fabio Estevam
c6682fedee ASoC: fsl_ssi: Use the tolower() function
Code can be simplified by using the standard tolower() funtion.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-10 18:49:17 +01:00
Fabio Estevam
580556774a ASoC: fsl_ssi: Remove FSLSSI_I2S_RATES definition
The comment for the FSLSSI_I2S_RATES definition states that the
driver currently only supports I2S slave mode, which is no longer
correct.

As FSLSSI_I2S_RATES is the same as the standard SNDRV_PCM_RATE_CONTINUOUS,
just remove its definition and its comments to make the code simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-10 18:48:49 +01:00
Takashi Iwai
7480316c26 ALSA: hda - Allow to enable/disable vmaster build explicitly
Another preliminary patch for the dual-codec support: since the
support of vmaster over multiple codecs is difficult, simply disable
it by a new flag to hda_codec struct.  A new user hint is added as
well for consistency.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-10 17:45:27 +02:00
Takashi Iwai
9f3dadb156 ALSA: hda - A new flag to enforce prefix to each pin
This is a preliminary patch for a smooth multi-codec support, and it
introduces a new flag, force_pin_prefix, to struct hda_codec.
This flag is used to force to add the pin location prefix to each
input pin.  For example, when there is only one microphone pin,
usually the auto-parser assigns the string "Mic".  With this flag on,
it'll be like "Front Mic".  Also, the creation of "Master" or "PCM"
playback volume for a single pin is suppressed, too.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195305
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-10 17:44:39 +02:00
Takashi Iwai
ff21b250e9 Merge branch 'for-linus' into for-next 2017-04-10 17:12:01 +02:00
Libin Yang
1f9d3d9869 ALSA: hda - set intel audio clock to a proper value
On some Intel platforms, the audio clock may not be set correctly
with initial setting. This will cause the audio playback/capture
rates wrong.

This patch checks the audio clock setting and will set it to a
proper value if it is set incorrectly.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=188411

Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-07 10:39:21 +02:00
Libin Yang
dde5bff541 ALSA: hda - add more ML register definitions
This patch refines the definition of AZX_MLCTL_SPA and AZX_MLCTL_CPA
and add more definitions of ML registers

Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-07 10:39:18 +02:00
Arnaud Pouliquen
d05d862ead ASoC: STI: Fix null ptr deference in IRQ handler
With RTlinux a race condition has been found that leads to NULL ptr crash:
- On CPU 0: uni_player_irq_handler is called to treat XRUN
 "(player->state == UNIPERIF_STATE_STOPPED)" is FALSE so status is checked,
 dev_err(player->dev, "FIFO underflow error detected") is printed
and then snd_pcm_stream_lock should be called to lock stream for stopping.
- On CPU 1: application stop and close the stream.
Issue is that the stop and shutdown functions are executed while
"FIFO underflow error detected" is printed.
So when CPU 0 calls snd_pcm_stream_lock, player->substream is already null.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06 19:29:07 +01:00
Charles Keepax
77b329d194 ASoC: cs35l35: Correct handling of PDN_DONE with external boost
When using an external boost supply the PDN_DONE bit is not set, update
the handling in this case to use to use an appropriate fixed delay.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06 19:27:56 +01:00
Charles Keepax
2c84afb52e ASoC: cs35l35: Improve power down time
Shorten the time it takes to power down the amp by disabling the volume
ramp whilst doing the final shutdown. The driver has already muted the
amplifier at this stage so doing the volume ramp serves no purpose.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06 19:27:52 +01:00
Daniel Baluta
303e8954af ASoC: codec: wm8960: Stop when a matching PLL freq is found
When a matching PLL freq is found, searching continues even this is
not necessary. The problem was introduced with the following refactoring
commit 84fdc00d51 ("ASoC: codec: wm9860: Refactor PLL out freq search)

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06 19:26:41 +01:00
Ryan Lee
7c0c200071 ASoC: Add support for Maxim Integrated MAX98927 Amplifier
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06 19:25:57 +01:00
Kuninori Morimoto
32973dcf71 ASoC: rsnd: merge rsnd_kctrl_new_m/s/e into rsnd_kctrl_new()
Current rsnd driver is using rsnd_kctrl_new_m/s/e function,
but the differences are very few.
This patch merge these rsnd_kctrl_new_m/s/e into rsnd_kctrl_new

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06 11:50:18 +01:00
Mark Brown
3a37471551 Merge branch 'fix/rcar' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-rcar 2017-04-06 11:50:04 +01:00
Kuninori Morimoto
fc99d23f6d ASoC: rsnd: tidyup src->convert_rate reset timing
Current src->convert_rate will be set on .hw_param, and
be reset on .quit timing.
But, .hw_param will not be called again if user did Ctrl-Z + fg.
It should be reset on initial of .hw_param to keep its value.
Here, ctu.c already do this.
This patch solves this issue, other wise, MIXed sound will be
strange if user did like below.

	> aplay -D plughw:0,0 sound_44100.wav &
	> aplay -D plughw:0,1 sound_96000.wav
	> Ctrl-Z
	> fg # 96kHz will be played as 44.1kHz

Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06 11:47:15 +01:00
Takashi Sakamoto
7e1621de14 ALSA: firewire-lib/bebob/oxfw: improve response evaluation for AV/C commands
In ALSA firewire stack, some AV/C commands are supported, including
vendor's extensions. Drivers includes response parser of each command,
according to its requirements, while the parser is written with loose
fashion in two points; error check and length check. This doesn't cause
any issues such as kernel corruption, but should be improved.

This commit modifies evaluations of return value on each parsers.

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:37:23 +02:00
Takashi Sakamoto
5b33504bad ALSA: firewire-motu: remove invalid bitshift for register value
In protocol version 3, drivers can read current sampling clock status from
register 0x'ffff'f000'0b14. 8 bits of LSB of this register represents type
of signal as source of clock.

Current driver code includes invalid bitshift to handle the parameter. This
commit fixes the bug.

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Fixes: 5992e30034 ("ALSA: firewire-motu: add support for MOTU 828mk3 (FireWire/Hybrid) as a model with protocol version 3")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:36:11 +02:00
Takashi Sakamoto
3d016d57fd ALSA: oxfw: fix regression to handle Stanton SCS.1m/1d
At a commit 6c29230e2a ("ALSA: oxfw: delayed registration of sound
card"), ALSA oxfw driver fails to handle SCS.1m/1d, due to -EBUSY at a call
of snd_card_register(). The cause is that the driver manages to register
two rawmidi instances with the same device number 0. This is a regression
introduced since kernel 4.7.

This commit fixes the regression, by fixing up device property after
discovering stream formats.

Fixes: 6c29230e2a ("ALSA: oxfw: delayed registration of sound card")
Cc: <stable@vger.kernel.org> # 4.7+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:35:14 +02:00
Takashi Sakamoto
fdb2b2eee6 ALSA: firewire-digi00x: remove transaction handler for unknown purpose
For digi00x series, asynchronous transaction is not used to transfer MIDI
messages to/from control surface. One of transction handlers in my previous
work loses its practical meaning.

This commit removes the handler. I note that unit of console type
transfers 0x00001000 to registered address of host space when switching
to 'standalone' mode. Then the unit generates bus reset.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:34:13 +02:00
Takashi Sakamoto
0c3f15f39c ALSA: firewire-digi00x: allow user space applications to read/write MIDI messages for all ports
At a commit c5fcee0373 ("ALSA: firewire-digi00x: add MIDI operations for
MIDI control port"), I described that MIDI messages for control surface is
transferred by a different way from the messages for physical ports.
However, this is wrong. MIDI messages to/from all of MIDI ports are
transferred by isochronous packets.

This commit removes codes to transfer MIDI messages via asynchronous
transaction, from MIDI handling layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:34:11 +02:00
Takashi Sakamoto
8820a4cf0c ALSA: firewire-digi00x: handle all MIDI messages on streaming packets
At a commit 9dc5d31cdc ("ALSA: firewire-digi00x: handle MIDI messages in
isochronous packets"), a functionality to handle MIDI messages on
isochronous packet was supported. But this includes some of my
misunderstanding. This commit is to fix them.

For digi00x series, first data channel of data blocks in rx/tx packet
includes MIDI messages. The data channel has 0x80 in 8 bit of its MSB,
however it's against IEC 61883-6. Unique data format is applied:
 - Upper 4 bits of LSB represent port number.
  - 0x0: port 1.
  - 0x2: port 2.
  - 0xe: console port.
 - Lower 4 bits of LSB represent the number of included MIDI message bytes;
   0x0/0x1/0x2.
 - Two bytes of middle of this data channel have MIDI bytes.

Especially, MIDI messages from/to console surface are also transferred by
isochronous packets, as well as physical MIDI ports.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:34:10 +02:00
Takashi Sakamoto
13e005f9f9 ALSA: firewire-digi00x: add support for console models of Digi00x series
Digi00x series includes two types of unit; rack and console. As long as
reading information on config rom of Digi 002 console, 'MODEL_ID' field
has a different value from the one on Digi 002 rack.

We've already got a test report from users with Digi 003 rack. We can
assume that console type and rack type has different value in the field.

This commit adds a device entry for console type. For following commits,
this commit also adds a member to 'struct snd_digi00x' to identify console
type.

$ cd linux-firewire-utils/src
$ python2 ./crpp < /sys/bus/firewire/devices/fw1/config_rom
               ROM header and bus information block
               -----------------------------------------------------------------
400  0404f9d0  bus_info_length 4, crc_length 4, crc 63952
404  31333934  bus_name "1394"
408  60647002  irmc 0, cmc 1, isc 1, bmc 0, cyc_clk_acc 100, max_rec 7 (256)
40c  00a07e00  company_id 00a07e     |
410  00a30000  device_id 0000a30000  | EUI-64 00a07e0000a30000

               root directory
               -----------------------------------------------------------------
414  00058a39  directory_length 5, crc 35385
418  0c0043a0  node capabilities
41c  04000001  hardware version
420  0300a07e  vendor
424  81000007  --> descriptor leaf at 440
428  d1000001  --> unit directory at 42c

               unit directory at 42c
               -----------------------------------------------------------------
42c  00046674  directory_length 4, crc 26228
430  120000a3  specifier id
434  13000001  version
438  17000001  model
43c  81000007  --> descriptor leaf at 458

               descriptor leaf at 440
               -----------------------------------------------------------------
440  00055913  leaf_length 5, crc 22803
444  000050f2  descriptor_type 00, specifier_ID 50f2
448  80000000
44c  44696769
450  64657369
454  676e0000

               descriptor leaf at 458
               -----------------------------------------------------------------
458  0004a6fd  leaf_length 4, crc 42749
45c  00000000  textual descriptor
460  00000000  minimal ASCII
464  44696769  "Digi"
468  20303032  " 002"

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:34:08 +02:00
Takashi Sakamoto
76fdb3a9e1 ALSA: fireface: add support for Fireface 400
Fireface 400 is a second model of RME Fireface series, released in 2006.
This commit adds support for this model.

This model supports 8 analog channels, 2 S/PDIF channels and 8 ADAT
channels in both of tx/rx packet. The number of ADAT channels differs
depending on each mode of sampling transmission frequency.

$ python2 linux-firewire-utils/src/crpp < /sys/bus/firewire/devices/fw1/config_rom
               ROM header and bus information block
               -----------------------------------------------------------------
400  04107768  bus_info_length 4, crc_length 16, crc 30568 (should be 61311)
404  31333934  bus_name "1394"
408  20009002  irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 0, max_rec 9 (1024)
40c  000a3501  company_id 000a35     |
410  1bd0862a  device_id 011bd0862a  | EUI-64 000a35011bd0862a

               root directory
               -----------------------------------------------------------------
414  000485ec  directory_length 4, crc 34284
418  03000a35  vendor
41c  0c0083c0  node capabilities per IEEE 1394
420  8d000006  --> eui-64 leaf at 438
424  d1000001  --> unit directory at 428

               unit directory at 428
               -----------------------------------------------------------------
428  000314c4  directory_length 3, crc 5316
42c  12000a35  specifier id
430  13000002  version
434  17101800  model

               eui-64 leaf at 438
               -----------------------------------------------------------------
438  000261a8  leaf_length 2, crc 25000
43c  000a3501  company_id 000a35     |
440  1bd0862a  device_id 011bd0862a  | EUI-64 000a35011bd0862a

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:54 +02:00
Takashi Sakamoto
f656edd5fb ALSA: fireface: add hwdep interface
This commit adds hwdep interface so as the other drivers for audio and
music units on IEEE 1394 have.

This interface is designed for mixer/control applications. By using this
interface, an 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>
2017-04-05 21:31:49 +02:00
Takashi Sakamoto
4b316436ab ALSA: fireface: add support for PCM functionality
This commit adds PCM functionality to transmit/receive PCM frames on
isochronous packet streaming. This commit enables userspace applications
to start/stop packet streaming via ALSA PCM interface.

Sampling rate requested by applications is used as sampling transmission
frequency of IEC 61883-1/6packet streaming. As I described in followed
commits, units in this series manages sampling clock frequency
independently of sampling transmission frequency, and they supports
resampling between their packet streaming/data block processing layer and
sampling data processing layer. This commit take this driver to utilize
these features for usability.

When internal clock is selected as source signal of sampling clock, this
driver allows user space applications to start PCM substreams at any rate
which packet streaming engine supports as sampling transmission frequency.
In this case, this driver expects units to perform resampling PCM frames
for rx/tx packets when sampling clock frequency and sampling transmission
frequency are mismatched. This is for daily use cases.

When any external clock is selected as the source signal, this driver
gets configured sampling rate from units, then restricts available
sampling rate to the rate for PCM applications. This is for studio use
cases.

Models in this series supports 64.0/128.0 kHz of sampling rate, however
these frequencies are not supported by IEC 61883-6 as sampling transmission
frequency. Therefore, packet streaming engine of ALSA firewire stack can't
handle them. When units are configured to use any external clock as source
signal of sampling clock and one of these unsupported rate is configured
as rate of the sampling clock, this driver returns EIO to user space
applications.

Anyway, this driver doesn't voluntarily configure parameters of sampling
clock. It's better for users to work with appropriate user space
implementations to configure the parameters in advance of usage.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:46 +02:00
Takashi Sakamoto
75d6d89897 ALSA: fireface: add stream management functionality
This commit adds management functionality for packet streaming.

As long as investigating Fireface 400, there're three modes depending
on sampling transmission frequency. The number of data channels in each
data block is different depending on the mode. The set of available
data channels for each mode might be different for each protocol and
model.

The length of registers for the number of isochronous channel is just
three bits, therefore 0-7ch are available.

When bus reset occurs on IEEE 1394 bus, the device discontinues to
transmit packets. This commit aborts PCM substreams at bus reset handler.

As I described in followed commits, The device manages its sampling clock
independently of sampling transmission frequency against IEC 61883-6.
Thus, it's a lower cost to change the sampling transmission frequency,
while data fetch between streaming layer and DSP require larger buffer
for resampling. As a result, device latency might tend to be larger than
ASICs for IEC 61883-1/6 such as DM1000/DM1100/DM1500 (BeBoB),
DiceII/TCD2210/TCD2220/TCD3070 and OXFW970/971.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:44 +02:00
Takashi Sakamoto
6fb7db902b ALSA: fireface: add unique data processing layer
As long as investigating Fireface 400, format of payload of each
isochronous packet is not IEC 61883-1/6, thus its format of data block
is not AM824. The remarkable points of the format are:
 * The payload just consists of some data channels of quadlet size without
   CIP header.
 * Each data channels includes data aligned to little endian order.
 * One data channel consists of two parts; 8 bit ancillary field and 24 bit
   PCM frame.

Due to lack of CIP headers, rx/tx packets include no CIP headers and
different way to check packet discontinuity. For tx packet, the ancillary
field is used for counter. However, the way of counting is different
depending on positions of data channels. At 44.1 kHz, ancillary field in:
 * 1st/6th/9th/10th/14th/17th data channels: not used for this purpose.
 * 2nd/18th data channels: incremented every data block (0x00-0xff).
 * 3rd/4th/5th/11th/12th/13th data channels: incremented every 256 data
   blocks (0x00-0x07).
 * 7th/8th/15th/16th data channels: incremented per the number of data
   blocks in a packet. The increment can occur per packet (0x00-0xff).

For tx packet, tag of each isochronous packet is used for this purpose.
The value of tag cyclically changes between 0, 1, 2 and 3 in this order.
The interval is different depending on sampling transmission frequency.
At 44.1/48.0 kHz, it's 256 data blocks. At 88.2 kHz, it's 96 data blocks.

The number of data blocks in tx packet is exactly the same as
SYT_INTERVAL. There's no empty packet or no-data packet, thus the
throughput is not 8,000 packets per sec. On the other hand, the one in
rx packet is 8,000 packets per sec, thus the number of data blocks is
different between each packet, depending on sampling transmission
frequency:
 * 44.1 kHz: 5 or 6
 * 48.0 kHz: 5 or 6 or 7
 * 88.2 kHz: 10 or 11 or 12

This commit adds data processing layer to satisfy the above specification
in a policy of 'best effort'. Although PCM frames are handled for
intermediate buffer to user space, the ancillary data is not handled at all
to reduce CPU usage, thus counter is not checked. 0 is always used for tag
of isochronous packet. Furthermore, the packet streaming layer is
responsible for calculation of the number of data blocks for each packet,
thus it's not exactly the same sequence from the above observation.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:42 +02:00
Takashi Sakamoto
3b196c394d ALSA: firewire-lib: add no-header packet processing
As long as investigating Fireface 400, IEC 61883-1/6 is not applied to
its packet streaming protocol. Remarks of the specific protocol are:
 * Each packet doesn't include CIP headers.
 * 64,0 and 128,0 kHz are supported.
 * The device doesn't necessarily transmit 8,000 packets per second.
 * 0, 1, 2, 3 are used as tag for rx isochronous packets, however 0 is
   used for tx isochronous packets.

On the other hand, there's a common feature. The number of data blocks
transferred in a second is the same as sampling transmission frequency.
Current ALSA IEC 61883-1/6 engine already has a method to calculate it and
this driver can utilize it for rx packets, as well as tx packets.

This commit adds support for the transferring protocol. CIP_NO_HEADERS
flag is newly added. When this flag is set:
 * Both of 0 (without CIP header) and 1 (with CIP header) are used as tag
   to handle incoming isochronous packet.
 * 0 (without CIP header) is used as tag to transfer outgoing isochronous
   packet.
 * Skip CIP header evaluation.
 * Use unique way to calculate the quadlets of isochronous packet payload.

In ALSA PCM interface, 128.0 kHz is not supported, and the ALSA
IEC 61883-1/6 engine doesn't support 64.0 kHz. These modes are dropped.

The sequence of rx packet has a remarkable quirk about tag. This will be
described in later commits.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:40 +02:00
Takashi Sakamoto
ff0fb5aaa8 ALSA: firewire-lib: use the same prototype for functions to handle packet
Audio and music units of RME Fireface series use its own protocol for
isochronous packets to transfer data. This protocol requires ALSA IEC
61883-1/6 engine to have alternative functions.

This commit is a preparation for the protocol.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:38 +02:00
Takashi Sakamoto
d3fc7aac11 ALSA: fireface: add proc node to help debugging
Drivers can retrieve the state and configuration of clock by read
transactions.

This commit allows protocol abstraction layer to to dump the
information for debugging, via proc interface.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:36 +02:00
Takashi Sakamoto
ff2c293efa ALSA: fireface: add support for MIDI functionality
In previous commit, fireface driver supports unique transaction mechanism
for MIDI feature. This commit adds MIDI functionality for userspace
applications.

As I wrote in a followed commit, user space applications get some
requirement from this driver. It should not touch a register to which
units transmit MIDI messages. It should configure a register in which
MIDI transmission is controlled.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:34 +02:00
Takashi Sakamoto
1917429578 ALSA: fireface: add transaction support
As long as investigating Fireface 400, MIDI messages are transferred by
asynchronous communication over IEEE 1394 bus.

Fireface 400 receives MIDI messages by write transactions to two addresses;
0x'0000'0801'8000 and 0x'0000'0801'9000. Each of two seems to correspond to
MIDI port 1 and 2.

Fireface 400 transfers MIDI messages by write transactions to certain
addresses which configured by drivers. The drivers can decide upper 4 byte
of the addresses by write transactions to 0x'0000'0801'03f4. For the rest
part of the address, drivers can select from below options:
 * 0x'0000'0000
 * 0x'0000'0080
 * 0x'0000'0100
 * 0x'0000'0180

Selected options are represented in register 0x'0000'0801'051c as bit
flags. Due to this mechanism, drivers are restricted to use addresses on
'Memory space' of IEEE 1222, even if transactions to the address have
some side effects.

This commit adds transaction support for MIDI messaging, based on my
assumption that the similar mechanism is used on the other protocols. To
receive asynchronous transactions, the driver allocates a range of address
in 'Memory space'. I apply a strategy to use 0x'0000'0000 as lower 4 byte
of the address. When getting failure from Linux FireWire subsystem, this
driver retries to allocate addresses.

Unfortunately, read transaction to address 0x'0000'0801'051c returns zero
always, however write transactions have effects to the other features such
as status of sampling clock. For this reason, this commit delegates a task
to configure this register to user space applications. The applications
should set 3rd bit in LSB in little endian order.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:31 +02:00
Takashi Sakamoto
53eb086750 ALSA: fireface: add an abstraction layer for model-specific protocols
As of 2016, RME discontinued its Fireface series, thus it's OK for us
to focus on released firmwares to drive known units.

As long as investigating Fireface 400 with Windows driver and comparing
the result to FFADO implementation, I can see these firmwares have
different register assignments. On the other hand, according to manuals
of each models, features relevant to packet streaming seem to be common,
because GUIs for these models have the same options. It's reasonable to
assume an abstraction layer of protocols to communicate to each models.

This commit adds the abstraction layer for the protocols. This layer
includes some functions to operate common features of models in this
series.

In IEC 61883-1/6, the sequence of packet can transfer timing information
to synchronize receivers to transmitters. Units of each node on IEEE 1394
bus can generate transmitter's timing clock by handling value of SYT field
in CIP header with high-precision clock. For audio and music units on
IEEE 1394 bus, this recovered clock is designed to used for sampling clock
to capture/generate PCM frames on DSP/ADC/DAC. (Actually, in this world,
there's no units to implement this specification as is, as long as I
know).

Fireface series doesn't use this mechanism. Besides, It doesn't use
isochronous packet with CIP header. It uses internal crystal unit as its
initial sampling clock. When detecting input signals which can be
available for sampling clock (e.g. ADAT input), drivers can configure
units to use the signals as source of sampling clock. When something goes
wrong, e.g. frequency mismatching between the signal and configured value,
units fallback to the other detected signals alternatively. When detecting
no alternatives, internal crystal unit is used as source of sampling
clock. On manual of Fireface 400, this mechanism is described as
'Autosync'.

On the units, packet streaming is controlled by write transactions to
certain registers. Format of the packet, e.g. the number of data channels
in a data block, is also configured by the same manner. For this purpose,
.begin_session and .finish_session is added.

The remarkable point of this protocol is to allow drivers to configure
arbitrary sampling transmission frequency; e.g. 12.345 Hz. As long as I
know, there's no actual DAC/ADC chips which support this kind of
capability. I think a pair of packet streaming layer and data block
processing layer is isolated from sampling data processing layer in a
point of governed clock. In short, between these parts, resampling layer
exists. Actually, for Fireface 400, write transactions to
0x'0000'8010'051c has an effect to change sampling clock frequency with
base frequencies (32.0/44.1/48.0 kHz) and its multipliers (x2/x4),
regardless of sampling transmission frequency.

For this reason, the abstraction layer doesn't handle parameters for
sampling clock. Instead, each implementation of .begin_session is
expected to configure sampling transmission frequency.

For packet streaming layer, it's enough to get current selection of
source signals for the sampling clock and its frequency. In the
abstraction layer, when internal crystal is selected, drivers can sets
arbitrary sampling frequency, else they should follow configured
frequency. For this purpose, .get_clock is added.

Drivers are allows to bank up data fetching from a pair of packet
streaming/data block processing layer and sampling data processing layer.
This feature seems to suppress noises at starting/stopping packet
streaming. For this purpose, .switch_fetching_mode is added.

As I described in the above, units have remarkable mechanism to manage
sampling clock and process sampling data. For debugging purpose,
.dump_sync_status and .dump_clock_config are added. I don't have a need
to common interface to represent the status and configuration,
developers can add actual implementation of the abstraction layer as they
like.

Unlike PCM frames, MIDI messages are transferred by asynchronous
communication over IEEE 1394 bus, thus target addresses are important for
this feature. The .midi_high_addr_reg, .midi_rx_port_0_reg and
.midi_rx_port_1_reg are for this purpose. I'll describe them in following
commit.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:30 +02:00
Takashi Sakamoto
ed90f91a17 ALSA: fireface: add model specific structure
RME Fireface series has several models and their specifications are
different. Currently, we find no way to retrieve the specifications
from actual devices and need to implement them in this driver.

This commit adds a structure to describe model specific data. This
structure has an identical name for each unit, and maximum number of
data channels in each mode. I'll describe about the mode in following
commits.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:28 +02:00
Takashi Sakamoto
324540c4e0 ALSA: fireface: postpone sound card registration
Just after appearing on IEEE 1394 bus, this unit generates several bus
resets. This is due to loading firmware from on-board flash memory and
initialize hardware. It's better to postpone sound card registration.

This commit schedules workqueue to process actual probe processing
2 seconds after the last bus-reset. The card instance is kept at unit
probe callback and released at card free callback. Therefore, when the
actual probe processing fails, the memory block is wasted. This is due to
simplify driver implementation.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:26 +02:00
Takashi Sakamoto
17c4e5eadc ALSA: fireface: add skeleton for RME Fireface series
This commit adds a new driver for RME Fireface series. This commit just
creates/removes card instance according to IEEE 1394 bus event. More
functions will be added in following commits.

Three types of firmware have released by RME GmbH; for Fireface 400, for
Fireface 800 and for UCX/802/UFX. It's reasonable that these models use
different protocol for communication. Currently, I've investigated
Fireface 400 and nothing others.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05 21:31:24 +02:00
Kuninori Morimoto
c12c1aad98 ASoC: soc-core: verify Sound Card normality
Current ALSA SoC Sound Card basically consists of CPU/Codec/Platform
components. If system uses Kernel modules, we can disable these drivers
by using rmmod command. In such case, we can't disable
CPU/Codec/Platform driver without disabling Sound Card driver.

But on the other hand, we can disable these drivers by using unbind
command. In such case, we can disable these drivers randomly.
In this case, we can create dirty Sound Card which is missing necessary
components.

(1) If user disabled Sound Card first, but did nothing to other drivers,
user can't use Sound because Sound Card is no longer exists.
(2) If user disabled CPU/Codec/Platform driver randomly, but did nothing
to Sound Card, user still be able to use Sound Card, because dirty Sound
Card still exists. In this case, Sound system will be crashed if user
started sound playback/capture. But we can't block such random unbind
now.

To avoid Sound Card crash in (2) case, we need to unregister Sound Card
whenever CPU/Codec/Platform component were unregistered.
This patch solves this issue.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05 18:24:07 +01:00
Daniel Baluta
84fdc00d51 ASoC: codec: wm9860: Refactor PLL out freq search
Add a separate function for deriving (sysclk, lrclk, bclk)
when the clock is auto or pll.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05 18:23:15 +01:00
Javier Martinez Canillas
7b87463edf ASoC: rt5677: Add OF device ID table
The driver doesn't have a struct of_device_id table but supported devices
are registered via Device Trees. This is working on the assumption that a
I2C device registered via OF will always match a legacy I2C device ID and
that the MODALIAS reported will always be of the form i2c:<device>.

But this could change in the future so the correct approach is to have an
OF device ID table if the devices are registered via OF.

Before this patch:

$ modinfo sound/soc/codecs/snd-soc-rt5677.ko | grep alias
alias:          i2c:RT5677CE:00
alias:          i2c:rt5676
alias:          i2c:rt5677

After this patch:

$ modinfo sound/soc/codecs/snd-soc-rt5677.ko | grep alias
alias:          of:N*T*Crealtek,rt5677C*
alias:          of:N*T*Crealtek,rt5677
alias:          i2c:RT5677CE:00
alias:          i2c:rt5676
alias:          i2c:rt5677

Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05 18:22:56 +01:00
Javier Martinez Canillas
5cf015d9cb ASoC: wm8978: Add OF device ID table
The driver doesn't have a struct of_device_id table but supported devices
are registered via Device Trees. This is working on the assumption that a
I2C device registered via OF will always match a legacy I2C device ID and
that the MODALIAS reported will always be of the form i2c:<device>.

But this could change in the future so the correct approach is to have an
OF device ID table if the devices are registered via OF.

Before this patch:

$ modinfo sound/soc/codecs/snd-soc-wm8978.ko | grep alias
alias:          i2c:wm8978

After this patch:

$ modinfo sound/soc/codecs/snd-soc-wm8978.ko | grep alias
alias:          i2c:wm8978
alias:          of:N*T*Cwlf,wm8978C*
alias:          of:N*T*Cwlf,wm8978

Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05 18:22:27 +01:00
Javier Martinez Canillas
ea22a26e67 ASoC: uda1380: Add OF device ID table
The driver doesn't have a struct of_device_id table but supported devices
are registered via Device Trees. This is working on the assumption that a
I2C device registered via OF will always match a legacy I2C device ID and
that the MODALIAS reported will always be of the form i2c:<device>.

But this could change in the future so the correct approach is to have an
OF device ID table if the devices are registered via OF.

Before this patch:

$ modinfo sound/soc/codecs/snd-soc-uda1380.ko | grep alias
alias:          i2c:uda1380

After this patch:

$ modinfo sound/soc/codecs/snd-soc-uda1380.ko | grep alias
alias:          of:N*T*Cnxp,uda1380C*
alias:          of:N*T*Cnxp,uda1380
alias:          i2c:uda1380

Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05 18:21:16 +01:00
Javier Martinez Canillas
9abe464821 ASoC: sta529: Add OF device ID table
The driver doesn't have a struct of_device_id table but supported devices
are registered via Device Trees. This is working on the assumption that a
I2C device registered via OF will always match a legacy I2C device ID and
that the MODALIAS reported will always be of the form i2c:<device>.

But this could change in the future so the correct approach is to have an
OF device ID table if the devices are registered via OF.

Before this patch:

$ modinfo sound/soc/codecs/snd-soc-sta529.ko | grep alias
alias:          i2c:sta529

After this patch:

$ modinfo sound/soc/codecs/snd-soc-sta529.ko | grep alias
alias:          of:N*T*Cst,sta529C*
alias:          of:N*T*Cst,sta529
alias:          i2c:sta529

Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05 18:21:01 +01:00
Javier Martinez Canillas
71c314d7ef ASoC: ssm4567: Add OF device ID table
The driver doesn't have a struct of_device_id table but supported devices
are registered via Device Trees. This is working on the assumption that a
I2C device registered via OF will always match a legacy I2C device ID and
that the MODALIAS reported will always be of the form i2c:<device>.

But this could change in the future so the correct approach is to have an
OF device ID table if the devices are registered via OF.

Before this patch:

$ modinfo sound/soc/codecs/snd-soc-ssm4567.ko | grep alias
alias:          acpi*:INT343B:*
alias:          i2c:ssm4567

After this patch:

$ modinfo sound/soc/codecs/snd-soc-ssm4567.ko | grep alias
alias:          acpi*:INT343B:*
alias:          of:N*T*Cadi,ssm4567C*
alias:          of:N*T*Cadi,ssm4567
alias:          i2c:ssm4567

Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05 18:20:19 +01:00
Javier Martinez Canillas
9ba2da5f5d ASoc: rt5645: Add OF device ID table
The driver doesn't have a struct of_device_id table but supported devices
are registered via Device Trees. This is working on the assumption that a
I2C device registered via OF will always match a legacy I2C device ID and
that the MODALIAS reported will always be of the form i2c:<device>.

But this could change in the future so the correct approach is to have an
OF device ID table if the devices are registered via OF.

Before this patch:

$ modinfo sound/soc/codecs/snd-soc-rt5645.ko | grep alias
alias:          acpi*:10EC3270:*
alias:          acpi*:10EC5640:*
alias:          acpi*:10EC5650:*
alias:          acpi*:10EC5648:*
alias:          acpi*:10EC5645:*
alias:          i2c:rt5650
alias:          i2c:rt5645

After this patch:

$ modinfo sound/soc/codecs/snd-soc-rt5645.ko | grep alias
alias:          of:N*T*Crealtek,rt5650C*
alias:          of:N*T*Crealtek,rt5650
alias:          of:N*T*Crealtek,rt5645C*
alias:          of:N*T*Crealtek,rt5645
alias:          acpi*:10EC3270:*
alias:          acpi*:10EC5640:*
alias:          acpi*:10EC5650:*
alias:          acpi*:10EC5648:*
alias:          acpi*:10EC5645:*
alias:          i2c:rt5650
alias:          i2c:rt5645

Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05 18:19:58 +01:00
Javier Martinez Canillas
13023ff3b3 ASoC: cs53l30: Set .of_match_table to OF device ID table
The driver has an OF device ID table but the struct i2c_driver
.of_match_table field is not set.

Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05 18:19:46 +01:00
Javier Martinez Canillas
56af0e4cd2 ASoC: max9867: export OF device ID as module aliases
The I2C core always reports a MODALIAS of the form i2c:<foo> even if the
device was registered via OF, this means that exporting the OF device ID
table device aliases in the module is not needed. But in order to change
how the core reports modaliases to user-space, it's better to export it.

While there, move the MODULE_DEVICE_TABLE(i2c, max9867_i2c_id) just next
to the I2C device table declaration, for consistency with other drivers.

Before this patch:

$ modinfo sound/soc/codecs/snd-soc-max9867.ko | grep alias
alias:          i2c:max9867

After this patch:

$ modinfo sound/soc/codecs/snd-soc-max9867.ko | grep alias
alias:          i2c:max9867
alias:          of:N*T*Cmaxim,max9867C*
alias:          of:N*T*Cmaxim,max9867

Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05 18:19:20 +01:00
Mayuresh Kulkarni
51a2c944ea ASoC: wm_adsp: add support for DSP region lock
Newer ADSP2V2 codecs include a memory protection unit that can
be set to trap illegal accesses. When enabling an ADSPV2 core we
must configure the memory region traps so that the firmware can
access its own memory.

Signed-off-by: Mayuresh Kulkarni <mkulkarni@opensource.wolfsonmicro.com>
Signed-off-by: Nikesh Oswal <Nikesh.Oswal@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05 16:14:15 +01:00
Richard Fitzgerald
e1ea1879f2 ASoC: wm_adsp: Add support for ADSP2V2
Adds support for ADSP2V2 cores. Primary differences are that
they use a 32-bit register map compared to the 16-bit register
map of ADSP2V1, and there are some changes to clocking control.

Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05 16:14:15 +01:00
Peter Ujfalusi
0636e8b380 ASoC: twl6040: Add control for HS and HF mono to stereo selection
The new controls will give user the ability to route the left PDM channel
data to the right headset/handsfree DAC.
HS mono to stereo switch: PDM channel 1 (or mono) data to both HS DAC.
HF mono to stereo switch: PDM channel 3 data to both HF DAC.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-03 18:52:01 +01:00
Takashi Iwai
f87e7f2589 ALSA: hda - Improved position reporting on SKL+
Apply the same methods to obtain the current stream position as ASoC
Intel SKL driver uses.  It reads the position from DPIB for a playback
stream while it still reads from the position buffer for a capture
stream.  For a capture stream, some ugly workaround is needed to
settle down the inconsistent position.

Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-03 08:43:17 +02:00
Takashi Iwai
70eafad849 ALSA: hda - Move SKL+ vendor specific register definitions to hda_register.h
They may be used by both legacy and ASoC drivers.

Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-03 08:43:07 +02:00
Takashi Iwai
2c1f81381e ALSA: hda - Avoid tricky macros
The macros _snd_hdac_chip_read() and *_write() expand to different
types (b,w,l) per their argument.  They were thought to be used only
internally for other snd_hdac_chip_*() macros, but in some situations
we need to call these directly, and they are way too ugly.

Instead of saving a few lines, we just write these macros explicitly
with the types, so that they can be used in a saner way.

Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-03 08:42:43 +02:00
Matthias Kaehlcke
d1600401fa ALSA: hda/ca0132: Limit values for chip addresses to 32-bit
With the previous unsigned long value clang generates warnings like
this:

sound/pci/hda/patch_ca0132.c:860:37: error: implicit conversion from
'unsigned long' to 'u32' (aka 'unsigned int') changes value from
18446744073709551615 to 4294967295 [-Werror,-Wconstant-conversion]
        spec->curr_chip_addx = (res < 0) ? ~0UL : chip_addx;
                             ~             ^~~~

Signed-off-by: Matthias Kaehlcke <mka@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-01 10:46:18 +02:00
Dan Carpenter
a8c006aafe ALSA: timer: Info leak in snd_timer_user_tinterrupt()
The "r1" struct has memory holes.  We clear it with memset on one path
where it is used but not the other.  Let's just memset it at the start
of the function so it's always safe.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-31 17:27:05 +02:00
Dan Carpenter
e8ed68205f ALSA: timer: remove some dead code
We just checked "id.card < 0" on the lines before so we know it's not
true here.  We can delete that check.

Also checkpatch.pl complains about some extra curly braces so we may as
well fix that while we're at it.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-31 17:27:02 +02:00
Dan Carpenter
5885615e44 ALSA: emux: stop if copy_from_user() fails
If we can't fill the "patch" struct because "count" is too small (it can
be as low as 4 bytes) or because copy_from_user() failed, then just
return instead of using unintialized data.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-31 16:23:52 +02:00
Takashi Iwai
03a1f48e53 ALSA: usb-audio: Fake also USB device id when alias is given
Recently snd-usb-audio driver received a new option, quirk_alias, to
allow user to apply the existing quirk for a different device.  This
works for many quirks as is, but some still need more tune-ups:
namely, some quirks check the USB vendor/device IDs in various places,
thus it doesn't work as long as the ID is different from the expected
one.

With this patch, the driver stores the aliased USB ID, so that these
rest quirks per device ID are applied.  The transition to use the
cached USB ID was already done in the past, so what we needed now is
only to overwrite chip->usb_id.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-31 11:19:19 +02:00
Hui Wang
2f726aec19 ALSA: hda - fix a problem for lineout on a Dell AIO machine
On this Dell AIO machine, the lineout jack does not work.

We found the pin 0x1a is assigned to lineout on this machine, and in
the past, we applied ALC298_FIXUP_DELL1_MIC_NO_PRESENCE to fix the
heaset-set mic problem for this machine, this fixup will redefine
the pin 0x1a to headphone-mic, as a result the lineout doesn't
work anymore.

After consulting with Dell, they told us this machine doesn't support
microphone via headset jack, so we add a new fixup which only defines
the pin 0x18 as the headset-mic.

[rearranged the fixup insertion position by tiwai in order to make the
 merge with other branches easier -- tiwai]

Fixes: 59ec4b57bc ("ALSA: hda - Fix headset mic detection problem for two dell machines")
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-31 10:58:26 +02:00
Kuninori Morimoto
b5aac5a9ad ASoC: rcar: call missing of_clk_del_provider() when remove
adg is calling of_clk_add_provider() when probe time,
thus, remove should call of_clk_del_provider(), it doesn't now.
This patch fix this issue.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-30 22:22:32 +01:00
Kuninori Morimoto
d7f298197a ASoC: rcar: fixup of_clk_add_provider() usage for multi clkout
Current adg is calling of_clk_add_povider() multiple times,
but it is not correct usage. This patch fixup its parameter
and call it once.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-30 22:22:32 +01:00
Takashi Iwai
9dfcce42b0 ASoC: Fixes for v4.11
A relatively large pile of fixes for mainline, the first since the merge
 window.  The biggest block of changes here by volume is the sun8i-codec
 set, the driver was newly added in the merge window but it was realized
 that renaming some of the user visible controls was required so these
 are being pushed for v4.11 to avoid the original code appearing in a
 release.  Otherwise it's all fairly standard bugfix stuff.
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Merge tag 'asoc-fix-v4.11-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v4.11

A relatively large pile of fixes for mainline, the first since the merge
window.  The biggest block of changes here by volume is the sun8i-codec
set, the driver was newly added in the merge window but it was realized
that renaming some of the user visible controls was required so these
are being pushed for v4.11 to avoid the original code appearing in a
release.  Otherwise it's all fairly standard bugfix stuff.
2017-03-30 20:03:25 +02:00
Colin Ian King
5f75b19ef9 ASoC: Intel: bxtn: fix spelling mistake: "Timout" -> "Timeout"
trivial fix to spelling mistake in dev_err error message

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-30 11:15:30 +01:00
Clemens Ladisch
ac310dc9fa ALSA: oxygen: simply setting of the shortname for Xonar DG cards
We don't need to manually set the card name; with an entry in the
names[] array, this happens automatically.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-29 21:14:20 +02:00
Mark Brown
2ddaa67626 Merge remote-tracking branches 'asoc/fix/rt5665', 'asoc/fix/simple', 'asoc/fix/sti' and 'asoc/fix/sun8i' into asoc-linus 2017-03-29 12:55:08 +01:00
Mark Brown
367b1301cc Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/atmel', 'asoc/fix/hdac-hdmi' and 'asoc/fix/mtk' into asoc-linus 2017-03-29 12:55:06 +01:00
Mark Brown
0cb3a12f2a Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus 2017-03-29 12:55:05 +01:00
Mark Brown
4368c27666 Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus 2017-03-29 12:55:05 +01:00
Jeeja KP
473a4d516c ASoC: Intel: Skylake: Fix module state after unbind and delete
When DSP module is unbound, the module state needs to be in INIT_DONE
state instead of UNINT. Also the state needs to be set to UNINIT after
module is deleted from DSP pipeline.

So, set the module state to INIT_DONE after unbind and then UNINIT after
module is deleted.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:53:43 +01:00
Hardik T Shah
fdd85a054b ASoC: Intel: Skylake: Fix DMA position reporting for capture stream
As per hardware recommendation, for every capture stream completion
following operations need to be done in order to reflect the actual
data that is received in position buffer.

1. Wait for 20us before reading the DMA position in buffer once the
interrupt is generated for stream completion.
2. Read any of the register to flush the DMA position value. This is
dummy read operation.

Signed-off-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:53:39 +01:00
Jeeja KP
b26199eae8 ASoC: Intel: Skylake: Rearrangement of code to cleanup SKL SST library
Skylake driver topology header/driver structure is referenced and used
in SST library which creates circular dependency. Hence the
rearrangement.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:53:34 +01:00
Vinod Koul
6ad0005f17 ASoC: Intel: Skylake: remove hard coded ACPI path
We should not hard code the ACPI path to get acpi_handle. Instead use
ACPI_HANDLE macro to do the job.

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:53:32 +01:00
Vinod Koul
9a1e350709 ASoC: Intel: Skylake: Remove redundant vmixer handler
Initially vmixer and mixer widget handlers were bit different, but over
time they became same so remove the duplicate code.

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:53:30 +01:00
Vinod Koul
f7ea77772d ASoC: Intel: Skylake: Don't unload module when in use
A module may have multiple instances in DSP, so unload only when usage
count is zero.

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:53:27 +01:00
G Kranthi
e59ed0875b ASoC: Intel: Skylake: Add 16-bit constraint to FE bxt_rt298 machine
Add constraint to FE to restrict sample format to 16-bit for bxt_rt298
machine

Signed-off-by: G Kranthi <gudishax.kranthikumar@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:53:24 +01:00
Jeeja KP
66d6bbc6c0 ASoC: hdac_hdmi: Update sig_bits based on converter capability
When creating the codec dai, use sig_bits to update the max bps based
on the codec capability. So both the link DMA and codec format will be
calculated based on DAI sig_bits.

So update the sig_bits with converter capability and use the sig_bits
for HDA format calculation.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:53:21 +01:00
Jeeja KP
7f975a385b ASoC: Intel: Skylake: Use the sig_bits to define dai bps capability
For calculating the HDA DMA format, use the max_bps supported by the
DAI caps instead of fixing it to 32/24. For host DMA the Max bps support
is 32, but in case of link DMA, this depends on the codec capability.
So use the sig_bits to define the bps supported by dai.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:53:06 +01:00
Mark Brown
240a07dbc3 Merge branch 'topic/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-intel 2017-03-29 12:50:26 +01:00
Daniel Baluta
db22d18945 ASoC: imx-wm8962: Fix codec_clk cleanup
Resource managed devm_clk_get only works with platform's device dev.

Reported-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:50:12 +01:00
Daniel Baluta
fd8ba1e309 ASoC: imx-wm8962: Let codec driver enable/disable its MCLK
WM8962 needs its MCLK when powerup in wm8962_resume(). Thus it's better
to control the MCLK in codec driver. Thus remove the clock enable in
machine driver accordingly.

While at it, get rid of imx_wm8962_remove function since it is now
empty.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:48:33 +01:00
B, Jayachandran
ccfdf9f6a4 ALSA: hda: Fix LLCH register read
LLCH is a 16 bit register. Use readw instead of readl API.

Signed-off-by: B, Jayachandran <jayachandran.b@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:01:35 +01:00
Kuninori Morimoto
e0c4211854 ASoC: rcar: remove rsnd_kctrl_remove()
Current rcar driver is trying to remove kctrl when remove time.
But, 1) rcar driver can't/shouldn't remove before removing sound
card driver, 2) sound card driver will call snd_ctl_dev_free()
and removes all kctrls by snd_ctl_remove().
Thus, rsnd_kctrl_remove() is not necessary. Current implementation
will get Oops when removing rcar driver after sound card.
This patch fix this issue.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 12:01:10 +01:00
Ryder Lee
8625c1dbd8 ASoC: mediatek: Add mt2701-wm8960 machine driver
Add wm8960 machine driver and config option for MT2701.

Signed-off-by: Ryder Lee <ryder.lee@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-29 11:37:53 +01:00
Takashi Sakamoto
5992e30034 ALSA: firewire-motu: add support for MOTU 828mk3 (FireWire/Hybrid) as a model with protocol version 3
MOTU 828mk3 (FireWire/Hybrid) is one of third generation in MOTU FireWire
series, produced in 2008/2014. This model consists of three chips for
functionality on IEEE 1394 bus:

 * TI TSB41AB2 (Physical layer for IEEE 1394 bus)
 * Xilinx Spartan-3E FPGA Family (Link layer for IEEE 1394 bus, packet
   processing and data block processing layer)
 * TI TMS320C6722 (Digital signal processing)

This commit adds a support for this model, with its unique protocol as
version 3. This protocol has some additional features to protocol
version 2.

 * Support several optical interfaces.
 * Support a data chunk for return of reverb effect.
 * Have a quirk of tx packets.
 * Support heartbeat asynchronous transaction.

In this protocol, series of transferred packets has some quirks. Below
fields in CIP headers of the packets are out of IEC 61883-1:
 - SID (source node id): always 0x0d
 - DBS (data block size): always 0x04
 - DBC (data block counter): always 0x00
 - EOH (End of header): always 0x00

Below is an actual sample of transferred packets.

quads CIP1       CIP2
520   0x0D040400 0x22FFFFFF
  8   0x0D040400 0x22FFFFFF
520   0x0D040400 0x22FFFFFF
520   0x0D040400 0x22FFFFFF
  8   0x0D040400 0x22FFFFFF

Status of clock is configured by write transactions to 0x'ffff'f000'0b14,
as well as version 2, while meanings of fields are different from the
former protocols. Modes of optical interfaces are configured by write
transactions to 0x'ffff'f000'0c94.

Drivers can register its address to receive heatbeat transactions from the
unit. 0x'ffff'f000'0b0c is for the higher part and 0x'ffff'f000'0b10 is
for the lower part. Nevertheless, this feature is not useless for this
driver and this commit omits it.

Each data block consists of two parts in a point of the number of included
data chunks. In both of 'fixed' and 'differed' parts, the number of
included data blocks are a multiple of 4, thus depending on models there's
some empty data chunks. For example, 828mk3 includes one pair of empty
data chunks in its fixed part. When optical interface is configured to
S/PDIF, 828mk3 includes one pair of empty data chunks in its differed part.
To reduce consumption of CPU cycles with additional conditions/loops, this
commit just exposes these empty chunks to user space as PCM channels.

Additionally, 828mk3 has a non-negligible overhead to change its sampling
transfer frequency. When softwares send asynchronous transaction to
perform it, LED on the unit starts to blink. In a worst case, it continues
blink during several seconds; e.g. 10 seconds. When stopping blinking,
the unit seems to be prepared for the requested sampling transfer
frequency. To wait for the preparation, this commit forces the driver
to call task scheduler and applications sleeps for 4 seconds.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:34:13 +02:00
Takashi Sakamoto
2128f78f75 ALSA: firewire-lib: add a quirk of packet without valid EOH in CIP format
In IEC 61883-1, when two quadlets CIP header is used, the most significant
bit in second CIP header stands. However, packets from units with MOTU
protocol version 3 have a quirk without this flag. Current packet streaming
layer handles this as protocol error.

This commit adds a new enumeration constant for this quirk, to handle MOTU
protocol version 3.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:34:11 +02:00
Takashi Sakamoto
949613e366 ALSA: firewire-motu: add support for MOTU 828mk2 as a model with protocol version 2
MOTU 828mk2 is one of second generation in MOTU FireWire series, produced in
2003. This model consists of four chips:
 * TI TSB41AB2 (Physical layer for IEEE 1394 bus)
 * PDI 1394L40BE (Link layer for IEEE 1394 bus and packet processing layer)
 * ALTERA ACEX 1K EP1K30 Series FPGA (Data block processing layer)
 * TI TMS320VC5402 (Digital signal processing)

This commit adds a support for this model, with its unique protocol as
version 2. The features of this protocol are:

 * Support data chunks for status and control messages for both
   directions.
 * Support a pair of MIDI input/output.
 * Support a data chunk for mic/instrument independent of analog line in.
 * Support a data chunk for playback return.
 * Support independent data chunks for S/PDIF of both optical/coaxial
   interfaces.
 * Support independent data chunks for each of main out and phone out.

Status of clock is configured by write transactions to 0x'ffff'f000'0b14.
Modes of optical interfaces are configured by write transactions to
0x'ffff'f000'0c04.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:34:08 +02:00
Takashi Sakamoto
5aaab1bf37 ALSA: firewire-motu: enable to read transaction cache via hwdep interface
MOTU FireWire series can transfer messages to registered address. These
messages are transferred for the status of internal clock synchronization
just after starting streams.

When the synchronization is stable, it's 0x01ffffff. Else, it's 0x05ffffff.

This commit adds a functionality for user space applications to receive
content of the message.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:34:06 +02:00
Takashi Sakamoto
71c3797779 ALSA: firewire-motu: add hwdep interface
This commit adds hwdep interface so as the other sound drivers for units
on IEEE 1394 bus have.

This interface is designed for mixer/control applications. By using this
interface, an 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>
2017-03-28 12:34:02 +02:00
Takashi Sakamoto
9e796e7d59 ALSA: firewire-motu: add MIDI functionality
In MOTU FireWire series, MIDI messages are multiplexed to isochronous
packets as well as PCM frames, while the way is different from the one
in IEC 61883-6.

MIDI messages are put into a certain position in message chunks. One data
block can includes one byte of the MIDI messages. When data block includes
a MIDI byte, the block has a flag in a certain position of the message
chunk. These positions are unique depending on protocols.

Once a data block includes a MIDI byte, some following data blocks includes
no MIDI bytes. Next MIDI byte appears on a data block corresponding to
next cycle of physical MIDI bus. This seems to avoid buffer overflow caused
by bandwidth differences between IEEE 1394 bus and physical MIDI bus.

This commit adds MIDI functionality to transfer/receive MIDI messages.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:56 +02:00
Takashi Sakamoto
dd49b2d1f0 ALSA: firewire-motu: add PCM functionality
This commit adds PCM functionality to transmit/receive PCM samples.

When one of PCM substreams are running or external clock source is
selected, current sampling rate is used. Else, the sampling rate is
changed according to requests from a userspace application.

Available number of samples in a frame of PCM substream is determined at
open(2) to corresponding PCM character device. Later, packet streaming
starts by ioctl(2) with SNDRV_PCM_IOCTL_PREPARE. In theory, between them,
applications can change state of the unit by any write transaction to
change the number. In this case, this driver may fail packet streaming due
to wrong data format.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:53 +02:00
Takashi Sakamoto
4638ec6ede ALSA: firewire-motu: add proc node to show current statuc of clock and packet formats
This commit adds a proc node for debugging purpose.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:51 +02:00
Takashi Sakamoto
9b2bb4f2f4 ALSA: firewire-motu: add stream management functionality
This commit adds a functionality to manage packet streaming for MOTU
FireWire series.

The streaming is not controlled by CMP, thus against IEC 61883-1. Write
transaction to certain addresses start/stop packet streaming.

Transactions to 0x'ffff'f000'0b00 results in isochronous channel number for
both directions and starting/stopping transmission of packets. The
isochronous channel number is represented in 6 bit field, thus units can
identify the channels up to 64, as IEEE 1394 bus specification described.

Transactions to 0x'ffff'f000'0b10 results in packet format for both
directions and transmission speed. When each of data block includes fixed
part of data chunks only, corresponding flags stand.

When bus reset occurs, the units continue to transmit packets with
non-contiguous data block counter. This causes discontinuity detection in
packet streaming engine and ALSA PCM applications receives EPIPE from any
I/O operation. In this case, typical applications manage to recover
corresponding PCM substream. This behaviour is kicked much earlier than
callback of bus reset handler by Linux FireWire subsystem, therefore
status of packet streaming is not changed in the handler.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:34 +02:00
Takashi Sakamoto
2e76701bbb ALSA: firewire-motu: handle transactions specific for MOTU FireWire models
All models of MOTU FireWire series can be controlled by write transaction
to addresses in a range from 0x'ffff'f0000'0b00 to 0x'ffff'f000'0cff.

The models support asynchronous notification. This notification has 32 bit
field data, and is transferred when status of clock changes. Meaning of
the value is not enough clear yet.

Drivers can register its address to receive the notification. Write
transaction to 0x'ffff'f000'0b04 registers higher 16 bits of the address.
Write transaction to 0x'ffff'f0000'0b08 registers the rest of bits. The
address includes node ID, thus it should be registered every time of bus
reset.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:32 +02:00
Takashi Sakamoto
4641c93940 ALSA: firewire-motu: add MOTU specific protocol layer
MOTU FireWire series uses blocking transmission for AMDTP packet streaming.
They transmit/receive 8,000 packets per second, to handle the same number
of data blocks as current sampling transmission frequency. Thus,
IEC 61883-1/6 packet streaming engine of ALSA firewire stack is available
for them.

However, the sequence of packet and data blocks includes some quirks.
Below sample is a sequence of CIP headers of packets received by 828mk2,
at 44.1kHz of sampling transmission frequency.

quads CIP1        CIP2
488   0x020F04E8  0x8222FFFF
  8   0x020F04F8  0x8222FFFF
488   0x020F0400  0x8222FFFF
488   0x020F0408  0x8222FFFF
  8   0x020F04E8  0x8222FFFF
488   0x020F04F0  0x8222FFFF
488   0x020F04F8  0x8222FFFF

The SID (source node ID), DBS (data block size), SPH (source packet header),
FMT (format ID), FDF (format dependent field) and SYT (time stamp) fields
are in IEC 61883-1. Especially, FMT is 0x02, FDF is 0x22 and SYT is 0xffff
to define MOTU specific protocol. In an aspect of dbc field, the value
represents accumulated number of data blocks included the packet. This
is against IEC 61883-1, because according to the specification this value
should be the number of data blocks already transferred.

In ALSA IEC 61883-1/6 engine, this quirk is already supported by
CIP_DBC_IS_END_EVENT flag, because Echo Audio Fireworks has.

Each data block includes SPH as its first quadlet field, to represent its
presentation time stamp. Actual value of SPH is compliant to IEC 61883-1;
lower 25 bits of 32 bits width consists of 13 bits cycle count and 12 bits
cycle offset.

The rest of each data block consists of 24 bit chunks. All of PCM samples,
MIDI messages, status and control messages are transferred by the chunks.
This is similar to '24-bit * 4 Audio Pack' in IEC 61883-6. The position of
each kind of data depends on generations of each model. The number of
whole chunks in a data block is a multiple of 4, to consists of
quadlet-aligned packets.

This commit adds data block processing layer specific for the MOTU
protocol. The remarkable point is the way to generate SPH header. Time
stamps for each data blocks are generated by below calculation:

 * Using pre-computed table for the number of ticks per event
  *  44,1kHz: (557 + 123/441)
  *  48.0kHz: (512 +   0/441)
  *  88.2kHz: (278 + 282/441)
  *  96.0kHz: (256 +   0/441)
  * 176.4kHz: (139 + 141/441)
  * 192.0kHz: (128 +   0/441)
 * Accumulate the ticks and set the value to SPH for every events.
 * This way makes sense only for blocking transmission because this mode
   transfers fixed number or none of events.

This calculation assumes that each data block has a PCM frame which is
sampled according to event timing clock. Current packet streaming layer
has the same assumption.

Although this sequence works fine for MOTU FireWire series at sampling
transmission frequency based on 48.0kHz, it is not enough at the frequency
based on 44.1kHz. The units generate choppy noise every few seconds.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:30 +02:00
Takashi Sakamoto
9dae017bf6 ALSA: firewire-lib: enable CIP_DBC_IS_END_EVENT for both directions of stream
Commit c8bdf49b9935("ALSA: fireworks/firewire-lib: Add a quirk for the
meaning of dbc") adds CIP_DBC_IS_END_EVENT flag just for tx packets.
However, MOTU FireWire series has this quirk for rx packets.

This commit allows both directions with the flag.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:28 +02:00
Takashi Sakamoto
9863874f02 ALSA: firewire-lib: add support for source packet header field in CIP header
In IEC 61883-1, CIP headers can have a SPH field. When a packet has 1 in
SPH field of its CIP header, the packet has a source packet headers. A
source packet header consists of 32 bit field (= 1 quadlet) and it
transfers time stamp, which is the same value as the lower 25 bits of the
IEEE 1394 CYCLE_TIMER register and the rest is zero.

This commit just supports source packet header field because IEC 61883-1
includes ambiguity the position of this header and its count. Each
protocol layer is allowed to have actual implementation according its
requirements.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:26 +02:00
Takashi Sakamoto
a04513f8b1 ALSA: firewire-lib: record cycle count for the first packet
Currently, packet streaming layer passes generated SYT value to data block
processing layer. However, this is not enough in a case that the data block
processing layer generates time stamps by its own ways.

For out-packet stream, the packet streaming layer guarantees 8,000 times
calls of data block processing layers per sec. Therefore, when cycle count
of the first packet is recorded, data block processing layers can calculate
own time stamps with the recorded value.

For the reason, this commit allows packet streaming layer to record the
first cycle count. Each data block processing layer can read the count by
accessing a member of structure for packet streaming layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:24 +02:00
Takashi Sakamoto
59f6482c21 ALSA: firewire-motu: add an abstraction layer for three types of protocols
In an aspect of used protocols to communicate, models of MOTU FireWire
units are categorized to three generations.

This commit adds an abstraction layer of the protocols for features
related to packet streaming functionality. This layer includes 5
operations.

When configuring packet streaming functionality with sampling rate and
sampling transmission frequency, .get_clock_rate and .set_clock_rate are
called with proper arguments. MOTU FireWire series supports up to 192.0kHz.

When checking current source of sampling clock (not clock for packetization
layer), .get_clock_source is used. Enumeration is added to represent the
sources supported by this series. This operation can be used to expose
available sampling rate to user space applications when the unit is
configured to use any input signal as source of clock instead of crystal
clock.

In the protocols, the path between packet processing layer and digital
signal processing layer can be controlled. This looks a functionality to
'mute' the unit. For this feature, .switch_fetching_mode is added. This
can be used to suppress noises every time packet streaming starts/stops.

In a point of the size of data blocks at a certain sampling transmission
frequency, the most units accept several modes. This is due to usage of
optical interfaces. The size differs depending on which modes are
configured to the interfaces; None, S/PDIF and ADAT. Additionally, format
of packet is different depending on protocols. To cache current size of
data blocks and its format, .cache_packet_formats is added. This is used
by PCM functionality, packet streaming functionality and data block
processing layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:23 +02:00
Takashi Sakamoto
5e03c33e3d ALSA: firewire-motu: add a structure for model-dependent parameters.
MOTU FireWire series doesn't tell drivers their capabilities, thus
the drivers should have and apply model-dependent parameters to detected
models.

This commit adds a structure to represent such parameters. Capabilities
are represented by enumeration except for the number of analog line
in/out. Identification name also be in the structure because the units has
no registers for this purpose.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:21 +02:00
Takashi Sakamoto
8865a31e0f ALSA: firewire-motu: postpone sound card registration
Just after appearing on IEEE 1394 bus, this unit generates several bus
resets. This is due to loading firmware from on-board flash memory and
initialize hardware. It's better to postpone sound card registration.

This commit applies this idea.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:19 +02:00
Takashi Sakamoto
6c3cef4890 ALSA: firewire-motu: add skeleton for Mark of the unicorn (MOTU) FireWire series
This commit adds an new driver for MOTU FireWire series. In this commit,
this driver just creates/removes card instance according to bus event.
More functionalities will be added in following commits.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-03-28 12:33:16 +02:00
Charles Keepax
8e71321d19 ASoC: cs35l35: Clear reset_gpio on the error path in probe
The error path in probe attempts to put the device back into reset.
Should we fail to get the reset_gpio (such as a probe defer) we will
leave the error value in there, which the gpiod_set_value_cansleep on
the error path will attempt to deference.

Fix this issue by clearing reset_gpio before we head into the error
path.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-27 17:26:53 +01:00
Colin Ian King
74a4ce4c8e ASoC: intel: remove unused variable data and associated code
The variable 'data' is assigned null and never re-assigned. There
is also a redundant check for data being non-null which is always
false, so remove this and the variable data and dma_addr as they
are not used once the dead code has been removed.

Detected with CoverityScan, CID#1324015 ("'Constant' variable gaurds
dead code")

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-27 12:31:43 +01:00
Takashi Iwai
36d96039e7 ASoC: intel: Don't print FW version repeatedly
Intel SST driver spews an info message "FW Versoin xxxx" at each time
the device gets initialized.  Since it's triggered at each PM (or even
runtime PM), it appears so ofetn, and rather becomes annoying than
useful.

This patch suppresses the superfluous messages by checking the
currently loaded FW version with the previously loaded one.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-27 12:30:55 +01:00
Daniel Baluta
3c01b9ee2a ASoC: codec: wm8960: Relax bit clock computation
WM8960 derives bit clock from sysclock using BCLKDIV[3:0] of R8
clocking register (See WM8960 datasheet, page 71).

There are use cases, like this:
aplay -Dhw:0,0 -r 48000 -c 1 -f S20_3LE -t raw audio48k20b_3LE1c.pcm

where no BCLKDIV applied to sysclock can give us the exact requested
bitclk, so driver fails to configure clocking and aplay fails to run.

Fix this by relaxing bitclk computation, so that when no exact value
can be derived from sysclk pick the closest value greater than
expected bitclk.

Suggested-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-24 18:53:06 +00:00
Daniel Baluta
3ddc97211c ASoC: codec: wm8960: Refactor sysclk freq search
Add a separate function for finding (sysclk, lrclk, bclk)
when the clock is auto or mclk. This makes code easier to
read and reduces the indentation level in wm8960_configure_clocking.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-24 18:53:04 +00:00
Dmitry Torokhov
73548dd316 ASoC: jack - check status of GPIO-based pins on resume
For GPIO-backed pins that are not configured as wakeup sources, we may
miss change in their state that happens while system is suspended. Let's
use PM notifier to refresh their state upon resume.

Signed-off-by: Dmitry Torokhov <dtor@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-24 18:52:32 +00:00
Kuninori Morimoto
6b8530cc05 ASoC: rcar: ssi: don't set SSICR.CKDV = 000 with SSIWSR.CONT
R-Car Datasheet is indicating "SSICR.CKDV = 000 is invalid when
SSIWSR.WS_MODE = 1 or SSIWSR.CONT = 1".
Current driver will set CONT, thus, we shouldn't use CKDV = 000.
This patch fixup it.

Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-24 18:49:45 +00:00
Hiroyuki Yokoyama
dc2721564f ASoC: rcar: enable PCM RATE untile 192000
R-Car sound can handle untile 192000 rate.

Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-24 18:48:24 +00:00
Bhumika Goyal
381ca1d12e ASoC: blackfin: constify snd_soc_ops structures
Declare snd_soc_ops structures as const as they are only stored
in the ops field of a snd_soc_dai_link structure. This field is
of type const, so snd_soc_ops structures having this property
can be made const too.

Cross compiled the .o files for blackfin architecture.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-24 18:48:10 +00:00
Lucas Stach
971edb0a00 ASoC: simple-card: fix simple_dai clk lookup
The clock needs to be stored in the simple_dai structure, so it can
be enabled later on. This has been broken during the conversion to use
devm_* functions for the clk lookup.

Fixes: e984fd61e8 (ASoC: simple-card: use devm_get_clk_from_child())
Signed-off-by: Lucas Stach <l.stach@pengutronix.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-24 18:46:36 +00:00
Arnaud Pouliquen
3c9d3f1bc2 ASoC: STI: Fix reader substream pointer set
reader->substream is used in IRQ handler for error case but is never set.
Set value to pcm substream on DAI startup and clean it on dai shutdown.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-24 18:44:07 +00:00
Kuninori Morimoto
991454e170 ASoC: core: remove pointless auxiliary from snd_soc_component
commit 1a653aa447 ("ASoC: core: replace aux_comp_list to ...")
tried to replace aux_comp_list to component_dev_list,
but it failed because of binding timing. Thus, Sylwester fixuped it by
commit d2e3a1358c ("ASoC: Fix binding and probing of auxiliary...").

One of main purpose of commit 1a653aa447 ("ASoC: core: replace...")
was remove replaceable list (= list_aux) from snd_soc_component by using
new "auxiliary" flags (but it failed).
Because of this background, current code has reborned card_aux_list
(= same as original list_aux), and almost pointless "auxiliary" flags.

Let's remove pointless "auxiliary" flags by this patch
This means, it is same as revert both
commit 1a653aa447 ("ASoC: core: replace aux_comp_list to ...") and
commit d2e3a1358c ("ASoC: Fix binding and probing of auxiliary...").

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-24 18:41:26 +00:00